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1. Eiter 7 Expression Clear Apply ho 005409 OD 059635 063055 e Ur ILZ 2 077793 102208 655721 3 4 5 6 ri z a 4 0000 DULO 0020 0030 004g OOo goap 0070 ANR Dee er Dosta SrA Plo 19m 6 11 4 Observe Registration Failure Outside the test lab you probably won t have occasion to captureSIP packets unless something is working incorrectly But then why else would you need a packet sniffer Simulating a registration failure is very easy Just alter the registration username of theSIP proxy profile in X Lite to one that doesn t match a SIP peer on the SIP server Be sure that your softPBX requires SIP clients to use a username and password you can configure Asterisk to allow anonymous registration Then start the Ethereal capture with the filter string used earlier restart X Lite and watch the registration crast and burn 6 11 5 Capture SIP Statistics Once you have saved a capture file you should be left with a trace screen that looks something like Figure 6 6 showing all the data packets captured downloaded from lib ommolketab ir ai E I ee a pa p pon pa d a a a E P F uo ii ga Lii Da LEF a Sal prone Probocc SIF SIP SIP SIF SIP SIP SIF Info Request REGISTER stoplu ls LI Status 100 Trying tl bindings status 401 unauthorized 1 bindings Request REGISTER sip id0 1 1 10 Status 100 Trying 1 bindings
2. downloaded from lib ommotketab ir The idea is to eliminate unnecessary Transmission Control Protocol Internet Protocol TCP IP listeners reducing the likelihood of an attacker discovering vulnerability So if you don t need Telnet the Trivial File Transfer Protocol TFTP talk and finger for goodness sake disable them The fewer services that have listening ports the more secure yourserver will be 7 2 3 3 Optimize the local firewall on the softPBX To build a local firewall policy on the softPBX server you ll need to identify which VolP protocols you re using and plan a policy based on the kind of TCP and User Datagram Protocol UDP port access needed by each one Table 7 1 lists the important protocols and their respective ports Table 7 1 Ports used for each VolP protocol Protocol Ports t 5060 eO and 5061 TCP and UDP H H323 Both TCP and UDP on Both TCP and UDP on ports 2099 and 2517 2099 and 2517 H 332 Video a k a H A TCP and UDP on port 2979 MEGACO H 248 248 TCP and UDP on TCP and UDP on ports 2944 and 29450 2944 and 2945 TFTP TCP and UDP on port 69 ASTMAN 5038 TCP Depends on configuration of capabilities negotiation preferences of the RTP endpoint RTP implementation many RTP agents use 5000 5001 5004 5005 10000 10001 8000 8001 or high numbered ports x AX ke 5036 UDP RSVP TCP TCP and UDP 3455 UDP 3455 TCP and UDP on ports 1756 1757 4056 and 4057 RTSP can vary by ses
3. downloaded from lib ommotketab ir Next you have a list of ports separated by spaces that will be placed in Class 2 the economy class In this class I ve put ports 10000 and 10001 Class 2 priority ports CLASS2PORTS 10000 10001 Once you have these values set to the correct port numbers save the script and exit Now all you have to do is run it astshape provider You shouldn t see any errors The astshape provider status Command will list the queues that have been defined 6 8 2 Explaining the AstShape Provider Script What is this doing How does this work What if need more Hold on Slow down The AstShape Provider script is actually quite simple Let me break it down for you on a line by line basis Here you can see that the first packet queue set up by tc is known as the root queue I ve told tc to use HTB queuing This is just one of many packet queuing techniques supported by the Linux kernel tc qdisc add dev SDEV root handle 1 htb default 30 This says to slow everything to SLINKSPEED to prevent queuing at our ISP Traffic class prioritization works only if you aren t being speed limited by the router on the other end of the connection te class add dev SDEV parent 1 classid 1 1 htb rate S LINKSPEED kbit burst 6k Now I ll add the first class of service the premium one as shown earlier tc class add dev SDEV parent 1 1 classid 1 10 htb rate S LINKSPEED kbit burst 6k prio 1 downloade
4. downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S D U M wW BI D 4 H 323 softphone SIP H 323 standard building H 323 gatekeeper using OpenH323 2nd 3rd 4th 5th VolP clients Hangup command hangup script hardening a server cleaning up xinetd removing unnecessary software hardware 2nd 3rd 4th configuring multiple IP phones at one time 2nd 3rd controlling house lights from IP phone 2nd creating a Zaptel interface card custom ringtone for Grandstream phone 2nd customizing Uniden IP phones from TFTP 2nd IP to IP phone calls with Grandstream BudgeTone 2nd 3rd recording calls from standard phone on a PC Sipura ATA 2nd 3rd using rotary dial phone with VolP 2nd headers 2nd headphones high availability telephony server high priority for voice media traffic historical newsgroup search tool Google Groups HotRecorder 2 0 for Windows HP iPAQ hx4700 Pocket PC HylaFAX downloaded from lib eommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H N J K L M N O P Q R S W U M wW K M K IAX Inter Asterisk Exchange protocol 2nd register feature with dynamic IP addresses support by JAJAH used with legacy signaling on Asterisk PBX VolP clients IAXy FXS gateway iChat ICMP packets ID3 library iLife applications images IMTO prefix incompatible phones inline
5. downloaded from lib ommotketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M wW K M K downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H H J K L M N O P Q R S W U M wW BI M K 3Com s 48 volt IntelliJack switch converter 3D Avatar Messenger 8 bit pulse code modulation formats 802 1p support by standalone Ethernet switches 802 1p precedence tagging checking support with pathping 802 3af standard for inline power 8x8 Inc s DI A 310 911 emergency service 2nd 3rd 4th 5th compromise solutions problems with VoIP emergency dialing TSPs and downloaded from lib eommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H N J K L M N O P Q R S O U M wW RI D Z access signaling by analog devices access signaling protocol SIP ACK method SIP 2nd Actiontec Internet Phone Wizard Address Book 2nd 3rd ADSL subscribers using Linux aLaw encoding Ambrosia Software Snapz Pro X WireTap Pro AMP Asterisk Management Portal 2nd 3rd 4th configuring MySQL database downloading and installing Perl modules running install script setup process software prerequisites analog modems analog phones and phone lines analog telephones connecting to Asterisk server connecting to SPA 2000 ATA hooking u
6. SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M wW K M 4 NAT Network Address Translation exploring NAT traversal 2nd 3rd 4th 5th 6th STUN protocol National Do Not Call Registry National Oceanic and Atmospheric Administration NOAA native Voice over IP ncurses Net Telnet Perl module NetFilter 2nd 3rd NetMeeting registering H 323 endpoint 2nd NOAA National Oceanic and Atmospheric Administration downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M wW K M K OhPhone 2nd OhPhoneX online forums online gaming 2nd OpenGK OpenH323 2nd 3rd 4th 5th 6th downloading and compiling OpenH323 Gatekeeper gnugk installing software packages OpenLDAP OpenMCU OpenSSL operating system OPTIONS method SIP 2nd Outbound Proxy setting X Lite Outlook 2nd 3rd Outlook contacts Ovolab downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S O U M wW BI D 4 packet interval packet jitter packets prioritizing to improve VolP quality panel discussion Party Line Gizmo Project Password setting X Lite pathping tool 2nd 3rd PBXs Asterisk server supporting four legacy phones 2nd 3rd 4th 5th building Mac PBX inking seve
7. Table 4 2 City Hostname Extension block Chicago chicago twidgets com 81XX downloaded from lib ommolketab ir downloaded from lib ommotketab ir City Hostname Extension block Tokyo tokyo twidgets com B2XX London london twidgets com 83XX 4 17 1 Configuring the Dial Plan On each Asterisk server you need to add a matching extensionfor each dial pattern So log into your server in Chicago and add the following to your internal context In etc asterisk extensions cont exten gt _82XX 1 Dial IAX2 guest tokyo twidgets com EXTEN 20 exten gt _82XX 2 Congestion exten gt _83XX 1 Dial IAX2 guest london twidgets com S EXTEN 20 exten gt _83XX 2 Congestion Let s take a look at what we have done so far In the first line we re telling Asterisk to create an extension that matches anything in the 82008299 range Remember thoseX s from before They signify to Asterisk any digit between O and 9 The first thing that Asterisk should do is try to reach that extension at the Tokyo office by using the AX protocol Version 2 with the usernameguest The guest username is just a placeholder You can use a more descriptive name if you want The IAX protocol is a signaling protocol like SIP which is more efficient at trunking multiple simultaneous calls between the same two locations If that extension at the Tokyo office is unreachable for any reason Asterisk will returncongestion Con
8. To make pure VolP calls using your TSP s service you have to be aware of the dialing shortcuts your TSP provides to route calls to other TSP networks using the Internetinstead of the PSTNas the carrier network Most VoIP TSPs will assume your call is destined for the PSTNjust because it s an 11 digit phone number So these shortcuts tell the TSP that you don t want to route your call to the PSTN Instead you want to route it over the Internet to another VoIP TSP Why do this If you have an unlimited calling plan it won t really save you any money The call probably won t sound any better either But this technique does conserve your TSP s public telephone network capacity when you use pure VoIP rather than VolP to PSTN calling If your VoIP TSP bills you by the minute it might not charge for calls that don t use its PSTN capacity Plus it s just cool to let the Internet replace the Bell System for your phone calls Here s how VoIP services such as FWD Vonage AXTel VoicePulse and Packet8 offer dialing shortcuts to allow calls between their customers If you re a Packet8 subscriber you can reach any FWD subscriber by dialing 0451 and the six digit FWD number assigned to that subscriber FWD subscribers don t have traditional 11 digit phone numbers because the service doesn t provide PSTN calling ConsultTable 1 2 for a rundown of the VoIP dialing shortcuts that you can use to route calls between the various VolP services Table 1 2 Pure
9. downloaded from lib ommolketab ir my S sec Smin hour mday mon Syear Swday yday isdst localtime time my Syyyymmdd sprintf 04d S02d 02d Syear 1900 Smon 1 Smday my Smonth S mon 1 my Svm_sound_dir vm sounds my Soffice_presence_file path to file office presence txt my SAGI Asterisk AGI gt new my sinput SAGI gt ReadParse my Sgreeting_start Start my Sgreeting_end_in_office endnormal my Sgreeting_end_out_of_office endooo my Sweekday_sound wday Swday my Smday_sound Smday my Smonth_sound month Smonth my files_to_play push files_to_play push Cfiles to play push files_to_play push files_to_play Sgreeting_start Sweekday_sound month Sound mday if e office_presence_file or Shour push files_to_play else push files_to_play 3 d Sgreeting_end_in_office Sgreeting_end_out_of_office foreach my Ssound files_to_play SAGI gt verbose Playing sound Ssound SAGI gt st ream file Svm sound dir Ssound Save the code to a file named vmautomate pl and place it in the var lib asterisk agi bin directory Add the following lines to yourextensions conf file where 8001 is your extension and 100 is your voice mailbox number exten gt 8001 1 Dial SIP 8001 20 rt exten gt 8001 2 AGI vmautomate pl exten gt 8001 3 Voicemail 100 After you reload Asterisk when you call your extension y
10. iptables A FORWARD P udp d 10 2 0 0 16 dest port 5060 j LOG log prefix ToDetroit iptables A FORWARD p udp s 10 3 0 0 16 dest port 5060 j LOG log prefix FromChicago iptables A FORWARD p udp d 10 3 0 0 16 dest port 5060 j downloaded from lib ommolketab ir downloaded from lib ommoltketab ir LOG log prefix ToChicago This example tags all the Detroit traffic separately from the Chicago traffic making it easier to discern later on when you re viewing the packet log A simple modification to the previous example would allow you to log RTP traffic port 8000 or whatever your endpoints use On a strategically placed Linuxfirewall this could provide valuable information about bandwidth consumption at the Detroit and Chicago sites in the example Another technique is to differentiate VolP traffic that is to from the private network from that which is to from the Internet or another foreign VoIP network iptables A FORWARD p udp s 10 0 0 0 8 dest port 5060 8000 j LOG log prefix Private VoIP iptables A FORWARD p udp d 10 0 0 0 8 dest port 5060 8000 j LOG log prefix Private VoIP iptables A FORWARD p udp s 0 0 0 0 0 0 0 0 dest port 5060 8000 j LOG log prefix Internet VoIP iptables A FORWARD p udp d 0 0 0 0 0 0 0 0 dest port 5060 8000 j LOG log prefix Internet VoIP In this example private trafficthat is traffic to or from a10 x x x hos
11. 10 1 1 203 OOODS344615E Apple Computer 10 1 1 10 OO104B64EEBC 3COM CORPORATION 10 1 1 201 OODEQS4BF9E4 Sipura Technology Inc OO01E1091434 Kinpo Electronics Inc 1 10 1 1 202 OOODSSB8E 254 Apple Computer E Hosts Routing J Passwords Lost packets 0 You re discovering all of these hosts and resolving their actual MAC addresses because Cain amp Abel will need these details to perform the ARP poisoning Once you can see the host you want to monitor in the list click the ARP tab and then click the toolbar icon again You ll see the New ARP Poison Routing dialog as shown in Figure 6 17 Figure 6 17 Cain amp Abel s ARP Poison Routing dialog downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Mew ARP Poison Routing WARNING II IAPR enables you to hijack IF traffic between the selected host on the lett list arid all selected hosts on the right list in both directions If a selected host has routing capabilities WAN traffic will be intercepted also Please note that since your machine has not the same performance of a router you could cause DoS tf you set AFR between your Default OO02FDO6DFSA 003046157305 000C I34AE 15E UNI U4BE4EE BL OOUE US4 amp 6F SE OOUTE 1031434 Gateway and all other hosts on your LAN DOODSSBSE ZARA 0001E 1091434 OODE SABF SES 001 O4B64EE BC OOUD 33446 7 5E 003046157305 000D SSB SE 254 Cance
12. After reading the User Guide it s time to boot Insert the CF make sure that the machine will boot from the CF and power on After POST you should see the GRUB menu with a few options available For now it s probably best to select the first entry By default AstLinux will attempt to obtain an IP address via DHCP on the first Ethernet interface that it finds and it will statically configure the secon interface with an RFC 1918 private address to do Network Address Translation NAT If this is not optimal for your situation will show you how to change this once the system boots After the usual kernel messages go by you should finally get to a login prompt log in with the username root and the password astlinux Now that you are logged in it s time to set up your system The first thing you are going to want to do is set up yourkeydisk As mentioned before a keydisk Is separate partition or device that AstLinux will use to store your configuration am going to assume that you are using a second Flash drive such as a USB pen drive for a keydisk and that the USB Flash drive is the Linux device dev sda When you boot with the CF disk it is considered an IDE device and should appear on your system as dev hdaso don t think I m asking you to overwrite your CF disk Verify that Linux can see the keydisk by typing the following fdisk 1 dev sda You should see the partition table for your device Make sure to take a good hard l
13. Andrew Latham Andrew is a networking consultant who offers VoIP IP networking and web development expertise via his web site http www lathama com Dave Mabe Dave http dave runningland com is an accomplished and largely self taught engineer and writer who strives to create simple elegant solutions to complex problems Dave has worked a AT amp T in the communications industry for eight years Always looking to save a few keystrokes and mouse clicks he is the kind of person who would rather spend several hours inventing an automated solution than spend a few monotonous moments each day performing a menial task Dave has been using Asterisk and other VolP solutions for two years He is the author of BlackBerry Hacks O Reilly Joel Sisko Joel has been a self proclaimed network convergence professional since 1992 He is the founder and CEO of Convergence Center LLC a company focused on delivering the next generation of convergence based applications and communications systems for value added downloaded from lib ommolketab ir downloaded from lib ommoltketab ir voice and data resellers downloaded from lib ommolketab ir downloaded from lib ommotketab ir Acknowledgments The contributors who spent their very scarce time working on material for this book have my earnest thanks These gentlemen are truly talented networking engineers and their input and technical expertise were invaluable Do business w
14. Now for the nitty gritty After you have AstLinux running and have made a keydisk you need to do away with the default Asterisk configuration There is just too much there for this simple task You can blowit away by using this simple command oh and if there s anything in this file you want to keep back it up first downloaded from lib ommolketab ir downloaded from lib ommotketab ir echo gt etc asterisk extensions conf You then need to add some basic meat back into extensions conf Open extensions conf in a text editor vi is included by default and add the following general Static yes writeprotect yes autofallback no globals VMBASE 8 XXX default includa gt viser exten gt 1 1 Hangup exten gt t 1 Hangup vmserv exten gt _S VMBASE 1 Voicemail uS EXTEN vmserv exten gt _S VMBASE 2 Hangup exten gt _9S VMBASE 1 VoicemailMain S EXTEN 1 vmserv exten gt _9S VMBASE 2 Hangup The first two lines under general tell Asterisk never to overwrite this file with something you tell it dynamically This is a good idea The next line tells Asterisk never to try to guess what to do if no action is assigned For this simple configuration it won t make much difference but it is generally a good idea The line under globals IS what you will want to pay the most attention to Here we are setting a variable named vmsase that will contain the value of our mailboxes In this confi
15. Save extensions conf and reload Asterisk with asterisk rx reload Hopefully after this hack you have realized that with four lines of configuration on three Linux boxes around the world Asterisk can revolutionize the way your organization communicates What would have been incredibly difficult and expensive to do just a few years ago has now been reduced to a few pages in a book It s truly amazing Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 57 Route Calls Using Distinctive Ring HAGK fol Do you have only one phone line but wish you could use two phone numbers with your Asterisk server Try distinctive ring Distinctive ring is a feature offered by some phone companies that permits you to use two or three phone numbers with the same POTS line Depending on which number is dialed the ring signal will differ causing the ring to sound unique for each number This feature allows parents to avoid answering their teenagers incoming calls With a fax voice ring switch device you can use distinctive ring aS an inexpensive way to receive both fax and voice calls on a single line On VolP trunks such functionality would be handled by out of band signaling Distinctive ring is a legacy signaling solution That is it works only with POTS With Asterisk you can use distinctive ring to route calls automatically from the PSTN trunk to a specific phone or grou
16. So don t delay dig in to this grab bag of desktop telephony ideas They re just the tip of the iceberg downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 8 Access Next Gen Voice Features HACK 8 Broadband VolP providers like Vonage don t just provide phone service If you know where to find the features they integrate with other applications on your desktopand with your digital life When you subscribed to your amazing new VolP telephone service you might have missed the fact that along with your new Internet calling money saving VolP service you also picked up some nifty desktop telephony enhancements Most of the broadband VolP phoneservice providers give you some cool extras that you d never get with a traditional phone companystuff like web based account management voicemail to email integration and even softphone calling from yourdesktop Did you know 2 2 1 Vonage Users Can Call Any Outlook Contact with One Click Vonage lets you place calls to your Outlook contacts with a special piece of software an add in called Click 2 Call which comes on the Vonage software CD Install it and launch Outlook You ll notice that your Outlook contacts now have a Click 2 Call option in their Actions menu Clicking this option dials the contact s phone number via your Vonage analog telephone adapter ATA and then connects the call with your phone Pick it up you should hear your call ringing in th
17. downloaded from lib ommolketab ir downloaded from lib ommotketab ir Pretty neat eh The other more difficult way to enable this feat is by modifying the boards themselves This means re creating the same modification that Digium does when it modifies Intel cards to create so called genuine X100P cards Remove the resistors marked R13 and R19 by unsoldering them But be careful and don t expect to return your Intel V 92 card as its warranty will now be invalid downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 65 Build a Speed Dial Service on Cisco IP Phones 6s Cisco s 7900 series IP phones have some powerful programmable firmware that you can harness for your own unique purpose answering the age old question Doesn t this LCD display seem a bit large for just caller I D That expensive Cisco phone on your desk has some great hidden capabilities Additional tools and toys that lurk beneath the 79xx s gray exterior could increase your productivity and foster some innocent fun I m talking about things like automated weather reports on the phone s display simple menu driven applications like a time card say and just about anything else you can program usinc an XML web site In fact you ll probably build so many cool tools and toys that you ll need a way to sort through them like a directory or a menu Using the Cisco VoIP phone s XML application capability you can set up such
18. Status 200 OK 1 bindings Reguest NOTIFY stp 2049G10 1 1 202 25060 Status 200 Ok Source port 5060 pestination port 5060 source port 3060 Destination port 5060 ER sip 0 Srp ero an Te 0 u 1 202 50 60 branch 2 AAS cord 20744 ccd Figure 6 6 An Ethereal packet capture log downloaded from lib eommolketab ir ethereal Lihace ain pi Ethereal Ee ee tev cane aro sa 6 he al Exoress0n Clear Apoly _ ee SS aamu iL oaIo IIM 1 0 000000 2 0 050048 3 0 060011 4 0 060391 1s ips 0 300053 SOANCE 192 168 1 100 64 69 76 10 192 168 1 100 192 Laps ee Ca 6S ae he 192 168 1 100 Destination 64 69 76 10 192 168 1 100 64 69 76 10 64 69 76 10 132 165 121 100 Protocol Info Tce 2616 gt http SYN Seq 0 Ack TCP http gt 2616 SYN ACK Seqe 0 2616 gt http ACK Seqel Ack GET fauto _lite yxten settin HTTP pi 1 aus Not Found tex cL 54 69 76 10 E Guar gt tack ee ne On a 1 92 16 3 1 1 00 5 1 1 ot PLS 43 4 Bee 16 Ws Ethernet II Srei Larkavnecaniec BS O070C 41 549 ec 88 Ost AambitMic_O4 7aiba O0 02 Internet Protocol Src 64 69 76 10 64 69 76 10 Dst 197 168 1 100 197 168 1 100 Transmission Control Protocol Src Port http 80 Ost Port 2616 2616 Seq 1 ack Hypertext Transfer Protocol Line based text data
19. X10 interface controllers connect to lights and other appliances in your house and your computer can send serial commands to the controllers to turn them on and off and adjust voltage like a dimmer Some X10 interfaces even offer telephone based user interfaces lettini you control them by calling them with your phone There are a few ways to integrate X10 controls with an Asterisk phone system The integration method depends on the type of X10 controller purchased For this hack chose the X1OTR16A phone controller that operates by DTMF digits Ordinarily you would hook a phone line to it and then call that phone line with a standard phone to operate the TR16A But with Asterisk you can connect directly to the TR16A as if you yourself are the phone company Then controlling the TR16A is as Simple as a dial plan hack in Asterisk The Asterisk system used to connect to the TR16A contains a single Wild card TDM400P using two foreign exchange station FXS modules and one FXO module The system has one analog phone anc one SIP enabled Polycom IP500 phone The SIP phone is the one I used as my remote control the phone from which sent my commands to the X10 controller performed the following steps to integrate the TR16A with the Asterisk system For the TRL6A controller 1 connected it to a suitable 120VAC outlet set the Answer delay switch to Minimum set the controller to the appropriate house code S set the four
20. don t have to play cards at all Just read Explore NAT Traversal Hack 76 l Il also show you how to clandestinely monitor and record actual VoIP phone callsthe IP equivalent of a phone tap downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 72 Monitor VoIP Devices N72 The only thing worse than having a VolP service outage is being the last to know about it It s the phone company s job to monitor traditional telephony links Some legacy phone vendors such as Avaya even monitor their PBXs in the field via phone links back to their support headquarters But no such convenience exists for downtime wary VolP administrators Thanks to Perl though good system monitoring for VoIP is within your grasp In this hack you ll develop a Perl script that monitors SIP hosts over the network and reports back on their availability This script determines that the SIP host is alive by sending a SIP OPTIONS packet to the remote host and receiving a response This determines not only whether the host is reachable via the network but also whether the SIP application on the other end is listening and responding to requests 6 2 1 The Code Your Perl development environment will need theTime HiRes module for this hack Grab it from http search cpan org dist Time HiRes usr bin perl use 10 Socket use POSIX strftime use Time HiRes qw gettimeofday tv_interval use Getopt Long use strict
21. downloaded from lib ommolketab ir downloaded from lib ommoltketab ir What could be more useful than tracking when and whom you Skyped This capability is actually buil into Skype via its call list feature But using Out look s J ournal feature to track the whens and whos of Skype is far more useful The Journal puts Office events and now Skype events in a chronological searchable view that resembles a timeline and provides a sort of audit trail for change made to files messages and contacts within Microsoft Office Long after a particular message contact or Skype call has been deleted the journal still retains a record of itwhen it happened who called or who you called and how long But the Journal option for Skype isn t enabled unless you tell it you want itan option you ll find when you click the Configuration option located in the drop down menu on the Skype Toolbar for Outlook If you d like a slightly different approach to saving a log ofcalls made and received with Skype try Avantlook http share skype com directory avantlook view another Skype toolbar add on for Outlook Instead of using the Office J ournal it actually stores Skype events as items in its own searchable sortable message folder downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 30 Skype People from the OS X Address Book 30 Using AppleScript and the Mac OS X Address Book you can Skype any phone number
22. downloaded from lib ommotketab ir VoIP Hacks By Theodore Wallingford Publisher O Reilly Pub Date December 2005 ISBN 0 596 10133 3 Pages 326 Table of Contents Index Overview Voice over Internet Protocol VoIP is gaining a lot of attention these days as more companies and individuals switch from standard telephone service to phone service via the Internet The reason is simple A single network to carry voice and data Is easier to scale maintain and administer As an added bonus it s also cheaper because VoIP is free of the endless government regulations and tariffs imposed upon phone companies VoIP is simply overflowing with hack potential andVolP Hacks is the practical guide from O Reilly that presents these possibilities to you It provides dozens of hands on projects for building a VoIP network showing you how to tweak and customize a multitude of exciting things to get the job done Along the way you ll also learn which standards and practices work best for your particular environment Among the quick and clever solutions showcased in the book are those for e gauging VoIP readiness on an enterprise network e using SIP H 323 and other signaling specifications providing low layer security in a VolP environment e employing IP hardphones analog telephone adapters and softPBX servers e dealing with and avoiding the most common VoIP deployment mistakes In reality VolP Hacks contains only a small subset of Vo
23. gt s 102 Playback carried away by monkeys exten gt s 103 Hangup downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 49 Build a Four Line Phone Server sag Create a simple fully functional small office PBX An Asterisk server can be the nerve center of two kinds of telephony networks an all VolP network that uses only IP based connections to route calls or a hybrid VolP legacy network that uses both IP and time division multiplexing TDM technologies to route calls In this hack you ll use a Digium TDM40OOP card to turn your Asterisk server into a full blown PBX that can support up to four legacy phones or four legacy phone lines at a time These legacy devices wil be able to call and be called by VolP phones This setup is depicted inFigure 4 10 Figure 4 10 Asterisk as a simple small office PBX The TDM400P card has four modular interfaces that can host either FXO or FXS modules FXO modules allow you to connect phone lines to your server and FXS modules let you connect phones You can use any combination of FXO and FXS modules up to four on a single TDM400P so you can connect two analog phones and two phone lines or one phone line and three analog phones and so on When you purchase the TDM400P card you can specify what combination of interfaces you d like Here s the main thing to remember so you don t get the wrong configuration the redcolored FXO modules connect phone line
24. my SUSAGE Usage sip _ping pl v t s lt src_host gt p lt src_port lt hostname gt my SRECV_TIMEOUT 5 how long in seconds to wait for a response my sock I0 Socket INET gt new Proto gt udp LocalPort gt 6655 ReuseAddr gt 1 or die Could not make socket Sa options my verbose Shost omy ip mmy port Sime GetOptions verbose v gt Sverbose downloaded from lib ommolketab ir downloaded from lib eommolketab ir sSource ip s s gt Smy_ip source port p n gt Smy_port time t gt Stime or die Invalid options n n USACE figure out who to ping my host shift ARGV or die SUSAGE my Sdst_addr inet_aton host or die Could not find host host my Sdst_ip inet_ntoa dst_addr my Sportaddr sockaddr_in 5060 Sdst_addr figure out who we are smy_ip 127 0 0 1 unless defined Smy_ip smy_port 6655 unless defined Smy_port callid is just 32 random hex chars my Scallid pcallid 70 209 a an E lane rand for I uw S25 today s date my Sdate strftime a e SB SY I M S Z localtime branch id see rfc3261 for more info using time for uniqueness my Sbranch z9hG4bK time my Spacket gq OPTIONS sip Sdst_ip SIP 2 0 Via SIP 2 0 UDP S my_ip my_port branch S branch From lt sip iping omy 10 gt To 221 host Contacti lt sipiping Gony 1p Call ID callid my_ip CSeq
25. 2L USZ Matoh 1p Spore 23 UXILII tlowid 1720 to filter add dav ODEV parent 1 0 protocol ip prio 24 Uso2 maton Ip Aport SS UXIIII ilowid Lazo DNS is a very time sensitive protocol where delays in name resolution can usually be noticed by a user or application very easily Also DNS queries are not very large so it is to our benefit to add them to a higher class of service You will see several lines that talk about Transmission Control Protocol TCP ACKs These are TCP acknowledgments and they won t be covered in this book Trust AstShape and me by leaving this alone Finally we assign whatever is left to the earlier default class te filter add dev SDEV parent 1 protocol ip prio 30 132 Maren ip dst 0 0 0 070 Llowid 1730 As you can see AstShape is a very simple yet powerful traffic QoSscript A big thanks goes to the folks at LARTC for providing WonderShaper which AstShape was based on For more information on traffic shaping QoS under Linux please visit the LARTC web site athttp www lartc org Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 78 Create a Premium Class of Service HACK 78 You can produce high octane voice service for premium users by building distinct classes of service In this hack we will be using theAstShape script that we played with in Shape Network Traffic to Improve Quality of Service Hack 77 However you ll now be prioritiz
26. Application The Asterisk software function handling the call downloaded from lib ommolketab ir downloaded from lib ommotketab ir CDR field Description Last Data Sent to Information the application uses to connect the call Application Start Time The time of first contact from the caller to the softPBX Answered Time The time the receiving endpoint answered if applicable The time of the end of the call regardless of whether it was End Time answered Duration The length in seconds from the first contact to the end time Billable Duration The length in seconds of connected billable time during the call Disposition The last known status of the call during this application AMA Flaas Automated Machine Accounting flags used by some telephony billing a software The field names marked with an asterisk record Asterisk proprietary information For example the Application field might not have a meaningful correlation on another softPBX because not all softPBXs refer to telephony functions as applications The idea here is that once the CDR is imported intoExcelor another data analysis toolyou can interpret it in interesting ways Suppose you want to figure out which customer places the most calls to your technical support department You can count occurrences of that customer s caller ID in the CDR Or if your teenage daughter is receiving a dozen calls a day you can bill he
27. If you want to run OpenH323 on Windows use the precompiled executables The instructions I m providing are for Linux First download and install PWLib Save pwlib 1 5 2 tar gz or the filename appropriate for the version you download to root as the root user Then unzip and untar it tar xvzf pwlib_1 5 2 tar gz Now you ll need to set some environment variables so that the OpenH323 software knows where to find the PWLIib libraries PWLIBDIR S HOME pwlib export PWLIBDIR OPENH323DIR HOME openh323 export OPENH323DIR LD LIBRARY PATH SPWLIBDIR 1ib SOPENH323DIR 1ib export LD_LIBRARY PAT HH HH H If you plan to make this H 323 setup permanent you should add the preceding environment variable commands to bash_ profile in root Now build the PWLib distribution by using make cd SPWLIBDIR configure make opt make install Next download the main OpenH323 file to root Then unzip and untar it substituting the filename that s appropriate for the version you download tar xvzf openh323_1 12 2 tar gz downloaded from lib ommolketab ir downloaded from lib ommotketab ir Now build OpenH323 cd SOPENH323DIR configure make opt make install The developers recommend a 128 MB swap partition to complete the build error free This need is minimized if you have enough physical RAM 256 MB of physical RAM should be plenty This build could run for 30 minutes or more so enjoy a delicious
28. Sip_ping pl script recording latency and jitter data 2nd 3rd 4th server responding to host sending packet instead of host in Contact header SIPphone Inc Sipura ATAs building a bat phone 2nd 3rd 4th tweaking dial plan 2nd 3rd Sipura SPA 2000 ATA Sipura SPA 3000 ATA 2nd 3rd sipX 2nd 3rd 4th finishing sipXpbx setup by web interface implementation of components of a SIP network installing sipXpbx aunching sipXpbx requirements sipXphone Skype 2nd 3rd 4th 5th 6th 7th 8th 9th 10th 11th 12th 13th 14th 15th 16th 17th 18th 19th 20th 21st 22nd 23rd 24th 25th 26th 27th 28th 29th 30th 31st 32nd capabilities and limitations of contact search function custom rings and sounds download site emoticons 2nd how it works ignoring or ending calls integraging into podcasts 2nd 3rd Jyve social networking service 2nd 3rd Jyve web browser plug in 2nd logging calls with Outlook Journal feature placing calls from Outlook 2nd putting Skype Me link on your web site or blog searching user directory for contacts security signaling protocol SkypeOut international calls using WiFi enabled Pocket PC as portable phone using with gaming Voicemail service Skype Me mode Snapz Pro X sniffing or capturing packets with Ethereal 2nd 3rd 4th 5th 6th 7th Snom 200 SIP hardphone softPBX 2nd softphones downloaded from lib ommolketab ir downloaded from lib eommotketab ir echo recording conversations with WireTap using with a VoIP TSP 2nd 3rd 4
29. Then click Start Firmware Upgrade After you ATA has rebooted the update will be finished downloaded from lib ommolketab ir nioaded from lib ommotlketab ir Chapter 2 Desktop Telephony nioaded from lib ommolketab ir downloaded from lib ommotketab ir 2 1 Hacks 827 Introduction To take advantage of computerized telephony you don t need a VolP gateway a fancy Internet Protocol IP phone or an open source PBX though those are certainly fun hackworthy telephony goodies Your desktop PC can be the nerve center of all your voice communications replacing your telephone your caller ID display your answering machine or voicemail and possibly even your phone bill some VoIP services will bill you electronically Some pretty amazing software goodies are available to make your voice communication life a real joy Programs like Gizmo Project and Skype let you make voice calls to buddies around the globefor free Some of these programs have built in voicemail and call recording and most are cross platform offering support for Mac Windows and Linux Hardware contraptions and telephony automation software bring even more exciting capabilities to the table With a telephone line interface for the Mac or a voice modem in a Windows PC all you need is the right software to tie your phone completely to your desktopbut don t forget your wireless headset VolPing is much cooler when you aren t physically bound to your PC
30. There s a nice Asterisk Manager API reference at the end of my bookSwitching to VolP O Reilly There nowyou have the tools to build an Asterisk empire using Perl and Perl alone Gi now and conquer downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 95 Build a SoftPBX with No Hard Drive 95 You don t really need a hard disk to run a phone system even an IP based softPBX doesn t need a hard drive A CompactFlash based PC will do the trick Sometime around September of 2004 was looking at one of PC Engines WRAP boards and wondering how well it could run Asterisk Knowing that would not want to run a full size distributior started pulling apart a Gentoo install removing components that are not critical to the functionality of Asterisk After a fairly significant amount of work was left with a slimmed down Gentoo that fit on a 256MB CompactFlash card which was the smallest that had at the time and would run mounted read only After working on the init system and writing some extra scripts decided to put it up on my web site just in case someone else found it interesting or useful decided to call it AstLinux version 0 1 0 After about 4 000 downloads think that had my answer and AstLinux was born By 2005 realized that to make AstLinux truly spectacular was going to have to make it smaller and more flexible Work on AstLinux 0 2 x began After messing around quite a bit with
31. a E M E M e Se iia nae nae E D E 2162252726 2162261266 2162454103 2163732776 21664326843 2167140500 43306454256 4402415946 4403284141 4408646084 4408887535 6505961907 6505961908 6505961911 68314622553 9037489990 9375564124 Grand Total i P imb bet bh pa a e aa a Lat a best Pig E e A l ii E po i ja he ned Sh heetl Master COR csv ii _ T zi Ready Page 1 1 Experiment with the other columns What can Excel tell you about your call activity With these reports you ll have a handle on precisely who called whom and when That way if your mom ever says Why don t you ever call you ll have the perfect response Mom we talked 108 minutes last week alone and 96 minutes the week before Figure 4 9 A Pivot Table Report that shows detailed minute totals for each extension for every phone number called downloaded from lib ommolketab ir downloaded from lib eommoltketab ir e00 gt A 1 Count of Duration 2 SOUTCI p Pew mine Ww my ells zi downloaded from lib ommolketab ir ji i e Ready Destination 2162252728 21637352776 12162252728 12162261266 13307862847 14402480100 14403241778 14403284141 19 14405033947 Sheetl Wi Ee downloaded from lib ommotketab ir Hack 48 Kindly Introduce Telemarketers to Mr Privacy HAGK 48 If you
32. downloaded from lib ommolketab ir downloaded from lib ommoltketab ir general port 5060 Port TO Dind Ce 51P 19 50G0 bindaddr 0 0 0 0 Address to bind to all addresses on machine disallow all allow ulaw allow alaw tallow gsm The G 711 and GSM codecs have been commented out This simulates a codec capabilities mismatch so the SIP client won t be able to pass the SDP negotiation and call setups will fail Don t forget to issue a reload command at the Asterisk command line Now we can run the capture again while X Lite tries to call extension 500 In Ethereal select Capture Start Only this time the call will fail because there s no suitable codec common to both the caller X Lite and the receiver Asterisk Packet 7 is shown in Figure 6 10 It s a SIP INVITE carrying SDP content that includes a list of a tokens These represent media attributes or capabilities Ethereal presents the SDP content in a parsed hierarchical fashion The raw ASCII SDP payload of this SIP packet which can be seen in X Lite s diagnostic log actually looks like this content Type application sdp User Agent X Lite release 1103m Content Length 290 v 0 0o 203 146336832 146337009 IN IP4 10 1 1 201 s X Lite c IN IP4 10 1 1 201 t O OQ m audio 8000 RTP AVP 0 8 3 98 97 101 a rtpmap 0 pcmu 8000 a rtpmap 8 pcma 8000 a rtpmap 3 gsm 8000 a rtpmap 98 iLBC 8000 a rtpmap 97 speex 8000 a rtpmap 101 telephone event 80
33. drwxr xr x 10 tedwalli admin 340 don 2 Aara drwxXrwxr X LO FOOL admin 544 Jun 6 19 50 rw r r 1 tedwalli admin 6148 Jun SS 1 tedwalli admin 94 Jun y 1 tedwalli admin 195002 Jun 7 1 tedwalli admin 7140222 Jun yr 1 tedwalli admin 7O0l0L aun 1 tedwalli admin 694450 Jun i r 1 tedwalli admin 801000 Jun A a A 1 tedwalli admin Lasila Jun 6 19 50 DS Store 6 19 50 greeting txt 9 22 39 greeting_friday aif 9 22 39 greeting_monday aif 9 22 39 greeting_thursday aif 9 22 39 greeting_tuesday aif 9 22 39 greeting_wednesday ailif 6 19241 ring air Now create a quick shell script like this one for each day of the week 71 Oin ea This script is for Monday cd Library Application Support Phlink Items cp f greeting_monday aif greeting aif Save each daily shell script in a convenient place perhaps in acronjobs folder in the user profile of that phlink user we made in Run Phlink Even When Logged Off Hack 16 Don t forget to make them executable run chmod 755 in the directory where you ve put them Then add each file to downloaded from lib ommolketab ir downloaded from lib ommotketab ir the last column of the etc crontab file which defines scheduled Unix activities that should run in the background on Mac OS X In this example the daily file rotation occurs at 6 30 a m from Monday through Friday the Friday greeting remains in place until Monday morning Gron Jobs to rotate the phlin
34. festival tts exten gt s 2 Dial SDEFAULTPHONES 7 6 2 Mac the Hack If your Asterisk server runs on a Macintosh Hack 93 you can do the caller ID announcements using the Mac Unix command Say instead of Festival One of the coolest things about the Mac s built in speech synthesis is its selection of different voice styles males females whispers and hysterical laughing all make Mac speech a lot of fun In fact you can use thesay command s v option to use those different voices depending on the variables of the call Here different voices are used depending on which phone is being dialed downloaded from lib ommolketab ir downloaded from lib ommotketab ir exten gt 10 1 System say v Agnes Kelly you have an incoming call exten gt 10 2 Dial SKELLYSPHONE exten gt 20 1 System say v Hysterical Jake you have an incoming call exten gt 20 2 Dial STEDSPHONE The preceding sample uses the Agnes voice to announce Kelly s calls and the Hysterical voice to announce Jake s calls downloaded from lib ommolketab ir downloaded from lib ommotketab ir PREY Hack 93 Build a Mac PBX 93 The Mac mini is a very tiny and rugged PC making it a great small office PBX When Apple introduced the Mac mini most onlookers were pleased to see a smaller machine with plenty of muscleenough to handle a few dozen VolP phone calls at a timeeven though most observer didn t have VoIP in mind fo
35. lathama com services cisco downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Leave the ending slash in there It will tell the web server to grab the default or index file Testing this with a web browser will be difficult as theXML has no headers to tell the browser what is going on Create a file called index html default htm for Windows web servers on a nearby web server that s accessible at the URL you supplied to the phone and fill it with XML content like this lt CiscolPPhoneMenu gt lt Title gt My New Menu lt Title gt lt Prompt gt Prompt This lt Prompt gt lt Menultem gt lt Name gt Menu Item lt Name gt lt URL gt http domain cisco services menuitem xml lt URL gt lt Menultem gt lt SortKeyltem gt lt Name gt Soft Key lt Name gt lt URL gt http domain cisco services softkey xml lt URL gt PoOSt1C1 On 1 lt Posititon lt SoftKeyItem gt lt CiscolPPhoneMenu gt This sample shows you a simple yet slick method of loading a menu When you press the Settings button on the phone the menu shows up as simple text bars and allows you to arrow down or up to the option you choose using the phone s navigation controls and softkeys After you make a selection the listed URL is queried This convention is the formula behind what Cisco callservices You can think of services as the little applications stock tickers games etc on modern cellular phones The Cisco phones spl
36. machine Before you can go any further though you need to be certain that yourAsterisk instance is running as a nonroot user To do so follow the recommendations in Run Asterisk Without Root for Security Sake Hack 54 because the rest of the AMP installation is going to assume your Asterisk instance runs as a nonprivileged user But keep your finger on this page because there s a lot more to do 4 14 2 2 Install Perl modules and telecom tools You can download AMP from http amportal sourceforge net Unpack the AMP source distribution using tar there are numerous examples of tar unpacking throughout this book into the usr src directory Once it s unpacked install theNet Telnet Perl module from CPAN which allows Perl based packages such as AMP to use Telnet sockets perl MCPAN e install Net Telnet Now to enable AMP s music on hold upload feature you can usevi to make a few modifications to the PHP configuration PHP 4 users might need to substitute etc php4 apache2 php ini in place of etc php ini The idea here is to increase the upload_max_filesize value to 20M and to change the corresponding LimitRequestBody value to 20000000 This way you ll be able to use AMP to upload large files like music on hold MP3s vi 482 etc php ini upload_max_filesize 20M vi 14 etc httpd conf d php conf LimitRequestBody 20000000 If you ve already modified your PHP configuration files these commands will not work corre
37. remote SIP endpoints am going to assume that you want incoming calls to the FXO port on the Sipura to be forwarded to an extension on that existing Asterisk server I m using 1000 for this hack I m also assuming the Sipura and the Asterisk server are on the same LAN and the server s IP address is 192 168 1 1 On your Asterisk server open up etc asterisk sip conf and create a new entry at the bottom of the file spa3k type friend username spa3k secret spa3k lt Pick a new password and write it down dtmfmode rfc2833 host dynamic context default nat yes allow all downloaded from lib ommolketab ir downloaded from lib ommotketab ir Save sip conf and reload Asterisk with asterisk rx reload If you would like to place outbound calls using your new SPA 3000 continue reading Otherwise you can skip ahead to the section Configuring the Sipura Next we will need to edit etc asterisk extensions conf Underneath the globals section add a new line TRUNK SIP spa3k If you already have a TRUNK variable defined it is up to you to figure out how you want to mix and match your existing trunk s with your SPA 3000 Now scroll down to the bottom of the file and add a new section here LSepa Trunk exten gt _NXXXXXX 1 Dial S TRUNK S EXTEN 20 exten gt _NXXXXXX 2 Congestion exten gt _NXXNXXXXXX 1 Dial S TRUNK S EXTEN 20 exten gt _NXXNXXXXXX 2 Congestion exten gt _1NXXNXXXXXX 1 Di
38. text html 7a ba 00 60 fe a8 40 00 34 s s acu 02 010 06 fe 45 4c 0a cO ab eed QEL 0020 01 64 00 50 Oa 38 82 Sf 7O be le bf ef 6c 50 18 vd PB Peeee IP 0030 83 2c 91 03 00 00 48 54 54 50 2f 31 2e 31 20 34 ereas HT TP 1 1 4 040 30 34 20 4e f 74 20 46 f 75 5e 64 Od Oa 44 l 04 NOT F ound Da D050 74 65 3a 20 57 65 64 2c 20 30 33 20 41 75 6 20 te wed 03 AUG 9060 32 30 30 35 20 30 36 3a 30 30 34 31 32 20 47 Ad 005 O06 00 12 GM To help sort through all this captured data you will useEthereal s handy SIP statistics tool by navigating to Statistics SIP This tool produces a simple report on the statistics of the SIP messages sent and received on the interface as shown in Figure 6 7 Figure 6 7 Ethereal s SIP statistics dialog downloaded from lib eommotketab ir downloaded from lib ommoltketab ir sip stationl OX SIP stats 17 packets 0 rasent packets Informational SIP ixx SIP 100 Trying SIP 160 Ringing 1 Success SIP 2xx SIP 200 Ox 5 Chert errors SIP 4x Server errors SIP Sxx Gobal failures SIP 6xx List of request methods INVITE 2 packets Ethereal has many built in tools for filtering and colorizing data traces but it can be a bit overwhelming to digest all of the data at once To help network administrators graphically view the call flow of a SIP signaling conversation navigate to Statistics VolP Calls You are presented with a list of captured voice calls f
39. usr sbin asterisk vvvc The more v s you tack on the more verbose Asterisk s debugging output will bevery useful when troubleshooting things If you ve issued this command in the OS X terminal and received the following message or something similar you weren t using the root account when you launched Asterisk Logger Warning Unable to open log file var log asterisk messages Permission denied Figure 7 6 The Asterisk installer package BOO mw install Asterisk Select a Destination Introduction Select a destination volume to install the Asterisk software B License B Select Destination You ll need to be root the all powerful Unix administrator account to launch Asterisk as installed by the installer package Becoming root on OS X isn t as simple as it is on Linux however The root account itself is actually hidden away and disabled until you go in and turn it on manually This practice is generally frowned upon as using the root account gives you enormous capacity to harm your Mac s filesystem and settings so don t say didn t warn you We re just going to do this once SO you can see how it s done to launch Asterisk and then we ll return to the safe zone of nonroot access As the default user the first user created when you first installed OS X on your Mac open a terminal and issue this command downloaded from lib ommolketab ir downloaded from lib ommoltketab ir s
40. 102 OPTIONS User Agent sip_ping pl Date date Allow ACK CANCEL Content Length 0 Ls send the packet print Sending n nSpacket n if Sverbose send Ssock Spacket 0 Sportaddr length packet or die cannot send to Shost p my S send_time gettimeofday start the stopwatch my Selapsed get the response eval local SSIG ALRM sub die alarm time out alarm SRECV_TIMEOUT Sportaddr recv sock Spacket 1500 0 or die couldn t receive Selapsed tv_interval S send_time stop the stopwatch alarm 0 1 or diel i downloaded from lib eommolketab ir downloaded from lib ommotketab ir t peint our ou tput if verbose printf After S0 2f ms host said n n Ss n Selapsed 1000 Spacket elsif Stime printf s0 2f n Selapsed 1000 else print host is alive n 6 2 2 Running the Code Execute this script using a command like this sip_ping pl 192 168 0 123 If you see output like this the SIP host is indeed running 192 168 0 123 is alive Using the v option with this command gives more verbose outputspecifically the contents of the SIF response and the round trip latency in milliseconds of the SIP message exchange You can work this technique into other monitoring tools and run it periodically to inform yourself of any outages Because the majority of commercial VoIP service providers use the SIP proto
41. 4th 5th 6th 7th 8th capturing failed capabilities negotiation Capturing successful capabilities negotiation observing SIP registration SIP registration failure sniffing out jittery calls 2nd 3rd Ethernet connections Ethernet interface Ethernet port Ethernet switches Tos feature euros extended range base station for wireless LAN extensions conf file context for SIP phones dialing out on Asterisk PBX with legacy phone lines context section dial plan for SIP peer incoming context extra features Gizmo Project downloaded from lib ommolketab ir downloaded from lib ommotketab ir Map It Record It downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H N J K L M N O P Q R S O U M wW BI D Z fast busy fax calls fax machine turning Linux box into 2nd 3rd using analog modems over traditional phone lines fax receiving support for AMP fax modem card faxes 2nd Festival soeech synthesizer 2nd 3rd reading weather report Festoon FIFO first in file format conversion sound files find me follow me call list Firefly support for SIP and IAX firewall functions in ATAs firewalls IAX and NAT NetFilter 2nd 3rd optimizing local firewall on softPBX SIP and Flash storage device forums four line phone server Asterisk setting up incoming calls setting up outgoing calls four wire patc
42. 64 156 66 10 LS penguin bigv kxdirect com 65267 1429 23 The following example which uses thet option checks to see whether each router along the path Supports 802 1p precedence tags These tags are used by routers and switches to prioritize real time delay sensitive packets such as the UDP datagrams commonly used in VoIP The more hops along a route that support 802 1p the better off your VoIP quality on that route is likely to be because thoss routers can prioritize voice data over nonvoice data C gt pathping www bigvoxdirect com T Checking for connectivity with Layer 2 tags UVediediead Orke SO wow 64 I OR 24 131 64 38 OK 12 244 65 61 OK Lloedecell GeizZk ORK 12 123 137 26 General failure Oy GI ae to bo fe The output from pathping first shows the route like the previous example but also adds the 802 1p feedback as far along the route as possible Not all devices along every route support 802 1p In this example the sixth hop does not because the router isn t configured for PType of Service ToS and 802 1p Since the 802 1p header can t be carried past the sixth hop subsequent hops cannot be tested for 802 1p support The R option will do a similar check for RSVP support in a similar fashion You aren t nearly as likely to find RSVP supporting hops on the public Internet as you are 802 1 aware hops But if RSVP is configured on a private network you can usepathping to help you evaluate that network s hard
43. And speaking of Google to get the most out of Google when using telephony read Google for Telephony Info Hack 21 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 11 Sound Like Darth Vader While You VoIP HACK 11 Using Audio Voice Cloak you can sound like Darth Vaderor like Alvin and the Chipmunkswhile you talk online Star Wars Episode III Revenge of the Sith hit the screens right around the time first tried this hack When filed into the very first midnight screening of the movie at my local cineplex was particularly excited by the prospect of again hearing the voice of the galaxy s most dysfunctional father There s just something about James Earl Jones and the flange effect After all who hasn t looked into a mirror in a private moment and said T am your father a few times OK maybe you re not as big a Star Wars geek as am but if you are a closet Wookiee lover I ve got the perfect hack for you to use the next time you chat with fellow fans If you think spy movies are cooler than Star Wars movies you can also use this hack to make yourself sound like one of those disguised voice phone informants that sound a lot like well Darth Vader Gold Software s nifty voice changing tool Audio Voice Cloak lets you tweak your speaking voice adding pitch shifting EQ echo and other sound effects in real time Figure 2 2 If you have Windows you re in luck Mac folks see the si
44. Call Forwarding to YolP e Forwarding Condition Unconditional Call Forwarding Number 7711 End of Call on PSTN Digits bar m es Sw ce I ee ___VoIP2 DGE VoIP a C C C Fi a i oo000 ogon oo00 C C i C I c E Unconditional Uncanditonal Unconditional PSTN2 PSTN3 PSTN4 E B E l D000 pooo ooo0 o e Unconditional Unconditional Unconditional 7712 7713 7714 C enable digits ee disable On this page you can see the FXO channels that correspond to the VoIP channels They re called PSTN1 PSTN2 etc and they represent the two or four RJ 11 jacks on the gateway s back panel To get incoming calls from the attached phone line to be forwarded automatically to the SIP server on the Asterisk machine click the radio button under PSTN1 labeled Call Forwarding to VoIP Now when calls come into the Clipcomm from the phone company line they ll be answered automatically and the Clipcomm will attempt to route them through to the Asterisk server Don t forget to save this configuration If we wanted SIP calls outgoing from Asteriskto the Clipcomm to be forwarded to the PSTN we would need to enable Call Forwarding to PSTN for each VolP channel too Keep this in mind for the next hack 4 4 2 Configure an Asterisk SIP Peer for the Gateway Asterisk refers to SIP endpoints as SIP peers and it uses etc asterisk sip conf to
45. DSL Internet access from the phone company you might not be able to disconnect your phone company voice service without inadvertently severing the DSLconnection too Sometimes DSL runs on the same pair of wires as a traditional analog phone service If you have DSL it must be on a separate pair of wires from your voice phone line or this hack won t work and you will have disconnected your Internet service to boot Cable Internet subscribers can hack without this worry If you re attempting this hack in an apartment it might be a little tougher Your lease agreement might prohibit you from making wiring changes like this In some jurisdictions the phone company itself or building codes might prohibit this type of wiring hack If you re not sure call the phone company and ask that a lineman come out to disconnect the wiring Once disconnected from the demarc mark the pair of wires with a tag that reads Phone company Do not reconnect This will prevent a well meaning phone company service technician from reconnecting your line and frying your ATA Assuming your phone company disconnect was successful you can now connect the ATA into any RJ 11 modular jack on the premises This will let you hear the dial tone generated by the ATA and make VoIP calls through any phones that are connected to the other modular jacks throughoutyour home Most ATAs are designed to handle the power requirementsof only a phone or two so check with the
46. IS a Quality of Service QoS mechanism that uses a policy based approach to enforcing different classes of service on the same WAN It s not a bad idea to switchbiffServMode to on but don t expect this to increase the quality of your phone calls over the Internet as most Internet routers don t support the DiffServ standard Class of service is useful only in a controlled enterprise environment VlanMode Disable VlanlD 1 PcVlanID Z Virtual LANs or VLANs are a way Ethernet switches can segment traffic to create logically independent networks on the same equipment In most enterprise VolP scenarios there are separate VLANs for voice and data traffic These settings allow you to specify which VLAN ID to join the phone with And since the phone has a built in Ethernet switch for piggyback connection of a PC you can specify the PC s VLAN too That way even though phone and PC are connected on a single uplink cable they can still be on separate VLANs This functionality is common on most IP phones that double as Ethernet switches Remember that since we re talking aboutunidencom txt the VLANs Specified here will apply to all phones that use this TFTP server TftpAddress Lg ecg a dle Just in case the DHCP server ever crashes and the phone can t acquire a TFTP server address from it you can set the address of the TFTP server here TimeZone 6 EnableDST YES FnableSNTP YEo SntpServerIP Ug ae chy LO SntpRetrySsec LOU These settings con
47. J K L M N O P Q R S O U M wW BI D Z S518 ADSL board from Sangoma Technologies SAM Skype Answering Machine 2nd 3rd sample set of Asterisk configuration files sampling resolution for phone calls SaRP Say command Scheduled Tasks Windows SDP Session Description Protocol 2nd 3rd 4th 5th inspecting successful capabilities negotiation search engine security Asterisk global VolP trunks Skype network wireless LAN sendmail services Cisco shift pitch effect show application command signaling FXO FXS devices pulse signaling signaling protocols Skype Simple Mail Transfer Protocol SMTP SIP Session Initiation Protocol 2nd calling a party without using SIP gateway or gatekeeper configuration of Uniden IP phone 2nd 3rd 4th 5th controlling port numbers default port examining SIP packets with Ethereal 2nd 3rd 4th 5th 6th 7th 8th firewalls and inspecting SIP message structure 2nd 3rd packet sent through router with SIP translation enabled poor NAT traversal capabilities use by VolP clients X Lite configuration settings 2nd 3rd 4th SIP Express Router SIP peers configuring for media gateway connected to the Asterisk server downloaded from lib ommolketab ir downloaded from lib eommotketab ir SIP phone 2nd 3rd 4th 5th enabling IP phone to place calls via Asterisk 2nd 3rd testing Asterisk SIP phones SIP Proxy setting X Lite SIP Uniform Resource Indicator URI sip cont file
48. Set static IP 121 mask address downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Option name SPron Valid options Notes name as gateway 130 None Reads current IP Set gateway IP Same as Set static IP 131 address address Check DNS 160 Moe server Set DNS server 161 Same as Set static IP IP address address n Resets all of the user changeable settings Geer Keser PAG none to their defaults Use with caution Resets all of the available configuration options to their defaults Use with caution Factory reset 73738 None Remember to add a trailing for each option So for example to have theSipura read its current IP address you should enter and then 110 Another thing to remember when you are entering IF addresses for the device default gateway and DNS server use the key to represent periods So you d enter the IP address 10 1 1 50 as 10 1 1 50 5 5 2 Various Tweaks After you have the Sipura connected to your network and you Know its IP address you can get to its web interface If you re used to dealing with web interfaces for Network Address Translation NAT firewalls and the like the Sipura web interface shown in Figure 5 2 is probably unlike anything you ve seen Most people that use the web interfaces of small office home office SOHO NAT firewall router devices are shocked when they see the web interface on a Sipura Here will attempt to point out the most common and
49. Used to show code examples the contents of files console output as well as the names of variables commands and other code excerpts Constant width bold Used to show user input in code and to highlight portions of code typically new additions to olc code Constant width iCalic Used in code examples and tables to show sample text to be replaced with your own values Gray type Used to indicate a cross reference within the text You should pay special attention to notes set apart from the text with the following icons This is a tip suggestion or general note It contains useful supplementary information about the topic at hand This is a warning or note of caution often indicating that your money or your privacy might be at risk downloaded from lib ommolketab ir downloaded from lib ommotketab ir The thermometer icons found next to each hack indicate the relative complexity of the hack f beginner f moderate f expert downloaded from lib ommolketab ir downloaded from lib ommotketab ir Using Code Examples This book is here to help you get your job done In general you may use the code in this book in your programs and documentation You do not need to contact us for permission unless you re reproducing a significant portion of the code For example writing a program that uses several chunks of code from this book does not require permission Selling or distributing a CD ROM of examples from O Re
50. Via SIP Z 0 UDP 127 0 0 1 66557 branch z9hG4bK1116720069 From lt sip pingel27 0 0 1 TO SIPIL eLp Ias Contact lt sI pipingi l2 7a0 U 1 gt Call ID 0436a2258bedd7 4d8618e587446810c9 127 0 0 1 CSeq 102 OPTIONS User Agent sip _ping pl Date Saty 41 May 2005 0501709 PDT Allow ACK Content Length 0 The response from the remote host is as follows SIF 2 0 Z200 OK downloaded from lib ommolketab ir downloaded from lib ommotketab ir Via SIP 2 0 UDP 127 0 0 1 6655 branch z 9hG4bK1116720069 received 192 168 0 52 Prom lt g1ipiping0 l2 s00 1 gt To ete 122 100 I2 Call ID 0436a2258bedd74d8618e587446810c9 127 0 0 1 CSeq 102 OPTIONS Contacti lt s20 102Z0192 160 Us L2 U CU LIne AA User Agent snom200 3 56m Accept Language en Accept application sdp Allow INVITE ACK CANCEL BYE REFER OPTIONS NOTIFY SUBSCRIBE PRACK MESSAGE INFO Allow Events talk hold refer Supported timer 100rel replaces Content Length 0 Notice the plain text structure of aSIP message packet Both the request and the response contain very human readable headers This is no accident The body responsible for SIP the IETF has a history of advocating human readable protocols As a result SIP avoids machine friendly ASN 1 encodings such as those used by SIP s predecessors The response captured here is from a Snom 200 SIP hardphone The User Agent field indicates that it s running version 3 56m of Snom s fir
51. VolP dialing between TSPs using the I nternet downloaded from lib ommolketab ir downloaded from lib ommotketab ir Desired action Dialing shortcut Call an AXTel user from FWD 1 700 and the seven digit AXTel number Call a Vonage user from FWD 2431 and the full 11 digit Vonage PSTN number Sa AN ATE 0110393 and the six digit or five digit FWD number onage maa Via uee 0451 and the six digit FWD number or five digit FWD number Call a Packet8 user from FWD 898 1 and the full 11 digit Packet8 PSTN number ny W DICE PE USE O 1 700 900 0000 and the full 11 digit VoicePulse PSTN number Call Sn LA E UCERO 1 700 and the seven digit AXTel number VoicePulse Call an FWD user from 1 700 9 and the six digit FWD number or 1 700 99 and the five VoicePulse digit FWD number downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 3 Wire Your House Phones for VoIP HACK 3 You can use your home phone wiring to connect all your home phones to your VolP service If you re happy with your VoIP service you might want to consider replacing your existing land line telephone service with that of your new VoIP TSP This means you must provide adial tone to all of your analog phones using the ATA instead of aconnection from the phone company Your problem is that most ATAs have only a single analog phone connector limiting you to just one phone Radio Shack sells two wire phone splitters th
52. a directory For this hack you re going to use XML to create your own custom menus that access hidden features of the Cisco phone To make your menu appear on the phone you ll need to configure the phone to look for your custom menu file Cisco phones like most IP phones have aFlash storage device onboard that is checked and optionally updated at every startup During startup the phones contact a TFTP server and attempt tc download settings stored in files on the server Hack 80 By setting the services_url property on the phone s console to the URL of your menu you can configure the phone to load your custom meni upon its next reboot To set this property press the Settings button on the Cisco phone Then select Unlock Config and use the keypad to enter the password the default iScisco Next select the XML URL option followed by the Services URL option Enter the URL like the following example using the phone s keypad you enter slashes and colons by cycling through the and keys httpe lathama com services cisco Now when the phone boots it will attempt to load a menu for the Services button by accessing an XML file at the URL You can also set this URL in a configuration file on the TFTP server If you do neither the Services button will be rather useless as it doesn t default to any built in settings Edit the SIPDefaults cnf file on the TFTP server to set the Services URL Use an entry like this services_url http
53. a little bit of legacy technology FXO FXS interfacing to set up the connection The second gotcha is that you ll need a Windows PC sitting next to your trusty Asterisk server and that Windows PC will need to haveSkype running and an Internet Phone Wizard USB interface attached The final gotcha is that the Internet Phone Wizard must already have speed dial numbers associated with the members of your Skype buddy list Hack 40 You would need this anyway if you were going to use the Internet Phone Wizard for its intended purpose connecting a traditional analog phone to the Skype network via a Skype client on the USB host PC Connect a standard RJ 11 phone cord from the telephone jack on the Internet Phone Wizard to an FXO port on your Asterisk box This FXO port can be on a Digium TDM400P a Digium X100P or a Sangoma WANPIPE PCI card several examples for setting up the TDM400P are given inChapter 4 Configure the Zaptel channel for this FXO port as you normally would if you wanted to connect the Asterisk server to a standard phone line The standard phone line is going to be substituted by the connection from the Internet Phone Wizard Next you ll need to add your Internet Phone Wizard speed dial numbers to your Asterisk dial plan so that they ll be dialed via the Zaptel FXO channel Remember I m assuming you ve already set up your speed dial numbers on Skype using the Internet Phone Wizard so if you haven t flip back to Skype with Your Ho
54. a server that bridges those previous generation devices with Voice over IP The Asterisk software is maintained by Digium Inc http www digium com a manufacturer of many of the interface cards and VoIP gateway devices Asterisk supports You can certainly use other interface cards with Asterisk such as those manufactured by Sangoma and VoiceTronix These manufacturers provide drivers for Asterisk s Zaptel driver framework that allow Asterisk to use them 4 1 1 1 To FXO or to FXS That Is the Question To use traditional analog telephones and lines with an Asterisk server you ll need to understand the difference between FXO and FXS Their definitions are a source of some confusion even among telecom folks FXS foreign exchange station interfaces are used to connect telephones which are FXS devices FXS interfaces cause the Asterisk server to appear like the telephone company s central office switch when you plug in a phone FXO foreign exchange office interfaces on the other hand are used to make your Asterisk server appear like a telephone so that you can connect it to the central office switch So FXO interfaces connect your server to the phone company and FXS interfaces connect your server to analog phones Keep this distinction in mind as you work through the hacks in this chapter FXS and FXO interfaces are manufactured by many companies including Intel Digium Sangoma downloaded from lib ommolketab ir downloa
55. all They can call each other directly by way of an IP address You re about to make a direct P to IP call with a BudgeTone hardphone A hardphone is an IP phone that isn t a softphone It looks just like an ordinary business phone but plugs directly into an Ethernet local area network LAN The Grandstream BudgeTone 100 phone model has a Menu key an LCD display and two arrow keys that you use to navigate its configuration menu options DHCP IP Address Subnet Mask Router Address DNS Server Address TFTP Server Address Codec Selection Order SIP Server Address and Firmware Versions called Code Rel on the phone s screen When you get to the option you want you press the Menu key to select it and then enter the numeric data required for each option using the keypad Use this menu only to set up the IP address subnet mask and router default gateway address because you ll be doing the rest of the phone s configuration using its web interface To get the phone enabled for the next configuration step turn DHCPoff and assign an IP address subnet mask and router address You can perform more advanced configuration using the BudgeTone s built in web configuration tool When you use your web browser to access the IP address you assigned you ll be prompted to log in to the phone The default password is admin Then you ll be confronted with a big page of configuration options many of which are available only through this interface
56. already great application If your social life is in need of new features though you might need the input of something other than J yve Aside from being able to publish your status on your home page visitors to your page will be able to request a phone call a Skype or old fashioned phone call from you via a text message that is sent to your Skype client That way even if you aren t there to chat the message will be waiting on your screen when you return Jyve even provides the equivalent of a V Card contact record for your web site which J yve calls aQ Card It s a web page hosted by J yve that contains your Skype information and a form that visitors can use to text message you even if they don t have Skype Using a bit of JavaScript pasted onto your home page you can integrate your Q Card into your web site A V Card is a small file that contains contact information for a certain person like a virtual business card V Cards are used by email programs for the most part 3 6 1 Get Signed Up Does all this web enabled Skype goodness sound like fun I m sure it does but first things first To use J yve s service you ve got to create an account on the J yve server To do so surf over to http www jyve com and find the Get your Skype Card Here link on the main page Here you ll have to fill out a pretty standard membership form On the following page you ll be able to create and customize your Q Card You can preview your Q C
57. analog call from the PSTN and to connect it using the AX protocol to a remote server across the Internet Note that although this chapter is aboutlegacy circuit switched telephony we re using AX to get our feet wet with VoIP Plus the AX demo is so easy to run with Asterisk out of the box it isn t broken by broadband routers the way SIP often is that it s a great way to demonstrate how a VolP signaling protocol can beused with legacy signaling on the PBX downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 45 Forward Your Home Calls to Your Cell Phone HAGK 45 Using Asterisk you can create a simple call forwarder so calls to your home can follow you whenever you go Asterisk is a programmable platform in the same way that the Apache Web Server is There are many ways to program Asterisk but all of them connect in some way to the core of Asterisk s functionalityits so called dial plan The dial plan begins and usually ends in etc asterisk extensions conf Using the dial plan you can program how your softPBX should behave which phones should ring when different digits are dialed how long they should ring and what to do if nobody answers when they ring So it s actually pretty straightforward to program the dial plan to forward all incoming calls from a certain line to another phone number via a second line There are many uses for this including having your phone calls follow you wherever
58. and bandwidth preserving codecs like the G 729 are the biggest processor hogs of all If you have a great need to support highly compressed codecs on a lot of phones and you ll be attaching your Asterisk server to the PSTN or to legacy phones using Zaptel you ll need more speed more RAM and then more speed again To relieve the burden on a softPBX try offloading processor intensive tasks to dedicated hardware Try to maintain an all SIP all uLaw environment on your softPBX Let off board equipment like SIP to PSTN gateways such as the Clipcomm Hack 43 handle codec processing tasks to preserve Capacity on the softPBX 7 2 3 Select an OS and Harden It If you re building a Linux voice server there s not much point using an older Linux distribution So get a recent revision of Fedora Core burn it to CD ROMs four of them and install it keeping a few things in mind about all of those optional software packages the installer will prompt you about e The more optional packages you install the less room you ll have for things like voicemail and logfiles both of which are important in the world oftelephony right e The more optional packages you install like the XWindowSystem the more security risks you take Security is a good thing right e The more optional packages you install the less dedicated processing power your server will have for telephony purposes Are you starting to see a pattern here Once Fedora Core is
59. application like telephony from which humans have come to expect 100 reliability over many decades a dilapidated hodge podge of PC equipment sitting under a leaky roof just isn t going to work A desktop gone server isn t going to cut it either Fortunately you re about to find out how to build avoice server right This means selecting the right PC or commercial telephone PBX chassis connecting it to the right components choosing the right operating system and hardening it And that means following a principled philosophy ofstability high availability and compatibility 7 2 1 The Three Things That Matter Most in Telephony When building a server for an enterprise telephone system you should keepthree areas of focus in mind Here they are in my order of importance Stability The predictable reliable operation of the server Downtime should be nonexistent and responsiveness should be instant High availability downloaded from lib ommolketab ir downloaded from lib ommotketab ir The server and network must be adequate to host real time applications without a noticeable impact when server resources are shared among many users The server must also be able to Survive a hard disk or power supply failure without interruption to the hosted application Compatibility The server OS and installed voice services must support well known standards to be adaptable to changes in the business such as growth strategic partn
60. as the Internet These technologies include protocols hardware and software standards and computer programs VoIP is employed in telephony applications from analog phones to next generation IP phones and wireless headsets and in desktory voice chat services from web based party line chat services like Yahoo Chat to the well known Skype desktop voice calling service VoIP has become an important technology that is integrating pervasively into the popular culture It is employed daily to drive new engines of commerceeverything from business class Vol P powered calling services to simple desktop chat tools such as Apple s iChat Other high profile companies like eBay Microsoft Google and AT amp T offer applications and services that utilize VolP too These big companies have recognized that the popular culture is moving to VoIP services en masse even as the telecom industry is being set on its ear by scrappy young VoIP startups like Vonage Packet8 and SpeakEasy net VoIP services deliver telephony applications less expensively than the old phone companies can hope to This is because VolP Is free of the continually burdensome legacy technology investment the old phone companies must make to keep the old global phone network running VoIP is also free of the endless government regulations and tariffs imposed upon the old phone companies In a nutshell the way society looks at the voice network has changed VoIP is the enabler of the chan
61. as your inevitable widget addiction grows When the wizard is complete Konfabulator Yahoo Widgets will automatically launch its default set of widgets Now to try out vonageGauge 2 3 2 Installing the Vonage Widget Download the vonageGauge widget from http www widgetgallery com view php widget 36334 and Save it in a temporary folder or in the standard spot where you put downloaded files Unzip the download or mount it by double clicking it if on a Mac and copy the enclosed widget file to the widget folder you selected during installation Mac users can launch the widget with no further issues Windows users however must do some manual configuration due to lack of SSL support for the curl web utility in the Windows version of Konfabulator Don t worry though This is hardly a painful thing to fix You need this because Vonage s web site requires as it should SSL encryption to be employed when accessing account information To rectify the matter download the most recent version ofcurl from its web site http curl haxx se latest cgi curl win32 ssl From the downloaded zip file note the filescurl exe and curl ca bundle crt as you ll need them in a moment Then grab the needed SSL libraries ssleay32 dll and libeay32 cll from http www2 psy ug edu au ftp Crypto The libraries will be located in abinaries directory on one of the FTP mirrors listed here The specific file you need to download will have a name lik amp SSLeay X
62. be sure because all machines are different One thing to note is that by default the CF writing utility will refuse to write to disks larger than 800 MB This will prevent you from accidentally overwriting your hard disk which should be much larger than 800 MB Once you have selected your disk follow the remaining prompts until you see the progress counter write the entire image and the words Press any key to continue appear You can now Safely remove the USB adapter Although your fresh new CF is now ready to be used would like you to take a look through the AstLinux User Guide which ts also available from the AstLinux Programs group Because AstLinux was created from scratch it bears little resemblance to any existing distributions and the User Guide attempts to familiarize the user with its features and configuration 7 9 5 Install from Linux As stated earlier go to the Downloads section ofhttp www astlinux org Here you should find compressed gzipped versions of the AstLinux images Download the image you would like and save it to a place on your hard drive Connect your USB CF adapter with CF inserted and look in var log messages to see what device it was assigned If you don t have any other USB or SCSI disks attached to the system it should be located in dev sda That is what will assume but make sure tc note whether your CompactFlash card is located in a different device To verify the location of your CF card ty
63. because you ll have many techniques to observe VoIP packet flow including the packet logging I ve just described 6 14 3 See Also e Graph Latency and Jitter Hack 75 e Peek Inside of SIP Packets Hack 81 e Sniff Out Jittery Calls with Ethereal Hack 83 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 85 Secretly Record VoIP Calls E G 711 uLaw is the most common codec used in enterprise VolP but it s far from secure The G 711 codec is the de facto standard for voice encoding on VolP networks because the earliest VolP gear and software didn t have enough processor power for real time transcoding from one code to another This means that if a call were to originate on the PSTN and terminate on a VolP device the entire call would have to be in the same codec The codec that s always been used on the North American PSTN is G 711 uLaw Unfortunately even as Cisco CallManagerarguably the world s first enterprise VolP platformbecame popular it was painfully clear that running G 711 uLaw across the Internet was a very insecure thing to do That s because the RTP packaging convention used by most VolP systems doesn t encrypt the media stream of a call making it the aural equivalent of clear text ripe for outside snooping Usingcpdump and a copy of vomit Voice over Misconfigured Internet Telephones you can actually capture phone calls midstream and convert them into WAV files How s
64. become the dominant multimedia communication protocol used by an overwhelming majority of VolP service providers and professional phone system vendors Aside from voice you car use SIP components to signal video and instant messaging conversations too I ll concentrate on SIF as it applies to voice though There are two kinds of SIP VoIP clients those that allow you to connect to a VolP system of your choice and those that are programmed for use only with a certain provider SIP supporting VolP client software includes products such as Yahoo Messenger Apple iChat sipXphone Firefly GnoPhone Gizmo Project and lots of others 2 4 1 3 IAX a really cool VoIP protocol Inter Asterisk Exchange protocol or AX pronounced eex is used by a growing number of VoIP client programs and service providers The coolest thing about AX is that it s firewall proof In situations where SIP and H 323 are rendered inoperable by NAT firewalls like your home broadband router AX shines The only problem is finding a service provider with which to use IAX visit http www teliax com to learn about one that offers an AX based VoIP telephone service AXPhone and Firefly use IAX 2 4 2 Understand VoIP Client Features You ultimately will decide on a VolP client based on features and compatibility While one VoIP client might support the protocol you needsay SIPit might not support the features you need iChat and X Lite are both SIP software but
65. com info php id 14067 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 12 Grow Your Social Network with Gizmo HAGK 12 If you love Skype but hate the fact that it isn t open and standards based you ll be right at home with Gizmo Gizmo Project sponsored by SIPphone Inc http www sipphone com seeks to create a free peer to peer softophone with instant messaging a la Skype but without the proprietary hindrances of Skype In this regard Gizmo does an excellent job Its features are the same on Mac Windows and Linux toowhich means no more waiting two months for Windows only features to show up in the Mac and Linux clients something Skype users are accustomed to Another cool plus that Gizmo brings to the table is free voicemail something Skype has yet to offer To get started with Gizmo hook up your headset and microphone and download and install the clien for your platform from http www gizmoproject com Launch the Gizmo app and register for your Gizmo name from the login screen This name is both your login ID and the name that other Gizmo callers will use to call you Once you re logged in set up your user profile as inFigure 2 3 Figure 2 3 Gizmo s profile dialog downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Gizmo Project My Profile Wallingford Oh United States 2 Engish Eril tw oreily mac com Your emal addre
66. default exten gt s 1 Dial SIP 100 30 exten gt s 2 Voicemail 100 exten gt s 3 Hangup So as of right now or at least after you reboot your Asterisk box or load the kernel modules manually incoming phone calls to the connected phone lines will ring on the SIP phone configured as SIP peer 100 For a refresher on SIP peers refer back to Connect a Phone Line Using an FXO Gateway Hack 43 4 10 2 Set Up Station to Station Calls If you d like your SIP phones to be able to call each other be sure to add the following extension to the default context exten gt _1XX 1 Dial SIP S EXTEN 30 exten gt _1XX 2 Voicemail S EXTEN exten gt _1XX 3 Hangup downloaded from lib ommolketab ir downloaded from lib ommotketab ir The _1xx pattern matches any phone numbers dialed that are three digits long and begin with 1 It deals with them by attempting to ring the SIP peer that is registered with a user ID that matches the extension number dialed and then sends them to the appropriate voicemail box after 30 seconds if the SIP peer doesn t answer 4 10 3 Set Up Outgoing Calls Now you ve got to make it so that any connected phones can place calls using the four lines that you ve just hooked up to the installed FXO modules This is accomplished by a special pattern matching extension in those phones contexts For SIP phones this is established insip conf Let s say that a SIP phone s context IS private phon
67. device to mimic 911 telephony service providers VoIP networks building PSTN gateway 2nd creating premium class of service 2nd 3rd 4th 5th intercepting and recording a VoIP call 2nd 3rd logging and recording VoIP streams 2nd 3rd logging VoIP traffic 2nd 3rd 4th 5th monitoring VoIP devices 2nd peeking inside SIP packets 2nd secretly recording calls vomit Voice over Misconfigured Internet Telephones 2nd Vonage 911 service calls to Outlook contacts with one click vonageGauge widget downloaded from lib eommolketab ir downloaded from lib ommotketab ir X PRO commerical counterpart of X Lite vonageGauge widget Vonage using downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H OY J K L M N O P Q R S O U M W RI D 4 WAV files using as custom Skype ringtones wcfxo c file wcfxs driver wctdm driver wctdm module weather forecast web browsers 2nd web sites simplifying communication for visitors tracking visits from Jyve users web based call management tools BroadVoice web based GUI webcams for Skype real time video Weird Solutions WEP Wireless Encryption Protocol widgets 2nd 3rd Gotta Go installing Yahoo Widgets vonageGauge WiFi enabled Pocket PC Wildcard TDM400 Windows Mobile 2003 Windows systems call handling software Messenger IM software podcasting tools Skype 2nd Skype Answering Machine SA
68. digit PIN to verify access when accessing the unit The device includes instructions on how to do this 5 used a modular cord to connect from the RJ 11 port to the Asterisk FXS port For the Asterisk system downloaded from lib ommolketab ir downloaded from lib ommotketab ir 1 assigned the FXS port an extension number using an AsteriskDial Command 2 set the SIP peer for the IP phone to pass digits in band tmfmode inband 3 set the SIP peer for the IP phone to use the G 711 uLaw codec Here s the bit from etc asterisk extensions conf that you would use assuming the FXS port iS zap 1 exten gt 100 1 Dial Zap 1 Here s the bit from etc asterisk sip conf 200 username 200 secret 200 type friend dtmfmode inband disallow all allow ulaw Verifying if the setup is correct requires a few test calls Use the Polycom IP phone to call the controller Go off hook on the IP phone and dial the extension number assigned to the controller 10 in this example The line will ring for about 15 seconds before the controller will answer When the controller answers you will hear three beeps Enter your PIN code and then a second set of three beeps will confirm that you entered the proper PIN code After the second set of three beeps enter the module number you want to control followed by or to turn the module on or off respectively mentioned that there are a few ways to integrate X10 controls with Asteri
69. does not match what was transmitted If you re trying to use SIP behind a firewall and you aren t aware that that well meaning firewall has altered your transmissions it will be next to impossible to get around the fundamental problem of SI using other more generally accepted solutions like STUN more on that later Here is an example c a SIP packet sent through a router with SIP translation enabled sip _ping pl v proxy01 sipphone com grep Via Via SIP 2 0 UDP 127 0 0 1 6655 branch z9hG4bK1116737081 Viet SIPs 2207UDP 10 1 1 5166557 branch 79nG4bh1l116 737061 Note that since the packet was rewritten the remote host s RFC 3581 implementation did not insert the received and rport fields However when the packet returned a different IP address was inserted in the Via header This can be especially troublesome on hosts with multiple IP addresses a the router rewriting the headers might be assuming that all responses should go to one host This can be a good thing when you have a single SIP device on the private network but it s a bad thing when you ve got a whole building full of SIP phones If the router is rewriting SIP headers arbitrarily it won t be possible for all of these phones to receive their SIP responses The sip ping pl script takes two additional arguments s for source and p for port that you can use to explore NAT traversal These are the source IP address and the source port to be placed in the Via and Contact head
70. effectiveness To buy afour line business phone system new is usually more expensive than equipping an Asterisk box like you ve done in this hack Plus you ve got access to Asterisk s programmable dial plan and application programming interfaces theAsterisk Gateway Interface and Asterisk Manager API giving you a metric ton more capabilities than a low end commercial PBX downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 50 Master Music on Hold 50 Can you hold on a minute the operator asks Suddenly you re listening to Frank Sinatra singing New York New York Before you know it you re tapping your finger and the wait doesn t seem so bad Few things are more dreaded among telephony end users than the short yet foreboding phrase Please hold Perhaps what bothers folks is that they never know quite how long they re going to be on hold or maybe it s the notion that they re going to have to re explain themselves to a whole new person who winds up on the line after the hold time is over with Fortunately music on hold makes that wait time a little bit more tolerable Asterisk gets its music on hold sound signals from MP3 files that are decoded and piped toAsterisk by one of two supported MP3 players Mpg123 and MPEG Audio Decoder MAD In this hack I m going to use MAD because there are some well documented security issues with Mpg123 that have yet to be dealt with Of course if you d like to u
71. em Tent _ lt Recording Voice Message Settings Mae recording time for new messages ra Mu Aeceved Yorce Messages Folder Settings Location C Documents and Settings twvallingford a _ Limit folder size to 0 ME Current 0 KB Moore _ Mute volume when answering calls Application Settings Automatically shart wher logging into Windows Restore SAM when new Volce Message is saved Show Balloon Info Messages Interface Language English English w Factor Defaults Click here if you need to restore all options to ther factory defaults 3 11 1 Let Windows Automatically Rotate Your Greetings If you want to use a different greeting depending on the time of day similar to the daily rotation used in Greet Callers Differently Each Day Hack 17 you can use a batch file combined with Windows Scheduled Tasks to rotate the greetings at predetermined intervals In this case let s do an a m p m rotation meaning we ll need just two greetings You can set up a single Windows batch file to swap two previously recorded greeting files stored in a greetings directory cd greetings copy Y greeting wav greeting_tmp wav copy Y greeting_bak wav greeting wav copy Y greeting_tmp wav greeting_bak wav del greeting_tmp wav Now copy your p m greeting to greeting wav and we ll schedule this to run beginning at noon using Scheduled Tasks available from Start All Programs Accessories System
72. enjoy being contacted by anxious telephone pitchmen promising a lower interest rate or offering a great deal on term life insurance while you re just sitting down to dinner skip this hack Using techniques similar to those in Selectively Forward Calls Hack 46 it s possible to discern between phone numbers that supply a caller ID and those that don t This is different from merely identifying a certain caller ID number and then handling it What we re doing here is shovelingll unidentified calls into a certain action If you like you can even have your phone server handle these calls without interrupting you putting a decisive end to those annoying dinner hour calls from Private or Unknown Using a great little feature in Asterisk the PrivacyManager Command we can fight fire with fire This dial plan command screens calls as described earlier identifying the caller ID or forcing the calling party to enter a calle ID if none is provided at the outset of the call Best of all everything can happen without your phone ever ringing saving you from the aggravation of a sales pitch when you re trying to enjoy a filet mignon Consider the following from the default context in extensions conf exten gt s 1 PrivacyManager exten gt s 2 Dial Zap 2 30 exten gt s 3 Hangup exten gt s 102 Hangup The first priority of this extension contains theprivacyManager command which prompts the user to enter his 10 digit telephone
73. establish settings for them everything from usernames and passwords to basic audio preferences Our objective here is to establish a SIP peer configuration for the media gateway we ve just configured So using your favorite text editor add the following to yoursip conf file 7711 callerid Outside Line lt 200 gt Canreinvite no context default downloaded from lib ommolketab ir downloaded from lib ommoltketab ir dtmfmode rfco2833 host dynamic port 5060 type friend username 711 Notice how the bracketed heading and the username setting of 7711 match the media gateway s User ID and Authentication ID settings respectively 4 4 3 Make Asterisk Answer Automatically Now Save sip conf and open up etc asterisk extensions conf This file tells Asterisk what to do whenever a user dials a phone number It contains the dial plan that guides the system wide call handling functionality of the Asterisk server In the default section of the file comment everything out and add these lines exten gt s l Answer exten gt s 2 Playback abandon all hope exten gt s 3 Hangup Save the file and launch or relaunch Asterisk asterisk rx reload Now calling the phone number of the line connected to the media gateway will result in the call bein answered by the Asterisk server You ll hear a voice message and then the call will be hung up downloaded from lib ommolketab ir downloaded from lib om
74. eth0 d 10 1 1 50 dport 5061 j ACCEPT iptables t nat A PREROUTING p tcp i eth0 d 201 101 1 1 dport 5000 j DNAT to 10 1 1 50 5000 iptables A FORWARD p tcp i eth0 d 10 1 1 50 dport 5000 j ACCEPT iptables t nat A PREROUTING p tcp i eth0 d 201 101 1 1 dport 5001 j DNAT to 10 1 1 50 5001 iptables A FORWARD p tcp i eth0 d 10 1 1 50 dport 5001 j ACCEPT Now any inbound SIP ports 5060 and 5061 or RTP ports 5000 and 5001 in this example traffic will be forwarded directly to host10 1 1 50 where your SIP softphone is running Ordinarily you wouldn t need to forward the SIP ports for outbound calls to work STUN takes care of that as long a the SIP client can use STUN For a list of VolP related port numbers refer to Log VolP Traffic Hack 84 Next up is configuration of the STUN client in the softohone Launch X Lite and configure it as you normally would then add the private address of yourNAT machine which is now running a STUN server as well to the Primary STUN Server entry of X Lite s preferences dialog Now call somebody outside your private network to see if it works if that person is also behind a NAT it might not downloaded from lib ommolketab ir downloaded from lib ommotketab ir downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Hack 77 Shape Network Traffic to Improve Quality of Service 77 There s a reason why t
75. for any phone field So if you want to Skype somebody from your address book using their Skype name rather than their PSTN phone number just enter their Skype name into a phone number field in the Address Book That will work just fine downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 31 Enable Site Visitors to Skype You HACK 31 Want to make it easier for folks to Skype you Put a Skype Me link on your web site or blog and your visitors will chime in Chances are if you have a web site you have more than enough HTML skills to pull off this simple hack If visitors to your web site have Skype installed on their computers they already have the Callto URI handler set upin other words they can click a link on your page to call you using Skype Here s an example of such a link at the upper left corner of the page in Figure 3 4 Figure 3 4 A Skype Me link embedded in a web page The nifty Skype Me graphic is provided by the good folks at Skype so all you need to do to create this link is plug in a bit of HTML with your Skype username like this lt a href callto Your_Skype_Name gt lt img src http goodies skype com graphics skypeme_btn_small_blue gif downloaded from lib ommolketab ir downloaded from lib ommotketab ir border 0 gt lt ae You ll need to replace your_ Skype_Name with your own Skype namehopefully that was obvious to you The image I ve used in Fig
76. immediately regardless of whether it matches the dial plan you can add to the dial string Thus in the previous example 12345678 will send 12345678 to the remote server immediately even though it does not match the dial plan string It s probably worth pointing out that there is a limit to how long a dial plan string can be A dial plan string has a maximum length of 2 047 characters On the Sipura SPA 3000 you can have eight dial plan strings for the Public Switched Telephone Network PSTN line The limitation for those is 511 characters each 5 5 4 Advanced Dial Plan Examples Here are some more advanced dial plan examples SLA lOOve 1 2 16 0 poe soUGL This is a slight modification of the dial plan string from Build a Bat Phone Hack 63 This string will call extension 1002 on the Sipura at 192 168 0 22 on port 5061 However it will do this only if you dial 111 This is a very inexpensive way to set up a PBX with no SIP server at all You could take several Sipuras with static IP addresses and assign them extensions You could even includean SPA 3000 for single line POTS termination origination Here is a more complete version of the preceding code Ce A de eg ie alt So A se ee ee de SL LLS ado Uslar unl If you had this same dial plan on every device you would be able to call between them simply by dialinc 111 112 and 113 This example is another slight modification Essentially here we are adding 1847 to any number th
77. in your address book with two mouse clicks Though the Skype API provides some power tools for Skype hackers on the Mac you can do quite a bit with Skype using plain old AppleScript This is great since not all hackers prefer the same tools or if you re like me you ve stubbornly refused to understand C In Skype Your Outlook Contacts Hack 29 you saw a few different ways to tie Windows based Outlook to Skype You can do the Same thing for the Mac OS X Address Book with a little bit of assistance from trusty old AppleScript Thanks to the simple but awesome script shown in the next section it s a snap to call any of your Mac OS X Address Book contacts using Skype 3 4 1 The Code For starters you ll want to key the following script into the AppleScript Editor You can find the AppleScript Editor in your Mac s Applications folder using terms from application Address Book on action property return phone end action property on action title for p with e return Call with Skype end action title on should enable action for p with e return true end should enable action on perform action for p with e set x to value of e as string if character 1 of x is not then set x to 41 amp x end if set SRYPREurl to callto amp x downloaded from lib ommolketab ir downloaded from lib eommolketab ir tell application Skype get URL SKYPEurl activate end tell return true end perform action end us
78. in server mode execute this command asterisk r Once the Asterisk client is connected to the Asterisk server you can use Asterisk scommand line interface to issue queries and commands about the telephony server These include listing calls in progress listing used and unused channels and stopping the Asterisk server You can shut down the server using one of several Asterisk CLI commands e Restart now e Restart when convenient e Stop now e Stop when convenient The restart commands stop and then restart the Asterisk server process which can be helpful in Situations where the server s configuration has changed significantly and needs to be restarted The stop commands just shut down the Asterisk server process You ll have to execute the Asterisk program in server mode to get it running again downloaded from lib ommolketab ir downloaded from lib ommotketab ir The now and when convenient arguments tell Asterisk how quickly to shut down or restart If you want to interrupt the current calls and tasks In progress on the server now is appropriate If you want Asterisk to wait until all the calls and tasks are finished and there is no call activity at all wher convenient is appropriate Generally especially if you re planning to have any callers besides yourself on the system get in the habit of using when convenient All of these commands ultimately shut down Asterisk If you make a configuration chang
79. into Google and you ll have the whole phone numberand your pizzain no time Hungry Howie s Lakewood 216 By giving Google the name of the pizza place the city it s in and the area code you expect its phone number to have the first Google result is almost always the right oneand the entire phone number you re looking for usually shows up in the short synopsis on Google s results page so you don t even need another click Hey when you re craving pizza time is of the essence right 2 15 3 Telephone Privacy Check While you re perusing Google s phone number department you might want to see if your phone number is in the Google search index If it is your privacy could be in question Try Googling your phone number with and without punctuation and see what results come back You might even turn up somebody who has published your phone number without your authorization This isn t entirely likely but remember a couple of years ago when a list of thousands of valid credit card numbers made it into the Google index so it s not out of the question 2 15 4 Research VoIP History on Google Groups Probably the best historical newsgroup search tool on the Web Google Groups lets you go back in time to search for public correspondence about all kinds of topics including Voice over IP By surfing to Google Groups http groups google com was able to find that the first mention of VoIP on record occurred in early 1996 and that the first
80. machine Of course if you ve already compiled and installed Asterisk Hack 41 installing Festival will seem fast While you re waiting for the compile to complete let me give you a quickFestival crash course Festival has an interactive mode where you can issuespeech commands as well as an execute and exit mode where you can pass instructions to it from the Unix shell Simply executingfestival ina Shell will put you in interactive mode where you can interact with the speech synthesizer festival gt SayText Hello world festival gt tts text file txt The SayText Command simply causes Festival to speak the quoted text using your PC s sound card and the tts command speaks the contents of the text file indicated By the way if you d like to quit interactive mode and return to the shell hit Ctrl D When you execute Festival from the command line you ve got some cool functionality at your disposal Executing with pipe causes Festival to take commands from standard input Recent builds of Festival also include the text2wave application which generates Wave format sound files from text input 7 6 1 The Hack For Asterisk to Support text to speech via Festival you have two approaches The first is to use Asterisk s built in Festival command which is a standard part of the Asterisk distribution when patched for Festival as described earlier To get to this command you might need to recompile downloaded from lib ommo
81. modem that you havecomplete control over When you use it with the PPPoE client software from Roaring Penguin you can eliminate your SpeedStream kludge of a modem and gain the enhanced speed logging and feature set provided by the 518 highly recommend it to anyone already using ADSL and Linux Plus you don t have to cap your link speed at 85 as the queuing on the S518 can be controlled from Linux Set this to the upstream speed of your connection Use the Internet speed test results from before and again subtract 15 from your results The best way to determine this number isby testing testing testing UPLINK 550 downloaded from lib ommolketab ir downloaded from lib ommotketab ir This is your external network deviceprobably etho or etH1 However if you are using the Sangoma S518 mentioned in the sidebar VolP QoS or dial upin which case you must really love torturing yourself this will probably bepppo DEV ethl This is a list of ports separated by spaces to be added to the VoIP class This class of traffic is given highest possible priority 4569 is the port for AX2 Asterisk s native Inter Asterisk Exchange lAX protocol Do notdo notput 5060 here ever more on this later VOIPPORTS 4569 This is a list of ports to be given interactive priority This is the next highest level of priority and by default it includes two common ports used for SIP signaling Please note that with common SIP devices sign
82. most IP phone configuration files are regular text files that look like a Unix conf file or an old fashioned Windows ini file Second most IP phones support two methods of configuration by TFTP default and phone specific Default configuration files apply to all phones of a specific make that connect to the TFTP server Phone specific configuration files apply only to an individual IP phone These tend to be denoted by the IP phone s MAC hardware address in the filename of the configuration file So if you were to browse through the contents of a typical TFTP server on a VoIP network you d see at least one filename without a MAC address the default configuration and a handful of files that have MAC addresses in their filenames denoting them as phone specific configurations Most IP phones will revert to the default configuration automatically if no phone specific file exists on the TFTP server Other phones will include settings from the default configuration file if theyaren t mentioned in the phone specific file Uniden s IP phones generally follow this convention Take a look at this sample tftpboot directory uniden00e01102ffb7 txt uniden00e01103000a txt unidenBase txt unident0e0ll02ffc3 txt unidenO0OeO01LLO3001lc txt unidencom txt uniden00e01102ffca txt uniden00e01103001 txt This directory contains several phone specific configuration files the unidenXxXXXXXXXXXXX txt ones and a default configuration file unidencom txt un
83. network traffic WAN interface for quality of Internet connection to phone network Gizmo quality of VolP calls 2nd downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H N J K L M N O P Q R S O U M wW BI D Z radio broadcasting RAID 5 disk array rebooting the ATA Record It Gizmo Project recorder applications recorder switches recording a videoconference audio from VolP conversations on Mac OS X recording calls Asterisk 2nd audio chat on your Mac on your Windows PC Recording Industry Association of America RIAA recording software redundant power supplies Register setting X Lite registrar registration 2nd Rejection Hotline remote phone jukebox requests residential style responses to SIP methods restart commands Asterisk reverb effect SoX reversal effect Sox RFC 3581 ring script pausing iTunes for calls ringtones for Grandstream phone Rogue Amoeba s Audio Hijack root user account rotary dial phones for bat phone RRDtool 2nd 3rd RSVP Resource Reservation Protocol 2nd checking support with pathping RTP Real time Transport Protocol control of port numbers downloaded from lib ommolketab ir downloaded from lib ommoltketab ir NAT and downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H N
84. not from the phone s keypad metu For this project the only settings we re concerned with are the codec selection ones Configure the first highest priority codec to be PCMU if you re in North America or PCMA if you re elsewhere in the world That s all we re going to cover about codecs for now After you apply any configuration changes you need to power cycle the BudgeTone Some IP phones offer a Telnet interface rather than a web based one To use these tools you must connect to the phone with a Telnet client rather than a web browser In any event once the network configuration is set on the IP phone ping its address from another host on the same network subnet downloaded from lib ommolketab ir downloaded from lib ommotketab ir to make sure it s speaking Transmission Control Protocol Internet Protocol TCP IP Many VoIP devices need access to a time clock The network time protocol NTP server we ve chosen is time nist gov More NTP servers are available from the list at http www nist gov 5 3 1 Make an IP to IP Phone Call With both IP phones connected to the same Ethernet switch or directly connected to each other using a crossover patch cable make a note of the IP address you ve established for each In this example we ll use 10 1 1 103 for the receiver and 10 1 1 104 for the caller If you have your phones configured for DHCP give them this static configuration instead The BudgeTone can place IP t
85. number if no caller ID signals have been sent on the channel to identify the caller If the caller doesn t enter his phone number he gets dumped to priority 102 100 plus the current priority where the call is disconnected using theHangup command When telemarketers call you pretend you re Scarface brandishing your Privacy Manager and saying Say hello to my little friend If the caller does successfully enter his 10 digit phone number the dial plan proceeds to the next priority In the previous example aDial command rings a phone connected to a Zaptel card that s what s referenced by zap 2 for 30 seconds before giving up and disconnecting the call if nobody downloaded from lib ommolketab ir downloaded from lib ommoltketab ir answers 4 9 1 Hacking the Hack You can combine this hack and Selectively Forward Calls Hack 46 to maintain privacy and to pick and choose functionality based on caller Dfor instance forwarding calls based on who s calling And as shown here you can make sure you know who s calling with your new friend Mr Privacy incoming exten gt s 1 PrivacyManager exten gt s 2 Gotolf S CALLERIDNUM 3138853352 6 3 exten gt s 3 GotolIf f S CALLERIDNUM 3132981848 76 4 exten gt s 4 Playback carried away by monkeys exten gt s 5 Hangup exten gt s 6 Dial S MYCELLPHONE 30 exten gt s Playback carried away by monkeys exten gt s 8 Hangup exten
86. ommolketab ir downloaded from lib ommotketab ir 1 2 VoIP Based Phone Service Providers The Golden Age of broadband began with catchphrases like surf the Web five times faster and with promises of ultra fast music downloads But in the late 1990s few would have predicted that Vol P based telephony would be one of the biggest beneficiaries of once hyped broadband technologies like cable Internet and DSL Sure web surfing at the speed of light and downloading music are greatbu can they save you money Legally VolP telephony canand does For roughly half the cost of a traditional phone line you can subscribe to a VoIP telephony service provider rather than to a phone company You ll get a standard phone number that people from the non VolIP world can use to call youand you won t have to pay 5 a month extra for voicemail and caller ID This chapter has a handful of hacks that will show you how to maximize yourbroadband voice service So if you subscribe to a VoIP service provider you re ready to hack If not what are you waiting for Get Connected Hack 1 describes some VolIP based phone providers that you should evaluate as you prepare to dive into VolP Hacks downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 1 Get Connected iE If you ve got broadband you re already using the I nternet for data communication Wouldn t it be great to use it for telephone calls too Internet t
87. ommolketab ir downloaded from lib ommotketab ir 4 1 Hacks 4158 Introduction The Linux domain of free software is a land flowing like milk and honey with telephony hacksthe hackers Promised Land so to speak Of course many of these hacks translate to BSD and even to Mac OS X since they re cast from a similar Unix mold In this chapter I ll cover Asteriskthe open source telephony server designed originally for Linux but now available for Mac OS X and BSD Asterisk is a workhorse a flexible open system that s the telephony equivalent of Apache the world s most widespread web server Because of its modularity and flexibility Asterisk is as much a platform as is Linux It s sort of become the cornerstone of Linux based telephony thanks to a vibrant developer community and a sound open source foundation 4 1 1 Getting Telephony Devices Connected to Asterisk Besides implementing pure VolIP voice calls over packet networks like the Internet or your Internet Protocol local area network IP LAN Asterisk can also handle legacy telephone technologies such as analog phones and phone lines T1 lines and various kinds of legacy signaling methods A large and growing selection of PC expansion PCI cards are available that facilitate connecting analog phones and phone lines to an Asterisk server So if you want you can build an Asterisk server that doesn t use VoIP at alljust legacy technologies like analog phones Or you can build
88. ommolketab ir downloaded from lib ommotketab ir foldera text file with the following contents Description Asterisk Provides Asterisk Uses Disks Teo 2 Once those two files are in place give your OS X machine a reboot and your Mac PBX is built Now all you need is somebody to call downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 94 Monitor Asterisk from Your Perl Scripts HACK 94 If you ve used Linux or FreeBSD or Mac OS X or Solaris for longer than an hour chances are good you ve used Perl Now use Perl to monitor and control your Asterisk PBX The Perl module of choice for Asterisk is appropriately calledasterisk perl It provides connections between the Asterisk Gateway Interface AGI and the venerable scripting language named after a misspelled maritime phenomenon It also links Perl with Asterisk s Manager interface a socket application programming interface API that lets you control and monitor Asterisk by sending messages to it on a TCP port5038 to be exact For ad hoc interaction with the Asterisk Manager you can telnet to that port on your Asterisk serveril the manager is enabled that is To ensure Asterisk Manager is indeed running and able to respond tc your requests so that your Perl programs will actually do something once asterisk perl is installed you need to pay a visit to etc asterisk manager conf Make it look roughly like t
89. order cause problemsbig problemsfor an ongoing call These problems can cause the voices on the call to sound robotic to cut in and out or to go silent altogether Most of the packet drop problems you ll encounter while VolPing will be the fault of your bandwidth limited ISP connectionthe link from the ISP s network to your broadband router If you re downloading songs to your iPod surfing the O Reilly Network and patching yourWorld of Warcraft client all at once you won t have enough bandwidth left over to support a VoIP call but there s a wa to curb all those applications thirst for bandwidth so that you can still Vol P successfully Read on To maximize call quality the network connection carrying VolP media packets must be as reliable and consistent as possible The data link to the ISP should treat all voice media traffic withhigh priority That is a VolP packet gets handled first as it is more important than another packetsay for your BitTorrent upload If the data link is swamped and is out of capacity to carry any more data less important packets are discarded before more important ones The net resultfor high priority services like voiceis better Quality of Service or QoS Several standards exist to ensure that QoS can occur in a broadband VoIP setup chief among them Type of Service ToS and 802 1p If your broadband router is relatively new it might support these standardsso enabling packet prioritization is just a matt
90. parts of a SIP network Registrar A SIP server that keeps track of SIP clients by tracking the IP addresses where they re located and the usernames associated with each registration a directory of active SIP clients if you will Proxy A server that relays SIP messages and media streams between disparate networks Client A user agent that uses SIP for telephony text messaging voice chat or some other media applications 7 14 1 sipX s Requirements SipX runs on Linux There s presently some limited support for BSD but Windows and Mac users at downloaded from lib ommolketab ir downloaded from lib ommotketab ir least for the moment are out of luck An installer is available for Fedora Core 3 a distribution of Linux put forth by Red Hat In fact Fedora Core 3 is an ideal environment for sipX 256 MB of RAM and 500 MB of available disk space are plenty for setting up a sipX test lab When setting up Linux for sipX be sure to install thePostgreSQL database as well as the Apache Web Server both of which sipX utilizes 7 14 2 Install sipXpbx Get logged on to your target machine as root I m going to assume that you re running Fedora Core 3 For other distributions such as Fedora Core 2 and Gentoo Linux see the official sipX compatibility list at http www sipfoundry org First download and run the sipX Fedora Core 3 install script wget http www sipfoundry org pub sipX sipXpbx 2 8 1 fc3 sh sh si
91. power inline recorder switches installing and testing server on Linux PC Asterisk Linux specific start and stop scripts starting and stopping server instant messaging recording voice calls on Windows PC Skype instant messaging integrating Skype into podcasting experimenting for perfect Skypecast Mac podcasting tools Windows podcasting tools Intel 537EP chipset Intel V 92 Data Fax Voice modem card interactive voice response IVR Sipura ATA interface cards international calls Internet routing calls over using pure VoIP dialing 2nd 3rd using to link several remote Asterisk PBXs 2nd 3rd 4th Internet Phone Wizard 2nd 3rd INVITE method SIP 2nd IP addresses setting for Uniden IP phones IP phones 2nd downloaded from lib eommolketab ir downloaded from lib eommolketab ir allowing to place calls via Asterisk configuration with TFTP 2nd 3rd configuring BudgeTone 101 recording calls on your PC setting to use SIP server IP Precedence IP telephony IP telephony access devices P to IP calling enabling for Uniden IP phone iptables policy commands for kernel firewall iTunes controlling from Phlink 2nd downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M wW K M K Jabber protocol JAJAH JavaScript JBOSS application server jitter buffer settings for Uniden IP phone network
92. re familiar with Crystal Reports or Access data grouping in reports should be a friendly concept If not read onyou re in good hands Dragging to the right column of the Pivot Table Report worksheet treats the data from the CDR column as summary data It s probably easier just to start dragging column headings and see what happens Start by dragging the Source column to the left column in the pivot table Next drag the Duration column to the right column in the pivot table These two drags will build a report like the one in Figure 4 8 which shows a sum of minutes for each caller on the system over the period of time covered by the CDR worksheet In Figure 4 8 the majority of callers are PSTN phone numbers the 10 digit numbers though the majority of minutes are from private extensions 104 200 etc Extension 200 has the most minutes261 Of course this report doesn t tell us how extension 200 spent all those minutes whom downloaded from lib ommolketab ir downloaded from lib ommotketab ir 200 was talking to so let s drag another CDR heading from the toolbar to the lefthand column Drac the Destination column and the report will now show the minute totals of each phone numberto whom each caller placed calls as shown in Figure 4 9 Figure 4 8 A Pivot Table Report that shows all the minutes of call activity on the system broken out by caller ce ee M e 1m y 2 Master CDR xls 2 Sees a ee 08 ee YO ee ee ae eed Oe
93. s hard to Know once the case is on There s one more thing to connect after the TDM4OOP is slipped into place a four wire hard drive power cable that runs from one of the PC s power leads to a power connector on the card This cable brings power to the card above and beyond what s available from the PCI bus so that the card can provide ring voltage to any phones that are connected You don t need to connect this little power cable if you aren t using FXS modules Once the card is in and your Linux box is booted up you ll need to make sure theZaptel drivers that were compiled when you first installedAsterisk are loaded before Asterisk is launched as a part of your normal system startup To accomplish this execute these commands before Asterisk is launched perhaps in etc rc d rc local sbin modprobe zaptel sbin modprobe wctdm ee 8S Note the difference between this startup routine which provides driver support for both analog phones and analog phone lines and that of Connect a Legacy Phone Line Using Zaptel Hack 44 which provides driver support only for phone lines The addition ofwctdm is the difference Run make config in your Zaptel and Asterisk source directories to create Startup scripts that are customized for your Linux distribution The wcfxs driver needs to be modprobed only if you re using FXO modules with phone lines downloaded from lib ommolketab ir downloaded from lib ommotketab ir attached and the
94. server on your Very useful for debugging network ae Upgrade Will use the URL from Upgrade Rule to upgrade prow Itonmng Enable 1 the SPA s firmware automatically This requires the Sipura profile compiler If you have more than 10 SPAs you should be able to obtain this tool to aid in configuration Contact Sipura for more information Provision Provisioning Enable Yes This controls the IP TOS value for Real time Transport Protocol RTP or audio packets from the SPA When used in conjunction with intelligent Switches and routers this can ensure excellent voice quality on your network This is a common request It will disable the splash ring for voicemail notifications RTP X Line TOS DiffServ Varies Value Otherwise your analog phone will chirp every so VMWI Ring aaa Splash Len j often if you have a voicemail Very annoying Bperenred Sets the preferred voice codec to use Various X Line Varies codecs are available with quality bandwidth trade Codec offs 5 5 3 Dial Plan Magic Of all of the options on the Sipura thedial plan lets you be the most creative The dial plan is a string of characters that tell the Sipura how to treat callswhere to send them any digits to add or remove etc In its most basic use the dial plan controls when to send calls VolP devices are much like cell phones You have to send the number as a whole to the remote server But how does the ATA know when you are don
95. so understanding your TSP s prescribed method for installing firmware patches onto your ATA is important I ve chosen Packet for this example Refer to your specific TSP s support site for details on its update procedures 1 9 1 1 Get the firmware update Packet8 for one offers a Windows executable that you can download from its web site http web packet8 net download This tool will automatically identify your Packet8 provided DTA 310 ATA download the patch and install it If you prefer not to use the tool you can install the patc using the Packet8 ATA s web interface Instead of downloading the executable installer tool just download the firmware file Save it and remember the path where you saved it 1 9 1 2 Locate your ATA Next if you don t know your ATA s IP address use Packet8 s IP address identification service to find out what it is This will be helpful if you ve forgotten it or if your ATA is configured to get its IP address via DHCP Simply pick up your phone and dial 0120003 This will play back a recorded greeting that includes your ATA s IP address Next go to that address with your web browser using a URL like this http 10 1 1 200 replacing 10 1 1 200 with your ATA s actual address downloaded from lib ommolketab ir downloaded from lib ommotketab ir When the ATA welcome page appears click the Upgrade Firmware link Click the Browse button to locate the firmware image file you downloaded earlier
96. still havetraceroute of course While not implicitly a QoS measurement tool traceroute can gather useful performance data from the VolP network 6 4 1 Using pathping pathping is similar to traceroute It first determines the IP route along all hops from the host where it s running to the host at which it s targeted Then it collects information from each hop along the way like latency times and displays what information it has collected The following command returns the hostname and IP address from each hop along the route to the destination if each hop provides an ICMP response C gt pathping www broadvoxdirect com The output shows the route to the destination similar totraceroute Tracing route to www bigvoxdirect com 65 67 129 23 over a maximum of 30 hops O kelly 6aizy9qdl cel client2 attbi com 10 1 1 202 ty lt ch ae downloaded from lib ommolketab ir downloaded from lib ommotketab ir 2 10 248 164 1 3 bicOl elyehel oh attbb net 24 131 64 38 4 12 244 65 61 E e eg oe e 6 Morz7 p 0 piloa io ealc ne 12 123 157 26 torl p0 4001 onpas inat Eeee lent 22 17 IUL thrl cloe noeny tpeettenes 12 122 2 17 o er7 O500gaSoany 2pselLaner lal ea 10 so 1 0 0 gar4 NewYorkl Level3 net 4 68 127 5 11 ge 2 1 0 bbrl1l NewYorkl Level3 net 64 159 4 145 12 so O0O O0 O0 mplsl1 Clevelandl Level3 net 209 247 11 134 13 ge 6 0O hsal Clevelandl Level3 net 209 244 22 98 14 BIGVOX DIS hsal Level3 net
97. talk format which is stridently avoided by most of those downloading teenagers mentioned earlier is a content miracle It won t get you in trouble with the RIAA either But you don t need to have Rush Limbaugh s golden microphone or Howard Stern s subtle wit to master the talk format in your podcasts You just need to be able to record your Skype or MSN Messenger or AIM VoIP conversations call them interviews and podcast them You can also use your computer to record your traditional phone calls for podcast purposes using tools like Phlink Hack 15 3 10 2 Mac Podcasting Tools Using a tool like WireTap Hack 23 grabbing the audio from a Skype conversation is a snap Then it s just a matter of editing it down the talk format isn t forgiving of coughing fits or belches and downloaded from lib ommolketab ir downloaded from lib ommotketab ir converting it to MP3 format Using Cacophony orGarageBand you can insert the recorded interview into the middle of a podcast or give it some bumper music just like the pro talkers Another cool app for recording audio from VoIP conversations on Mac OS X is Soundflower http www cycling74 com products soundflower html This app actually lets you treat the output of any application as a sound input device in other applications so you can use it to recordSkype conversations and to process them later on so that they sound differently With GarageBand for instance you can a
98. that for security I argue that it s actually harder to secretly record calls with VolP than it is on the PSTN but let me digress here To clandestinely record a G 711 uLaw phone call you ll need to be able to runtcodump the common packet capture utility or its Windows counterpart windump This means you ll need to be a privileged user on the machine you re going to record from for Windows this means Administrator for Unix it means root You ll also need the ability to view network traffic to and from the host s participating in the call This means running the capture on one of the hosts directly programming your switch to let you monitor the port where one of the hosts is connected or gasp connecting both hosts to a hub where you can capture packets to your heart s content To put this in plain English unless you re using a hub or a specially configured switch you ll be able to record calls only from a device that s actually on the call pathi e the caller s host the receiver s host or a VolP server in the middle of th conversation 6 15 1 The Hack It s possible to do this hack on Windows you ll need the sameWinPCap library you used when you installed Ethereal on your Windows PC Hack 81 you did install Ethereal already right However I ll assume you re using Unix sincetcodump is a standard Unix utility and because it s easier to install vomit on Linux or BSD than it is on Windows 6 15 1 1 Compile and instal
99. that you don t have to This kind of application is a lot of fun because while it technically doesn t use any VolP components it s still strictly legacy phone technology it will give you an idea of how much power you as a phone user have when you use software to enhance telephony applications After all your phone service is merely an application and you ll be using a PC application to enhance it You ll need caller ID service enabled on your phone line if you want your PC to handle your calls in this way 2 8 1 PhoneTray Free and PhoneTray Dialup A cool freeware app that provides a caller ID pop up window in the Windows system tray is appropriately named PhoneTray Free In addition to the pop up display PhoneTray will log all incoming callshandy when you ve been out and you want to know whom to call backand it has a feature called Privacy Manager that lets you block calls from certain callers Figure 2 7 PhoneTray also has a handy scheduler to establish your quiet time so you aren t receiving annoying calls in the middle of the night While these features might be available from your local phone company you can certainly save a few bucks by implementing them yourselfwith a PC tool like PhoneTray The only hardware requirement is a modem connected to your phone line Figure 2 7 PhoneTray Free s Privacy Manager downloaded from lib ommolketab ir downloaded from lib ommoltketab ir PhoneTray Free 1 20 MBR Privacy
100. use the web interface make sure that you are using a machine on the internal interface of the AstLinux machine eth1 and that you have obtained a DHCP address from the AstLinux machine Once you have done this simply point your web browser tohttps pbx this resolves only if you got your DHCP information from the AstLinux machine so be sure you are connected to the right interface You will be prompted for a username dmin and password astlinux Go to General then Setup and then Edit rc conf Therc conf file should open up in a small text edit window inside your browser Make any necessary configuration changes including setting thesxTIp family of variables To apply these changes simply save the file and reboot the system In future versions of AstLinux so many reboots won t be necessary but for now it is always nice to know that the system will come back up after you have made your changes 7 9 8 PBX Only Mode or Help Have Only One Ethernet Interface As noted earlier you don t really need two Ethernet interfaces If you don t want to use AstLinux as router you need only one interface when you configure AstLinux forPBX only mode PBX only mode will prevent AstLinux from attempting to configure your internal interface th1 and it will prevent the startup of certain services that are not necessary iptables routing QoS DHCP etc You can configure PBX only mode by commenting outINTIF in rc conf and rebooting Note th
101. wctdm driver needs to be modprobed only if you re using FXS modules with analog phones attached The ztcfg application tests the Zaptel driver configuration and returns nothing if it s valid If there s a hardware error or a problem with the Zaptel configuration files which we re about to discuss ztcfg will return an error description to the standard error output The etc zaptel conf settings tell the Zaptel driver framework whichmodules on the TDM4O0OOP card are which For a card with two FXO and two FXS modules a configuration file like this would be used zaptel conf example loadzone us default zone us fxsks 1 2 fxoks 3 4 The numbers assigned 1 through 4 are channel numbers thatAsterisk will use to refer to activity on each module Within Asterisk each legacy interface port on the TDM400P has its own channel Since the point of this hack is to build afour line phone server we re going to assume that all of the channels are using the same type of signaling zaptel conf example loadzone us default zone us fxsks 1 2 3 4 In this case there are four FXO interfaces to which we re going to connect one phone line apiece fxsks FXS Kewlstart signaling is specified because the phone company switch to which these phones connect expects us to be using a telephone Ordinarily telephones interface to that phone company switch called a foreign exchange office in signal ese using electro mechanical line signaling called FXS si
102. when your VolP traffic passes out of your local area network LAN and traverses the wide Internet For every router that your VoIP traffic passes through there is a chance for your VoIP packets to be delayed or worse dropped You can use thetraceroute tracert on Windows application to see how many hops it takes to get from your computer to a remote VolP provider Here is an example from my PC to a SIP based Internet telephony provider traceroute sip example com traceroute to sip example com 172 16 15 15 30 hops max 38 byte packets L Laslo oUrl 12216001 Uuagio te 0122 me U S me Gel Zend 10 09 10 L211 me 12 545 MS 22 1 o ms A led Oeeoeeo 10nd An Te2O0 MS 7 777 Ss Sio me 5 sip example com 172 16 15 15 74 124 ms 77 450 ms 77 033 ms This shows that my VolP traffic has to pass through five routers every time make a call If any router gets congested or starts having problems my call quality can suffer Though knowing about troublesome routers along the path is important to keep things simple you can focus most of your concern on the time delay between yourself and the destination host To determine this you can use the ping utility ping sip example com downloaded from lib ommolketab ir downloaded from lib ommotketab ir PING sip example com 172 16 15 15 56 84 bytes of data 64 bytes from 172 16 15 15 icmp_seg 0 ttl 53 time 74 4 ms 64 bytes from 172 16 15 15 icmp_seg 1 ttl 53 time 74 4 ms 64 bytes fr
103. you can t use iChat with your own VoIP server you need X Lite for that If you re reading this book from front to back you might be wondering if l m planning to show you how to build a VolP server For the record Iam but not until Chapter 4 Then again the protocol or innards of the software might make absolutely no difference to you plenty of folks use Skype which doesn t use a standard protocol at all Table 2 1 is a matrix of VoIP client software and their features and compatibility downloaded from lib ommolketab ir downloaded from lib ommotketab ir Table 2 1 VolP client software compared Uses Uses Uses eaan wit License Software Mac Windows Linux SIP H 323 1AX your own type server Gno Phone No No Yes Yes No No Yes Open source AX Phone N a c No K No ee Open source aele Tyee tno Tyee no two ree pen sr Yes no VoIP Free ware a e iChat No Free X Lite X cal PRO om S oza Free Comm Firefly Yes Free Gizmo Free Project Net No a gt No v No x No v Free meeting anome No No Yes No Yes No Yes Open source Meeting As you work through the hacks in this and the following chapters you ll become very comfortable with the differences and similarities of these programsand you ll have an even better feel for their strengths and weaknesses A quick Google on any of these program names will get you to a place where you can download and install the program
104. you go as well as using theAsterisk server as a screen so that you don t have to give people the number you areforwarding to This is a great way to keep your cell phone or parents home phone number private The Asterisk server will dial the cell phone on the second line and then bridge or conference the two lines together for the duration of the call The hardest part about setting up this configuration is connecting two lines or two SIP peers acting as lines Hack 43 to the Asterisk server Once that s done the forwarding part is simple But before we get to that let s check out the configuration for the two lines Let s assume that two SIP peers are connected to the Asterisk server vis vis a media gateway witt two SIP clients like the Clipcomm used earlier Hack 43 The sip conf configuration for these peers would look something like this 7711 callerid Outside Line 1 lt 200 gt canreinvite no context incoming dtmfmode rfc2833 host dynamic port 5060 type friend username 711 7712 callerid Outside Line 2 lt 201 gt Canreinvite no context default dtmfimode rfc2833 downloaded from lib ommolketab ir downloaded from lib ommotketab ir host dynamic port 5060 type friend username 712 Line 1 is SIP peer 7711 and line 2 is SIP peer 7 712 Let s say that the line we re going to receive calls on is line 1 and the line we re going to use to call the cell phone is line 2 Note context incoming T
105. 0 1 1 10 with se 8 4 231164 10 1 1 10 10 1 1 202 SIP Status 100 Trying i PE E E 20 11 10 a A SIP SOP Status 2400 OK WIth sess10n escript tan 10 6 047276 10 1 1 202 10 1 1 10 SIP Request ACK 571p 201610 1 1 10 11 10 255204 10 1 1 202 10 1 1 10 UDP Source port 5060 Destination port 5060 L2 11 1223220 10 1 1 10 10 1 1 202 SIP Request BYE 1p9 204 10 1 1 202 5060 y Connection Information Cc IN IP4 10 1 1 10 Connection Network Type IN Connection Address Type IP4 Connection Address 10 1 1 10 3 Time Description active time t 0 0 S2ssion Start Time Session Stop Time 6 j Media Description name and address m audio 19120 RTP AVP 1 Media Attribute a rtpmap 101 telephone event s000 3 Media Attribute a fmtp 101 0 16 a Medta Attribute aj silencesuppioftf a File Untitled 6715 byt IP 130 13M0 Don t forget to re enable the codecs after doing this experiment or you ll have a real problem to troubleshoot Call signaling issues can be frustrating especially when using a mixed bag of SIP products from different vendors and vintages J ust like for revealing SDP failures and authentication problems packet capture is the best tool for exposing any and all signaling problems downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 83 Sniff Out Jittery Calls with Ethereal 83 have seen the enemy and its name is Jitter One of the biggest concerns when usi
106. 00 a fmtp 101 O 15 Figure 6 10 The bottom pane of the capture window shows the media attribute list the SDP text payload that advertises the capabilities of the calling endpoint downloaded from lib ommolketab ir downloaded from lib ommoltketab ir 5 Untitled Ethereal EX File Edt View Go Capture Anahe Statisti Help No Time Source Desbnation Protocol Info l 4 240763 10 1 1 204 10 1 1 10 SIP SOP Request INVITE sip z019l0 1 1 10 with se 202 SIP Status 100 Trying 202 SIP SOP Status 200 OK with session description SIP Request ACK 51p 201610 1 1 10 UDP Source port 5060 Destination port 5060 SIP Request BYE sip 204010 1 1 202 5060 4 231162 10 1 1 10 1 9 6 040550 10 1 1 10 1 10 6 047276 10 1 1 10 1 11 10 255204 10 1 1 202 10 1 12 11 123220 10 1 1 1 1 H e Hef Fe Time Description active time t i 0 0 Session Start Time 0 Session Stop Time Media Description name and address mj audio S000 RTP AVP 0 8 3 98 97 101 Media Attribute a rtpmap 0 pomu so0o0 Media attribute rtpmap 8 poma so00 Media Attribute rtpmap 3 qsm s8000 Media Attribute aj rtpmap 98 iLec so00d Media Attribute a rtpmap 97 speex s000 Media Attribute a rtpmap 101 telephone event s000 3 Media Attribute a fmtp 101 0 15 we E 1 gt File Unttled 6715 byb P 13D 13M 0 Al EH HEHHEHE The capabilities are listed with a reference number following thertpmap keyword 0 pcmu 8000 ind
107. 00 U U RRA MAX 0 5 1 10000 This says that we will take 10 000 samples of SIP ping time One ping will happen every 5 minutes until 10 000 pings have occurred There are three data sources one for each of three providers that will be guinea pigs with which to build the graph You will notice that this creates a file called voiphacks rrd This is the database file that will be storing thelatency measurements The next step is to make the measurements The following shell script will launch an instance ofRRDtool taking latency measurements from the three providers using oursip ping pl utility once every 5 minutes bin sh SIP_PING sip_ping pl t while bin true do PROVIDER1 SIP_PING sip providerl com 2 gt dev null echo INF PROVIDER2 S SIP_PING sip provider2 com 2 gt dev null echo INF PROVIDER3 SIP_PING sip provider3 com 2 gt dev null echo INF rrdtool update voiphacks rrd N S PROVIDER1 5 PROVIDER2 5 PROVIDER3 done You can run this in the background on a Unix system Don t forget to make your script executable chmod 755 filename sh and put it in a place where it can be seen by any startup scripts if you decide to have it run on system boot If you d like to run the script periodically you can add it to you etc crontab file so that the cron daemon can run it automatically according to your own timetable After you have accumulated a day s worth of data you can graph the results Srrdtool gr
108. 000 The reverb effect is followed by a high pass filter which is an EQ technique that trims reduces Samples below a certain frequencyin this case 1 KHz You can experiment with the high pass and low pass features to trim frequencies letting you obtain a number of cool effects Make your music sound like it s coming through a megaphone or with a little reverb make it sound like you re singing in the shower Now it s up to you to find an appropriate venue for all this aural awesomeness in youl VolP setupper haps in your Skype answering machine Hack 37 downloaded from lib ommolketab ir downloaded from lib ommotketab ir 2 18 3 Resample and Re Level Sounds The SoxX bag of tricks has many compartments Aside from EQ effects and format conversion you can use SoX to downsample sounds as in the Cacophony example earlier in this chapter Hack 22 SoX can also alter a sound s volume level amplitude level To alter the sample rate use the r option and specify the desired sample rate in kilohertz Of course you can decrease downsample or increase the sample rate but increasing the sample rate won t result in a higher fidelity sound This example takes a file calledoytor wav and downsamples it to 8 KHz sox bytor wav r 8000 bytor_8khz wav To alter a sound s amplitude or volume level use the v option This example increases the volume of the sound by 25 using a negative value will decrease the volume sox v 0
109. 10 are keep alive packets that X Lite sends to its SIP registrar Not all SIP phones di this The bottom pane of the main capture window shows the actual hexencoded content of the packet an the ASCll encoded content of the packet that corresponds to it The hex is on the left and the ASCII is on the right This is where you can usually pick out problems an incorrect password or a botched username would be easy to spot this way Figure 6 4 Ethereal s Capture Options dialog downloaded from lib ommotketab ir downloaded from lib ommolketab ir Ethereal Capture Options Capture Interface Device NPF_ DOZ6E064 CO62 4695 BBE0 0C7191E6FA40 IP address 10 1 1 202 link laver header type jet C Capture packets in promiscuo C Limit each packet to E B v Update list of packets in real time Lise multiple Files Automatic scrolling in live capture Mest File DTEP Next File every ai L_ Hide capture info dialog Ring buffer with a Name Resolution Stoo caure alber E Meis ema ii L Enable network name resolution Oat i eje LI after a Enable transport name resolution Figure 6 5 An Ethereal capture of a SIP registration downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Live Capture in Progress Lthereal Fie Ect View Go Capire Anahe Spatictics Halp Bx eaiges OFRAR PDX D
110. 25 bytor wav bytor_loud wav downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 25 Mix the Perfect Announcement HACK 25 Put SoX to work mixing different sound filesmusic spoken wordsto make the ultimate announcement message Are you looking for an easy way to make a seriously cool announcement for your Skype answering machine or outgoing voicemail greeting on your Asterisk VolP server You could buy a copy of a high end audio package like Logic Express for all its cool soundmixing and effects abilities Even some of the simpler sound editing tools let you merge files but if you have Unix chances are good that you ve already got SoX If you re on a Mac or Windows box you re only a download away from having it so take the MacGyver approach and save some cash Mac Windows and Linux users can get SoxX from http sox sourceforge net In this little project we ll mix an announcement message with some background music and then trim the resulting file to just the right length all using the SoxX toolset Finally we ll save it in the appropriate sound format To get started find a piece of music that you think will make good background musicpreferably something that has no lyrics to interfere with the spoken message you ll bemixing in The music can be in any format SoxX can handle wav mp3 whatever If you can note the length in minutes and seconds of the music file as this could come in ha
111. 3 Simple Traversal of UDP NATs STUN is a protocol that helps IP phones deal with the problem of Network Address Translation NAT a common technique employed by many firewalls to mask a group of privately addressed devices like IP phones behind one or more public IP addresses The protocol is dealt with in more detail inChapter 6 Leaving stunServerAddr at 0 0 0 0 disables the UIP200 s STUN client and stunServerPort allows you to override the default port number DirectIpDialing No To enable direct dialing by IP address so that you can call another IP phone by its address rather than its phone number as shown in the example given in Make IP to IP Phone Calls with a Grandstream BudgeTone Hack 60 change this setting to Yes AdminPassword 1234 5678 The AdminPassword setting allows you to change the menu password rapidly on all of the phones that get their configs from this TFTP server The format isoldpassword newpassword downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 69 Customize Uniden IP Phones from TFTP HACK 69 Use unique configurations on each IP phone and while you re at it do some firmware revision control too There are two files for each Uniden IP phone on the TFTP server one that s shared by all of the phones on the network unidencom txt described in Configure Multiple IP Phones at One Time Hack 68 and one that s exclusive to each phone on the network These ex
112. 3 files for your on hold music source make an entry like this inmusiconhold conf to create a class default gt var lib streaming http 64 236 34 196 80 stream 1040 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Now create the directory var lib streaming and leave it empty and this class will play back the streamed audio after your next Asterisk restart downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 51 Record Calls rest Pitch the microcassette and stick on microphone With Asterisk all you need to record a phone call is Monitor There are two ways to record calls with Asterisk One way is to use a softphone that supports call recording or some other client side desktop solution in fact Secretly Record VoIP Calls Hack 85 describes precisely this scenario The other way is to haveAsterisk do all the recording and have Sox do all the mixing SoX short for SOund EXchange is the Swiss army knife of sound conversion tools It allows all kinds of format conversion resampling and mixing topics covered in more detail in Create Telephony Sounds with SoX Hack 24 To record a call with Asterisk you can use the built in Monitor dial plan command In extensions conf any extension can be monitored as follows exten gt s l Answer exten gt s 2 Monitor wav most recent call M This example creates a WAV file called most recent
113. 55 This can cause issues when your SIP host places its internal IP address in the Via header instead of the external address of the NATing router There are a plethora of solutions to this problem includinc Simple Traversal of UDP NATs STUN and RFC 3581 STUN is a client side technique used to determine the correct external IP address and port which are then placed in theContact header RFC downloaded from lib ommolketab ir downloaded from lib ommotketab ir 3581 is a server side solution where the server responds to the host that sent it the packet instead o the host in the Contact header You can see this in action by using thesip_ ping pl script from the previous hack sip ping pl v proxy01 sipphone com grep Via Via SIP 2 0 UDP 127 0 0 126655 branch Z9nG4bK1116 37044 Via SIP 2 0 UDP 127 0 0 1 6655 rport 14328 received 66 27 57 228 branch z9hG4bK1116737044 Note the addition of the rport and the received parameters in the response of this ping These show the remote port and IP address respectively While our client stated to the server that its Contact is 127 0 0 1 port 6655 the remote host decided to send the packet back the way it came to 66 27 57 228 on port 14328 Some router manufacturers have tried to get into the game and transparently rewrite SIP packets that pass through them Although this was borne out of good intentions it can cause problems on certain SIP implementations when the returning response
114. 955 July 03 1 55 PM aanren lt gt 3 Information a Mom Home phone 281 497 9958 July 01 5 08 PM 00T A Mom Home phone 201 498 0000 June 259 15AM E h 2 2 Ralph Randall 207 666 9971 June 2410 354M 0 201 684 0672 June 23236 PM li 1 H memm m m mee me o A 1 Important Calls 82 2 Mew O ea o A The program also offers a searchable call log can receive faxes which are saved as bitmapped files so that you can view or print them and turns your PC into a speakerphone so that you can listen to calls through your PC s audio output and speak into your PC s microphone input This allows you to use your PC as a phone and to record your conversations instead of merely using it for automated call processing Of course there s a good bit of that in Call Soft Pro too Automatic forwarding of recorded messages via email is supported and Call Soft Pro can provide music on hold for your callers too downloaded from lib ommotketab ir downloaded from lib ommoltketab ir Hack 15 Let Your Mac Answer and Log Your Calls PS Ovolab s amazingly simple Phlink telephony package lets you do some really cool stuff on your Maclike answer calls and remotely control your Mac with a telephone call Watching Steve Jobs pitching the digital lifestyle at Macworld Expo is a favorite pastime of Mac enthusiasts In fact there s little that Mac users love more than watching the leader of the Mac world tout ne
115. AP client will return the email address with which the fax number i e DID is associated as in this snippet of extensions conf incoming pstn exten gt s 1 SetVar DID 5 EXTEN exten gt s 2 Answer exten gt s 3 Ringing allow 4 seconds for the FAX detection exten gt s 4 Wait 4 1f no FAX send this call to be handled elsewhere exten gt s 3 GoTo incoming voice here s the fax handling extension which sends the call to the gt Fanc fax Context exten gt fax 1 Goto inc fax 1 1 faAne fa gt exten gt s 1 SetVar TIFFILE var spool faxes DID tif The mailfromdid LDAP inquiry is defined in Asterisk s ldap conf file exten gt s 2 LDAPGet EMAIL mailfromdid DID If the LDAP inquiry succeeds priority will be 2 1 exten gt s 3 rxfax S TIFFILE exten gt s 4 GoTo 105 If the LDAP lookup fails priority will be 2 101 exten gt s 103 SetVar EMAIL receptionist oreilly com exten gt s 104 rxfax S TIFFILE Now e mail the FAX file to whichever e mail address was decided upon exten gt s 105 System uuencode TIFFILE uuenc mail s FAX S EMAIL exten gt s 106 System rm S TIFFILE incoming voice non fax calls are handled here The result of all of this compiling and config tuning is that different email recipients now have assigned fax numbers on the PRI or assigned POTS lines for their exclusive use as inbound fax lines When you send a fax to
116. B C D E F G H CY J K L M N O P Q R S D U M wW BI D Z MAC addresses Pocket PCs Mac OS X installing vonageGauge podcasting tools recording videoconference with Snapz Pro X Skype 2nd Skype Events options dialog Soundflower widgets Macintosh answering and logging your calls 2nd 3rd 4th 5th 6th 7th 8th 9th Asterisk server caller ID announcement using Say command building PBX with Mac mini 2nd 3rd SoX utility MAD MPEG Audio Decoder madplay command Map It Gizmo Project masquerading Master csv file media gateways methods SIP mewencode command microphones for Skypecasts voice communication with gaming Microsoft Excel 2nd 3rd 4th 5th MIME Construct package MisterHouse mixing sound files with Sox modem card modular phone jacks ATA connection to Monitor dial plan command MP3s Asterisk music on hold editing for telephony with Cacophony recording on Windows PC streaming Internet radio station for music on hold MPEG Audio Decoder MAD Mpg123 player downloaded from lib eommolketab ir downloaded from lib eommolketab ir music downsampling sound files for telephony mixing with spoken greeting message music on hold Asterisk 2nd 3rd assigning different ohones lines to different music classes musiconhold conf file MySQL CDR interface for Asterisk downloaded from lib eommolketab ir downloaded from lib eommolketab ir Index
117. Cain amp Abel to launch the Cain amp Abel configuration dialog It s shown in Figure 6 14 Click the Filters and Ports tab and check the SIP RTP entry to ensure that you ll be capturing VoIP traffic Then click OK Figure 6 14 The Cain amp Abel Filters and Ports list downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Configuration Dialog Sniffer AFPR Arp Porton Routing HTTP Fields Filters and ports Traceroute TEF P HAVNE 5500 H TDS 1433 A Mysal 3306 SMB 139 445 i MS Kerberos 5 gg bz Radius IKE E SNMP IONS RIP H HSRP FIGRP R OSPF VRRFP Mea 65 4317 8000 40824 UDF Ports i oa aj JEE When the Configuration dialog disappears click the Start Stop Sniffer icon on the toolbar Now place a phone call on the locally running softphone This could be X Lite Firefly NetMeeting or whatever as long as it uses SIP or H 323 for signalingand RTP for voice transmission just about all VoIP applications do Click the Sniffer tab then the VoIP tab on the bottom of the GUI to reveal the call list Notice that as you place and receive VoIP calls on the machine whereCain amp Abel is sniffing your call log will begin to fill with entries on the VolP tab as shown in Figure 6 15 The call log will tell you the source and destination IP addresses of the media stream used in the VoIP call the codec that is employed if Cain amp Abel recognizes it and the por
118. Copyright 2003 Ambrosia Software lnc Figure 2 16 WireTap s sound settings Figure 2 17 WireTap s recording controls pretty simple downloaded from lib ommolketab ir downloaded from lib ommotketab ir If you didn t disable the option to launch therecording automatically upon completion it should immediately appear in a QuickTime window for you to listen to downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 24 Create Telephony Sounds with SoX roa Use the Swiss Army knife of sound conversion utilities for your Vol P setup Though dozens of utilities are available forconverting and tweaking audio files the cross platform open source audio tool called SoX really stands out If you ve got a Linux or BSD PC chances are pretty good that you ve got Sox installed Windows and Mac users will have to download a compatibl version of SoxX from http sox sourceforge net And since Sox is a command line utility you ll need to be at least a mediocre typist to get through this hack and the next two hacks You ll also need to know how to get to a command line on your particular platform On Windows this means running the MS DOS prompt On the Mac it s the Terminal Linux and BSD users need only to fire up xterm This hack will show you the ins and outs of using SoX to convert audio files from one format to another add audio effects and telephonize your audio through downsampling 2 18 1 File
119. D which indicates whether one of the analog phones is off the hook The rear panel has an Ethernet connector an Ethernet activity link indicator LED and a DC power connector More elaborate ATA devices integrate broadband routing and firewall functions allowing you to consolidate your VoIP ATA and residential firewall into a single unit TheZoom 5567 is one of these It has a broadband IP router with a firewall a four port switch an analog phone connector and a pass through connector for placing calls on a traditional Bell phone line in the event the Internet service fails So depending on your service setup could be a little more elaborate than just connecting the phone and the Ethernet to your ATA However in most cases the ATA is a simple no frills device designed to accomplish one thingget your analog phoneconnected to the world s biggest VoIP carrier network the trusty ol Internet After you ve gotten the ATA out of the package find a good place for it It should be close to where you intend to use the analog phone though a long enough phone cord would afford you more distance In Wire Your House Phones for VoIP Hack 3 you ll see how to use your house s existing phone wiring to hook up several phones to a single ATA Your ATA also needs to be close enough to your Ethernet switch or broadband router to connect to it with a CAT5 patch cable Once connected most ATAs will automatically register with your VoIP service provid
120. FILE In this context the Answer Command triggers fax detection If the call isn t a fax the dial plan calls for a call to SIP peer 202 If it is a fax thefax extension takes over saving the fax image into a TIFF file located in var spool faxes Another script can then process the file in any way you see fit perhaps printing it immediately like this exten gt fax 1 SetVar TIFFILE var spool faxes thisfax tif exten gt fax 2 rxfax S TIFFILE dump the FAX file to the default printer and remove the FAX file exten gt fax 3 System tiff2ps S TIFFILE lpr exten gt fax 4 System rm S TIFFILE tiff2PS is a utility provided in the libtiff package a library for dealing with TIFF files It s a standard part of many Linux distributions Red Hat included 7 4 2 Sending Faxes Receiving faxes with Asterisk is quite a bit easier than sending them because when receiving them the work of scanning them into digital form is done already This is the part neither Asterisk nor downloaded from lib ommolketab ir downloaded from lib ommotketab ir Spandsp addresses However these packages can very easily fax a TIFF file So it s up to you to get that TIFF file in a path where spandsp can grab it This can happen in any number of ways You can create a simple web interface that allows you to upload TIFF files to the server or if you have the right software you can just scan them directly using the Linux machin
121. Format Conversion File format conversion is perhaps SoX s biggest strength You can use SoX to convert from one format to another WAV AIFF etc and from one encoding to another uLaw MP3 etc It even Supports some fossilized sound formats like 8SVX and voc All of this format support is helpful if you want to use a file that you have only in some oddball format that yourtelephony software can t use In most telephony applications like voicemail andinteractive voice response IVR where recorded voice prompts are the user interface you ll encounter sounds in one of a few encoding formats GSM An encoding commonly used on cell networks and in Voice over IP calls This is the de facto encoding used by Asterisk voicemail greetings and other announcements ULaw or aLaw The two most common 8 bit pulse code modulation formats they are most frequently used in legacy telephony such as T1 voice connections and digital PBX telephone systems uLaw is common in North America and aLaw is common in Europe To convert a sound file from one format to another there are two ways to go SoxX can recognize the input and desired output formats merely by parsing the filename extensions you provide as in the following example downloaded from lib ommolketab ir downloaded from lib ommotketab ir sox basic _instructions ulaw basic_instructions gsm This syntax takes basic_instructions ulaw and creates a GSM encoded file called basic_instruc
122. M Skype Sound Alerts options dialog widgets windump Wireless Encryption Protocol WEP wireless LAN WireTap Pro workgroup Ethernet switches downloaded from lib ommolketab ir downloaded from lib ommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M w K M 4 X Lite softohone 2nd 3rd installing SIP configuration settings 2nd 3rd 4th making the call user manual X PRO Vonage version of X Lite X10 phone controller 2nd 3rd X100P FXO card XML files 2nd 3rd 4th downloaded from lib ommolketab ir downloaded from lib ommotketab ir Index SYMBOL A B C D E F G H 1 J K L M N O P Q R S U V W DX Y Z Yahoo Chat Yahoo Messenger Yahoo Widgets 2nd downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M wW K M Z zapata conf file distinctive ring settings of channels provided by TDM400P card Zaptel driver framework Asterisk 2nd 3rd Zaptel drivers Zaptel interface cards Zaptel interface channels configuring to detect distinctive signals zaptel module 2nd Zaptel compliant interface cards zaptel conf file Zoom 5567 ATA ztcfg application ztdummy driver downloaded from lib eommolketab ir
123. Manager mouiet Time MW Enable Quiet Time Message NotReceivingCalls wav b Weekdays 10 PM 7 aM Weekends 11 PM z a AM Mumber Local Default MHo Long Distance Long Distance Digital way Mo Toll Free Tall Free Default Mo I 500 927 1457 Newspaper Mo MoTelemarketing waw Auto 1 John Smith Reddlert way InvalidNumber wav Manual 4d Number Edit Name Number Delete Number Settings For 555 1235 4567 calls lw Ring Ringtone RedAlert wav E L Ignore Quiet Time M Announce Call Name C Number C Both lw Log Call iW Show Tray Balloon D Show Pop Up M Show Full Screen Alert for 15 seconds C Automatically after i rings Manually show Zap dialog For aj seconds For a small premium PhoneTray s developer will sell you a version of the package called PhoneTray Dialup which works with caller Denabled modems Using this feature if you re a dial up Internet user you can receive caller ID signals on your PCdesktop while remaining online You can obtain PhoneTray from http www traysoft com 2 8 2 Call411 Another nifty free application that handles caller IDs in your Windows workspace is Call411 While this app doesn t have the elaborate interface or the extensive callhandling options of PhoneTray It s an effective no frills tool for displaying caller IDs as shown inFigure 2 8 You can associate a custom ringtone with each caller ID if you like a
124. Mandarin or Japanese you get the idea you car search the Skype online forums at http forum skype com for learn German or learn Japanese You ll find bilingual users who would love for you to Skype them with whom you can try out your linguistic chops A casual perusal of the Skype Me forum will put you in touch with thousands of othe folks who are endpoints on the social networkendpoints with various interests e Pets http forum skype com viewtopic php t 29213 e Chinese genealogy http forum skype com viewtopic php t 29029 e CB radio http forum skype com viewtopic php t 28119 If you really want somebody to Skype you post a message in one of these forums and before too long you ll have a buddy list that s a mile long or a mile longer Maybe you ll even learn some Swahili 3 2 3 Still Don t Know Whom to Call Meet Kerli Here s something you ll find useful on both the desktop and mobile editions ofSkype the official though minimally documented Skypeecho test service This automated Skype service is a user named echo123 that records a 10 second sample of your voice and then plays it back for you to listen to This will give you a rough idea of how well your Skype setup is working andwhether it is working According to blog lore the voice that is heard in the sound test announcement iS purported to belong to Kerli a young Estonian woman There are also Chinese and Japanese sound test users that you can call e
125. NSE I2 123s v alms 320 ms 20 ms att gw y d l net 192 205 272 215 10 32 ms 28 ms 43 ms bb2 nye P1 O atdn net 66 185 151 66 11 29 ms 47 ms 34 ms bb2 vie P12 0 atdn net 66 165 152 201 12 64 ms 48 ms 62 ms bb2 chi P6 0 atdn net 166 185 152 214 13 60 ms 60 ms 62 ms RR DET atdn net 66 185 141 98 i4 59 rse 54 ms 66 mS O90 0 imilmil rtrle twmi rre com 24 1090249 065 lo of me 35 me 63 ms 1g0 1l fmhlmil ubr twmi rr com 24 1092292 16 64 ms 66 ms 68 ms www thelinuxfix com 65 31 69 11 Whether on Linux Windows or Mac traceroute s output tends to be the same This sample output is from Windows but alltraceroutes show you the minimum average and maximum latency to each hop along the route Not all IP networks permit ICMP trafficor traceroutes in particularbecause some system operators prohibit them for security reasons Most routes across the Internet should provide a valid response when using the traceroute command As you examine the output from the traceroute command pay Special attention to the variance in highest and lowest latency times fot the average latency time This variance is a good rough estimate of jitter between each hop If you don t know what jitter is downloaded from lib ommolketab ir downloaded from lib ommotketab ir don t freak out Just refer to Sniff Out Jittery Calls with Ethereal Hack 83 Keep in mind though that the latency and jitter patterns of VolP traffic which is UDP and
126. One of the more interesting concepts of AstLinux is the use of a single configuration file and the concept of a keydisk In AstLinux you can configure almost all of the system with the exception of Asterisk itself in one configuration file etc rc conf etc rc conf is a very simple text file with VARIABLE NAME VALUE pairs So for instance to set the IP address on the external interface you would uncomment ExTIP and change it to the desired network address You will also want to change EXTNM etc but will cover that in more detail later Now for the keydisk This is a perplexing concept to some people and it can be difficult to explain Think of it as a personality similar to a SIM card in a Global System for Mobile GSM phone The partition that AstLinux resides in is purely for AstLinux No user files or configuration is stored there this is how it can stay mounted read only and how the system can still function Also it provides a ton of flexibility and allows for some very interesting uses of AstLinux When the system first boots you will see several entries in the GRUB bootloader They all boot AstLinux they just pass different arguments to the kernel that the startup scripts then look at to determine what to do One of these arguments iSastkd So for the USB keydisk astkd should equal dev sdal To use another partition on the system just fill in the path to that partition 7 9 3 Hardware Requirements To use AstLinux you
127. P 104 b903 Dial SIP 104 44 r 2004 08 i 104 105 defaut 104 SIP 104 F162 SIP 105 F369 Dial SIP 105 40fr 2004 08 0 104 105 default 104 SIP 104 cS5a SIP 105 9ed0 Dial CSIR LOS Clr 2005 o8 q 104 Loa default 104 SIP 104 396F Dial SIP 105 40 r 004 08 Di 21 33 35 4403249517 s house w71 4403249557 4403249517 gt zap i at WdiceMail a ff Sz EFJ 104 105 defaut 104 S1P 104 1546 mii Dial g1p 105 40 2004 08 02 21 34 50 Egg 103 104 default 103 SIP 103 Sadc SIP 104 dfS8 Dial SIPF104 401r 2004 08 0 2 house anonymous wen Zapfi 1 1 S1P 103 4007 Dial SIp 1048Stp 1038S1P 10 gat 104 default 103 SIP 103 6c3f Dial SIP 104 40 r 2004 g G3 21 02 01 he i ee p F Sheet Sheet Sheets ME Je H Ms Once you paste the text or open the file select column A by clicking the A column heading Then use Excel s Text to Columns function on its Data menu This will launch a wizard that will help you organize the text file into columns so that it s actually useful within Excel You ll see a preview of the text you pasted in the bottom portion of the window that appears Leave the Delimited radio button selected and then click Next Select Comma as a delimiting character make sure no other delimiters are selected as in Figure 4 5 and click Finis
128. P clients the phones send packet messages to SIP servers such as proxies and telephone systems or to other SIP clients Such as other SIP phones In these packet messages are headers strings of data that form requests for specific functionality from the device on the receiving end The requests could be to establish a phone call or merely to let a SIP server know that the phone making the request is available to receive calls Another function of these requests is authentication On many systemslike your broadband TSP s VolP networkthe calling device must register and pass a uSername password authentication to place or receive calls 1 6 1 Different TSPs Different Policies downloaded from lib ommolketab ir downloaded from lib ommotketab ir SIP softohones such as CounterPath s X Lite have many many built in features They can signal cal transfers place callers on hold and even do conference calling so that three or more parties can talk together But whether these features are enabled by your TSP is another issue To conference call for example you might need to pay for an extra line Bear in mind that from one VoIP service provider to the next even a feature heavy softophone product could be impotent and then there are those TSPs such as Packet8 that don t support softphones at all 1 6 2 Install the Softphone To get X Lite download it from http www counterpath com X Lite is in fact a scaled down freeware ve
129. P clone check out Brew Your Owr Zaptel Interface Card Hack 64 Install the X100P card into your PC s PCI bus sorry Mac users you re stuck using a media gateway as covered in the previous hack and connect an RJ 11 standard phone patch cord from the wall jack of an active telephone company Plain Old Telephone Service POTS line into the appropriate port of the XLOOP On the X1OOP this port is the one marked with al etching of a telephone wall jack Now download and compile the Zaptel driver and Asterisk Hack 41 This creates the zaptel and wctdm modules which need to be loaded during startup by adding this code to the script that launches Asterisk right before the line where Asterisk itself is launched modprobe zaptel modprobe wctdm etc rc d init d asterisk start downloaded from lib ommolketab ir downloaded from lib ommotketab ir By now the card is in a PCI slot on the Asterisk server the phone line is connected and you ve compiled and installed the Zaptel drivers Your next step is to define the FXO trunk connection as a channel that is usable by Asterisk Once defined you can reference the channel within your Asterisk call routing scheme The POTS line can serve as the full time gateway for all PSTN calls and all telephones in your home or office Or the POTS line can just be a connection mechanism so that the Asterisk server can answer incoming calls on the POTS line if they aren t answered by a person withir a
130. P inquiry An LDAP client library for Linux openldap provides applications with the ability to access LDAP servers and perform such inquiries Your LDAP server might be an Exchange or Lotus Domino server where directory information is stored The Red Hat distribution includes OpenLDAP binary package and the OpenLDAF downloaded from lib ommolketab ir downloaded from lib ommotketab ir developer package But you ll need more than just the LDAP client library You ll also need an actual LDAP client for Asterisk such as Sven Slezak s LDAPget package Download it from http www mezzo net asterisk and unzip it into the Asterisk source directory usr src asterisk Next as root copy the app _lIdap so file into usr src asterisk apps Then use a text editor to add app Idap so to the list of applications that begins withapps in usr src asterisk apps Makefile While you have the Makefile open add the following rule just above theapp_ voicemail so line app I0ap 80 app Ladapo CC A SCOLINK 0 60 G llber lLidap Now LDAPget is ready to be compiled Save your changes in the text editor and exit back to the Shell where you ll issue these commands to compile the package cd usr src asterisk make make install If Asterisk isn t currently running start it Then go to the Asterisk command line and load the LDAPget module alternatively you can just restart Asterisk Coe Cile gt load apo tdao sc pbx CLI gt s
131. Pro automatic call answering automatic ring through 2nd Avantlook avatars buddy icons Away downloaded from lib eommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H H J K L M N O P Q R S W U M wW RI M 14 base stations extended range for wireless LANs bat phone automatic ring through 2nd 3rd 4th Bring Your Own Device BYOD service agreement broadband router 2nd broadband routing in ATAs broadband VolP service BroadVoice support of configurable softohones web based tool to place and manipulate calls Broadvox Direct VolP service web based toolset to configure find me follow me call list buddy list Skype BudgeTone 101 IP phone 2nd 3rd 4th Busy BYE method SIP downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S D U M wW x M 4 cable modems cables connecting ATA to your network RJ11 equipped telephone patch cable Cacophony sound editing tool using in podcasting Cain amp Abel 2nd 3rd 4th ARP poisioning used to intercept call 2nd call detail records CDRs 2nd 3rd 4th 5th 6th call events call forwarding from phone company lines to Asterisk server home phone calls to Skype home phone calls to your cell phone 2nd call quality 2nd 3rd 4th call sniffer recorder Call Soft Pro application
132. Pthat is if you know how As you might have already gathered you can configure IP phones in three ways directly using the phone s LCD and buttons through a web or Telnet interface or via a configuration file the phone downloads from a TFTP server during boot up Those three methods are presented in order of the level of detail to which you can administer the phone with TFTP configuration allowing the most precise control Once you ve got a TFTP server running Hack 80 you need only drop the right text files onto your TFTP server to mass configure your IP phones The Uniden IP phones don t offer the trademark high end look and feel of the Cisco 7900 series of phones but they do provide a good value nonetheless At half the price of a typical Cisco SIP phone the Uniden UIP200 SIP phone supports all of the standards fundamental to a VolP LAN inline power SIP and several of the key audio codecs Getting a UIP200 onto the network and doing useful things with your softPBX is a snap with the help of mass configuration via TFTP 5 11 1 Get the Uniden on the Network To get started l Il assume your UIP200 is connected to your Ethernet LAN and is powered up You might want to use a static IP configuration on the phone as opposed to DHCP so pop into the Uniden s Quick Setup utility by pressing the phone s Menu key Use the directional arrow keys to access the Network Settings menu Then press the Menu key to select it Now press the down a
133. Run Phlink Even When Logged Off _ Hack 17 Greet Callers Differently Each Day Hack 18 Use Caller IDs in AppleScripts Hack 19 Control iTunes from Phlink downloaded from lib eommolketab ir downloaded from lib eommotketab ir Hack 20 VolP While Fragging Hack 21 Google for Telephony Info Hack 22 Telephonize a Sound File _ Hack 23 Record an Audio Chat on Your Mac Hack 24 Create Telephony Sounds with Sox Hack 25 Mix the Perfect Announcement Hack 26 Sound Like a Pro Announcer _ Hack 27 Record a Videoconference _ _Chapter 3 Skype and Skyping Section 3 1 Hacks 2840 Introduction Hack 28 Get Skype and Make Some New Friends _ Hack 29 Skype Your Outlook Contacts Hack 30 Skype People from the OS X Address Book Hack 31 Enable Site Visitors to Skype You Hack 32 Speak Jyve Hack 33 Teach Your Browser to Speak Jyve Hack 34 Carry Skype in Your Pocket Hack 35 Degunk International SkypeOut Calls _ Hack 36 From Podcasting to Skypecasting Hack 37 Answer Your Skype Calls Even When You re Not Around Hack 38 Use Custom Rings and Sounds with Skype Hack 39 Emote by Sight and Sound with Skype Hack 40 Skype with Your Home Phone Chapter 4 Asterisk Section 4 1 Hacks 4158 Introduction Hack 41 Turn Your Linux Box into a PBX Hack 42 Attach a SIP Phone to Asterisk _ Hack 43 Connect a Phone Line Using an FXO Gateway _ Hack 44 Connect a Legacy Phone Lin
134. TAs configuration options via IVR top 10 options web interface two wire phone splitters Type of Service ToS 2nd downloaded from lib eommolketab ir downloaded from lib ommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S L U M WI K M K UDP User Datagram Protocol uLaw or aLaw encoding uLaw sound file Uniden IP phones mass configuring by TFTP Unit Converter widget Universal Currency Converter Unix Use Outbound Proxy setting X Lite User ID User Name setting X Lite using with a VolP TSP softphones differing policies UTP Ethernet patch cable downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H N J K L M N O P Q R S O U M wW RI DI Z V Cards V 92 PCI modem chip family Ventrilo verbose logging output Asterisk Via header vibrato effect SoX video conference video on Skype plug ins video4 IM visual emoticons VLANs Virtual LANs voice mailboxes voice recording techniques voice server 2nd 3rd 4th 5th 6th H 323 gatekeeper voicemail Asterisk service voicemail alternative for Skype voicemail and auto attendant Call Soft Pro voicemail server standalone 2nd 3rd 4th Voicemail service Skype VoicePulse service to dump unwanted girlfriends or boyfriends VoIP Voice over IP getting connected 2nd 3rd programming
135. TSP policies on congestion signaled by Asterisk source of jitter and latency connecting telephony devices to Asterisk FXO or FXS interfaces EE Contact header contact search contexts Asterisk dialing out on legacy phone lines connected to Asterisk PBX editing for TDM400P connected phone lines incoming CounterPath web site country codes for international calls downloaded from lib eommolketab ir downloaded from lib eommolketab ir credentials cron utility curl utility 2nd required for AMP currency conversions CVS repository for Asterisk at Digium downloaded from lib eommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S O U M wW BI D Z Dashboard widget system Apple Dataprobe AutoPAL date format for Asterisk logs demarc telephone company entry point disconnecting wires from phone company desktop telephony 2nd 3rd 4th 5th 6th 7th 8th 9th 10th 11th 12th 13th 14th 15th 16th 17th 18th 19th 20th 21st 22nd 23rd 24th Audio Voice Cloak choosing VolP client 2nd 3rd 4th creating telephony sounds with SoX getting info with Google Gizmo Project Mac 2nd 3rd 4th 5th 6th recording audio chat on Mac 2nd 3rd tracking Vonage account info 2nd 3rd VolP 2nd 3rd DHCP IP address assignment to ATAs problems with changes in dynamic IP addresses turning off for BudgeTone IP phone DHCP server 2nd Dial comm
136. Todd s fax number Todd receives the email When you send it to Susie s Susie receives the email and so on Of course it s up to you to populate yourLDAP server with the right information and to make sure the inquiry config inldap conf matches your LDAP server s schema downloaded from lib ommolketab ir downloaded from lib ommotketab ir Don t have an LDAP server You can use Asterisk s built in database commands to resolve DIDs to email addresses Chapter 17 of Switching to VolP O Reilly contains a command reference that covers these dial plan commands 7 5 2 Hacking the Hack In the previous hack you used thetiff2ps command to create a printable version of the fax but with a few extra steps you can turn a TIFF into aPDF file too PDF can be preferable to TIFF when using email aS we are in this project Consider the following dial plan changes to the inc fax context exten gt s 105 System tiff2ps 2eaz w 8 5 h 11 S TIFFILE ps2pdf gt S TIFFILE ps exten gt s 106 System uuencode S TIFFILE ps uuenc mail s FAX S EMAIL exten gt s 107 System rm f S TIFFILE Now instead of just encoding the TIFF file and emailing it the file is converted to a PostScript file an then to a PDF file before being uuencoded and emailed to the appropriate recipient downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 92 Teach Your Asterisk Box to Speak r92 Sometimes
137. Tools Schedule Tasks Set the task to start at 12 00 p m and enter the full path to the batch file you created earlier Click Advanced and you ll see a dialog like the one inFigure 3 12 Figure 3 12 Windows Scheduled Task s Advance Schedule dialog downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Advanced Schedule Options start Date Wednesday June 22 2005 J Deos i Repeat task caer fron E Until E Time Duration houris minute s _ F the task is still running stop t at this time Check the Repeat Task checkbox and set this task to run every 12 hours Then click OK and click OK again on the preceding Scheduled Tasks window Now every day at noon and midnight beginning at noon today if it s morning yourSkype Answering Machine greeting files will be swapped 3 11 2 Skype s Voicemail Service If you re using Skype on a platform that doesn t support the Skype API or if you would prefer a voicemail solution that allows you to retrieve your voicemail messages from anywhere not just from your Skype PC you should consider Skype s subscription basedVoicemail service You can sign up for the service at https secure skype com store member login html downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 38 Use Custom Rings and Sounds with Skype fius Skype allows you to use your own sounds to alert yourself to incoming calls and ev
138. Works AMP provides a web based GUI using Apache and connects to Asterisk using a combination of techniquesmost notably via the Asterisk Manager API It uses PHP to build the web pages you interact with and it controls Asterisk with code written inPerl MySQL provides a repository where the entire dial plan configuration is stored retrieved and modified by the web interface 4 14 2 The Setup Process AMP has a ton of software prerequisites as you can see But it s fairly easy to install The basic steps are spelled out here and are detailed in the following sections 1 Get the prerequisites including Apache and MySQL Install Perl modules and custom telecom tools Build the MySQL database SS E Run AMP s install script and finish up downloaded from lib ommolketab ir downloaded from lib ommoltketab ir 4 14 2 1 Get the prerequisites A few dependencies are standing between your Linux server and AMP Check to make sure that your Linux box is running Apache libtiff MySQL with development libraries installed PHP version 3 or higher OpenSSL Perl ncurses SoxX and curl If you re running a Red Hat 7 or later distribution you should have all of these packages either preinstalled or available via RPM If you re not using Rec Hat chances are still pretty good that you ve got everything you need because most of these packages are either commonplace orrequired by Asterisk and therefore are already installed on yout
139. X X DSA msw32 zip From inside this zip file copy the two SSLdll files to the Konfabulator wbin folder located at Program Files Pixoria Konfabulator UnixUtils usr local wbin Copy the two files from the curl download here too In total you should have copied four files into this folder 2 3 3 Gauging Your Vonage Utilization downloaded from lib ommolketab ir downloaded from lib ommotketab ir Launch the widget by double clicking its icon in yourwidgets folder The thermometer like display Shows you how many minutes are remaining on your monthly plan The more minutes youuse the lower the height of the mercury in the thermometer This can help you conserve your utilization and spread your usage out to control your Vonage burn rate Note that if you have an unlimited plan this isn t really doing much for you aside from showing you how many voicemails you have waiting To listen to unheard voicemails double click the text at the bottom of the thermometer and your web browser will launch Vonage s services page where you can hear them downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 10 Pick a Desktop VoIP Client HACK 10 There s no shortage of fantastic Vol P software for Windows Mac and Linux But which one or two do you need VoIP applications tend like email to have a few servers facilitating interaction on behalf of many clients In the case of email those clients are
140. Xy The Digium AXy FXS gateway supports pulse dialing providing a simple and complete solution In fact the IAXy is a complete gateway in and of itself it has an analog phone port on one side and an Ethernet RJ 45 port on the other complete with a little Inter Asterisk Exchange IAX user agent built in Figure 5 5 A 1920s vintage Western Electric candlestick phone downloaded from lib ommolketab ir downloaded from lib ommoltketab ir The Digium AXy supports pulse dialing out of the box Configure etc asterisk iax conf with some entries that will allow the AXy to be used as a peer rotary Cype peer username asterisk sedro C OU rer eor E In this case used rotary as the IAX peer name All you need to do once this config is done is register the IAXy with your Asterisk server as the IAX userasterisk The rotary phone connects to the AXy Now you can use the rotary phone as you would any valid endpoint in your dial plan by routing calls to AX rotary using a Dial command 9 14 2 Do Pulse with a Wildcard If you opt to use Digium s Wildcard TDM400 with an FXO module installed you ll need to add a line of code to the etc asterisk Zapata conf file on the Asterisk phone system In the Zaptel channel section of zapata conf for this particular FXO channel the following bit of config will enable the FXO port to Support pulse dialing pulsedial yes downloaded from lib ommolketab ir downloaded from lib om
141. a VoIP call is only as good as the network carrying it and many unsuspecting participants mistake VolP calls for cell phone calls downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 2 Use Pure VoIP Dialing with Your TSP HACK 2 By using dialing shortcuts you can keep your phone calls on the I nternet and avoid extra charges If you re able to make a phone call to a regular phone company subscriber using your new VolP service Hack 1 you re ready to learn some cool TSP tricks Your VolP phone bill is probably lower than that of your friends who still use traditional calling plans But a lower phone bill isn t the only luxury that comes with converting your service to VolP Because your call uses the Internet rather than the public telephone network to route your call you have access to several cool dialing shortcuts when you call subscribers of other VolP services When an IP network alone provides the pathway between caller and receiver it s said to bepure or native Voice over IP This can actually save you money especially if you make a lot of internationalcalls If you re a Free World Dialup FWD subscriber and you talk frequently with your buddy in Mexico who uses Vonage using dialing shortcuts will keep your calls pure VolP and allow you to circumvent any related long distance calling charges that would be assessed if your calls were to traverse the Public Switched Telephone Network PSTN
142. a new Skype username or reuse an existing one to log in Creating a Skype account is free though some Skype features such as voicemail require a paid subscription 3 2 2 Find Someone to Talk To The quickest way to find someone to talk to is to ask a real life friend for his Skype username Of course if you don t have a real life pal to Skype you can search the Skypeuser directory for a lonely soul willing to talk to someone Seriously Skype has a mode calledSkype Me that users can enable which says they are willing to talk to anyone who comes calling To find people willing to talk click the Add button in Skype s main window then click the magnifying glass icon to display the search window Here you ll see a checkbox labeled Search for people in Skype Me mode Enabling this option before clicking Search will hopefully display users looking for buddies to call them Double clic a search result to call that person One reason people have embraced Skype globally is because of its integrated social network People are able to find each other through Skype s built incontact search available through Contacts Search for People If you re looking for somebody with whom you can speak German for example it usually takes only a single visit to this window to find users in Germany with the status o downloaded from lib ommolketab ir downloaded from lib ommotketab ir Skype Me Of course if your objective is tolearn German or
143. a piece of the Festival distribution into a WAV which it plays back usinc Asterisk If you want to keep tabs on the weather in a few different cities you can create an extension for each If you don t have curl grab it from http curl haxx se downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 53 Put a Happy Face on Asterisk Using AMP 53 When you ve got Apache and MySQL on your Asterisk PBX you ve got the makings of a web based administration interface for your whole phone system Since Asterisk runs on Unix it is able to leverage many of the niceties of a modern Unix environment shell scripts Perl programs sockets and so on Historically one of the chief Shortcomings of Unixand of Linux in particularis the lack of a graphicaluser interface GUI Asterisk Shares Unix s general inferiority in the user interface department But there s something you can do about it Asterisk Management Portal AMP gives you some real interface power tools aweb based configuration tool suitable for nontechnical administrators database routines for storing and retrieving the PBX s dial plan and some handy preconfigured call flow and fax features that make day to day life with Asterisk much easier For instance AMP lets you upload music on hold files usin a web interface and lets you create IVR menus without having to type them directly into extensions conf or to program Asterisk macros 4 14 1 How AMP
144. a road warrior voicemail solution First you will need to download my Asterisk distribution AstLinux AstLinux is made to run from CompactFlash but it doesn t have to be that way Incidentally if you d like to run it from CompactFlash check out Build a SoftPBX with No Hard Drive Hack 95 Pay attention to the keydisk portion as you ll want one for this hack But because we want ourvoicemail server to be as reliable as possible am going to assume that you have a fair amount of storage spaceCompactFlast or hard diskavailable for use on this hack To build a standalone voicemail server you ll need the following e A standard PC with AstLinux e An DE to CF adapter e A CompactFlash card of 32 MB or greater a 256MB SanDisk is recommended Depending on what type of technology you are going to integrate this server with your hardware needs will vary If you are looking for an all VoIP solution possibly for use with another SIP PBX proxy etc you won t need any additional hardware and you can skip ahead to the actual setup However if you will be interfacing with a legacy PBX you will need to get yourself some PSTN interface hardware For some options visithttp www digium com and http www sangoma com The PCI interface cards from Digium and Sangoma allow you to connect to a legacy PBX using POTS lines or T1s For a crash course in configuring a Digium TDM card read Connect a Legacy Phone Line Using Zaptel Hack 441
145. ab ir downloaded from lib eommolketab ir 4408646084 6 6 05 8 34 PM OO 0F7 Re iy Today s calls ge This week s calls Monday Jun 6 2005 634 52 Ph Hote 2 9 1 Pop Up Caller ID Notifications In Phlink s Preferences window available by clicking Preferences from the Phlink menu you can enable an option that shows you a pop up window with the caller IDs of incoming calls on your screen sO you can decide whether you want to answer them without having to even take your eyes off the screen let alone leave your desk What s cooler than that Well how about telling yourMac to answer the call so that you can get on with what you re doing and not be bothered withanswering the phone To do so just click on the round phone button in the Phlink action window as shown in Figure 2 11 This starts the greeting to the caller and records the caller s message Figure 2 11 Phlink s action window 2 9 2 Custom Greetings When Phlink answers each incoming call it looks in thePhlink Items folder in the Library Application Support folder of your user profile for a file calledgreeting txt If it finds the file it uses the Mac s built in speech synthesis to speak the words in the file to the incoming caller To modify this greeting downloaded from lib ommotketab ir downloaded from lib ommotketab ir simply refill the contents of this file with whatever you like If Phlink can t find the gree
146. access your Skype buddies who have alphanumeric names by dialing your telephone keypad which has only telephone numbers The wizard s included software lets you associate speed dial numbers with contacts in your Skype buddy list simplifying the act of calling them Once you ve run Actiontec s installer your Skype buddy list will have an additional option in its contextual right click menu the Assign Speed Dial option shown in Figure 3 17 Click this option to define which two digit speed dial number to associate with each member of your buddy list That way when you want to Skype them using the attached phone you need only press the speed dial numbers Finally the Internet Phone Wizard has two LEDs that indicate what type of call you re engaged in a Skype call or a regular phone line call It gets its power from the USB port so that s one less power adapter to worry about too For more information about the Internet Phone Wizard see its manufacturer s web site at http www actiontec com To see how you can use the Internet Phone Wizard to provide Skype network access to an Asterisk PBX system check out Connect Asterisk to the Skype Network Hack 98 Figure 3 17 Skype s contextual menu with the Internet Phone Wizard installed downloaded from lib ommoltketab ir downloaded from lib ommoltketab ir downloaded from lib ommolketab ir downloaded from lib ommotketab ir Chapter 4 Asterisk downloaded from lib
147. ach with their own pleasant sounding announcements To sound check in Chinese try Skypingecho chinese If you d like to sound check in Japanese try soundtestjapanese These sound test users offer another neat trick If you send the text messagecallme they ll call you to initiate the sound check This will help you verify that your Skype is fully workingthat you can plac and receive calls hear and be heard 3 2 4 A Solution for Those Inevitable Antisocial Moments Sometimes you just don t want to be bothered That s why Skype tells you who is calling so that you can opt to ignore their calls Then again you might accept a call thinking it s going to be a short one only to hear the caller blather on about something about which they mistakenly believe you care In times like these you need a way out of the call Luckily there sGotta Go a Yahoo Widget that plays a sound like a fire alarm for example that gives you just the excuse you need to end the call The calling party hears the sound giving legitimacy to your claim that you ve botta go downloaded from lib ommolketab ir downloaded from lib ommolketab ir Check it out at http www widgetgallery com view php widget 2 7970 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 29 Skype Your Outlook Contacts P29 Place Skype calls from Outlook and even log your phone calls If you spend much time using Microsoft Office Outlook
148. ack is known to work only with the Intel V 92 Winmodem card The critical thing about using an Intel V 92 modem card that has not been purchased from Digium as an X100P but otherwise looks the same is that the vendor ID encoded into the card will read differently breaking the original Zaptel driver and rendering the card useless Fortunately there are two ways around this The most obvious solution is to hack the code of the driver Before you compil Asterisk and Zaptel from the Digium CVS archive Hack 41 you ll need to edit the zaptel wcfxo c file Here s the existing code snippet you ll need to change Static struct pci_device_id wcfxo_pci_tbl __devinitdata Oxel59 Ox0001 Ox8085 PCI_ANY_ID O O unsigned long amp wcxl0lp 0x1057 0x5608 PCI_ANY_ID PCI_ANY_ID 0 0O unsigned long amp wcx100S Change this section in zaptel wsfxo c to this Static struct pci_device_id wcfxo_pci_tbl _ devinitdata Oxel59 Ox0001 Ox8085 PCI_ANY_ID U O unsigned long amp wcxl0lp Oxe159 Ox0001 Ox8086 PCI_ANY ID 0 O unsigned long amp wcxl10l1p 0x1057 0x5608 PCI_ANY_ID PCI_ANY_ID 0 O unsigned long amp wcx100S The line added in the middle will allow the Zaptelwcfxo driver to work with standard Intel V 92 Winmodem boards while still keeping the driver compatible with official Digium X100P cards Recompile the Zaptel drivers Hack 41 and your Intel V 92 cards can be used as FXO interfaces
149. acket in the trace file screen navigate to Statistics RTP Stream Analysis Figure 6 12 shows the report analysis Figure 6 12 An Ethereal RTP stream analysis lets you know if you have the jitters downloaded from lib ommolketab ir downloaded from lib ommolketab ir Forware Direction H simihl Dohi bichi bio di biki iio E hi es dla Packet Sequence Delta ms BW kbps Marker Status 2600 0 00 af 1 60 ok 2601 19 02 af 3 20 0k 47602 21 60 Le 4 60 ak 42603 15 39 Pes 6 40 Ok 47604 20 07 2 00 ak Pe E a e ys r J Pi Fh Pu 4 Max deka 0 038832 sec at packet no 834 Total RTP packets 620 expected 621 Lost RTP packets 1 0 16 Sequence errors 1 By examining the stream analysis you can see that at least in this particular sequence the jitter is nearly nonexistent staying well below a rate of 1 ms In a problem scenario jitter would need to be 10 to 20 ms at a minimum to be audibly perceived of course the codec has a lot to do with how much jitter the human ear can tolerate G 711 is highly resilient to jitter and G 729 is less so 6 13 2 The Jitter Solution Once you ve got the jitters the only way to get rid of them is to implement QoS at the points on you network from which jitter is originating Typically these arerouters VolP servers or extremely busy switches More than a dozen different QoS specifications are available among themDiffServ RSVP VLAN and IP Precedence
150. active features of the SPA line of products Hack 62 It is the dial plan that is going to make this hack possible In your web browser for ATAIL click on the Admin link in the top right hand corner You should see several more options become available Then click Advanced You should see even more options become available Next click the Line 1 tab and scroll down to Username Enterata1 Do the same for Display Name Scroll down to Dial Plan In the Dial Plan edit box erase what is currently there and replace it with the following O02 a0e2 0197416060 tGZ 73000 Save your changes Now for ATA2 Switch over to the ATA2 browser window and click Admin and Advanced again Now move over to the Line 1 tab and down to Username and Display Name Fill in ata2 for both Again scroll down and fill the Dial Plan box this time using the values for ATAL POSA CALCI 92 1604 0 1012 300 Again save your changes Now any time you pickup either phone connected to line 1 on ATAI1 or ATA2 it will automatically call the phone attached to line 1 on the other ATA 5 6 3 Hacking the Hack Nothing says Holy phone mod Batman like a bright red rotary dial phone with the mechanical dial wheel removed Replicas of such phones are actually available on eBay as are plans to build ones downloaded from lib ommolketab ir downloaded from lib ommotketab ir that have flashing lights too But for this hack all you really need to do for an aut
151. aded from lib ommolketab ir downloaded from lib ommotketab ir Hack 33 Teach Your Browser to Speak Jyve 33 With a little extra help from the J yve web browser plug in you can extend Skype s presence features to your web browser If you ve signed up for a Jyve account you already know how cool the Jyve Q Card is But with the Jyve web browser plug in you can take J yve s intimate use of Skype s presence features to the next level This might mean being able to receive a Skype instant message whenever somebodwvisits a certain page on your web site or even letting people send you a voicemail from your web site Unfortunately to enable these features your web site visitors must install the Jyve browser plug in which works only with Windows You can get the plug in athttp www jyve com Once installed the J yve plug in enables a number of new protocol prefixes that allow your web browser to trigger different kinds of Skype functionality These prefixes includermMTo SVMTO and CREATECONFCALL Using these prefixes and a bit of HTML you can create links on your personal web page that leverage some of Skype s coolest features like instant messaging and voicemail Like SkypeOut and Skypeln Skype Voicemail is a paid service To use voicemail features you ll need to have paid for Skype Voicemail service 3 1 Add Skype Instant Messaging to Your Web Site To create a link on your site that allows surfers to send you an in
152. age rotary dial candlestick phonesand how to get them working together You might also pick up some handy tips for your cell phone as well downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 59 Record Calls the Old Fashioned Way P59 Digital and I P phone handsets are analog inside which means you can use a transducer microphone to record a phone call It s fairly easy to record from a standard telephone using aninline recorder switch These devices allow you to record the analog audio signal on a standard Plain Old Telephone Service POTS line or a handset to deskset line using an analogrecording device like a microcassette recorder or a personal dictation recorder An easy way to use one of theserecorder switches to produce a digital recording of a call is to connect the mono audio output to your computer s microphone line in Most of these switches such as Radio Shack s model 43 1237 offer a 1 8 inch male audio connector which is perfect for use with a PC sound card or a Mac line in which both tend to be 1 8 inch female connectors J ust plug the recorder switch into the phone line and your computer s audio input and you ve got an instant call recorder in the form of your favorite audio recording program such as Windows Sound Recorder Since inline recorder switches work only with analog lines you can t use them to record calls on digital or IP telephones If you want to use your PC to record f
153. ak your messagefor you but that is a little too cold Instead with a little work you can have Asterisk play the appropriate sound files that you ve recorded with your own voice used the sound recorder that comes with Windows to record several sounds one for each day of the week wdayl wav for Monday wday2 wav for Tuesday etc one for each month of the year month1l wav for January month2 wav for February etc and one for each day of the month 1 wav for the first 2 wav for the second etc You ll also want to record a beginning for your greeting and two different endings one for when you re in the office and one for when you are out named these filesstart wav endnormal wav and endooo wav When speaking these months and days you might be surprised how difficult it is to get the words to flow together without sounding choppy finally started saying an entire date and then cropping the file at the appropriate place to get it to flow better when Asterisk plays it By default Asterisk doesn t have a codec to play wav files Instead of installing a new codec you can use the SoX sound converter Hack 24 http sox sourceforge net to convert the files into the gsm format that Asterisk can play with its default installation Use the following command to convert a wav file into a gsm file sox foo wav r 8000 foo gsm resample ql downloaded from lib ommolketab ir downloaded from lib ommoltketab ir After you v
154. al TRUNK S EXTEN 20 exten gt _1NXXNXXXXXxX 2 Congestion exten gt _011 1 Dial TRUNK S EXTEN 20 exten gt _011 2 Congestion exten gt _NXX 1 Dial S TRUNK S EXTEN 20 exten gt _NXX 2 Congestion This dial plan will enable NANPA style dialing of local 10 digit local long distance international and emergency information services from your system to the SPA 3000 You will want to make sure to include this new section in your local phone configuration So if your SIP phones as defined in sip conf are in the local context you will want the local context inextensions conf to contain this line include gt spa trunk This will enable your SIP phones to use your new PSTN gateway Save extensions conf and reload Asterisk with asterisk rx reload 6 9 1 Configuring the Sipura downloaded from lib ommolketab ir downloaded from lib ommotketab ir Once you have unpacked the Sipura connect your POTS telephone line to the RJ 11 jack labeled LINE connect an analog telephone to the RJ 11 jack labeled PHONE connect Ethernet and then power up Once the Sipura has powered up dial from the analog telephone As soon as you hear the voice prompt dial LLO The answering voice will read back the SPA 3000 s IP address Moving to your PC enter the SPA 3000 s IP address in your web browser You should see a gray screen with some status information In the upper right hand corner click Admin and then cl
155. aling SIP and audio transmission RTP take place in two or more separate UDP connections and do not travel on the same port Many people make the mistake of adding port 5060 to their highest class of service for QoS This does nothing for audio quality and merely assures that SIP messaging call setup status etc is given highest priority While SIP is a time sensitive protocol RTP audio is much more so Also you might be thinking of adding port 22 SSH to this list Don t do it just yet You ll need to have more tricks up your sleeve for SSH INTPORTSS 5000 50cL This is a list of source ports to be added to the bulk class of service This should include all traffic that tends to be large sustained downloads uploads You might ask why port 22 SSH is listed here As mentioned before we have some special instructions for SSH later on Adding port 22 here essentially covers file transfers using SSH not SSH shell sessions NOPRIOPORTSRC 23 22 80 I10 143 943 This is the same as the NOPRIOPORTSRC line except it refers to destination ports NOPRIOPORTDST 25 22 80 110 143 943 6 7 1 The Actual Script downloaded from lib ommolketab ir downloaded from lib ommotketab ir Here will go over the actual commands from AstShape and attempt to break them down If you are not interested in modifying AstShape beyond adjusting the preceding values you do not need to reac this section If you need to do more tweaking or a
156. alling Once a call recipient informs you that his number is on the list it s illegafor your organization to call him again Here are some Google search queries that you can use to turn up phone numbers Suppose you want to turn up numbers in a given area code and prefix You can form your Google query like so 440 325 OR 440 328 The quotation marks surrounding the two expressions tell Google to treat them each literallythat is to return only instances of the entire expression 440 328 and not mere instances of the elements within the expression 440 or 328 Google will return web page hits that contain occurrences of the area code 440 and the prefix 328 you might get some non telephone related stuff too in its two most common forms with parentheses and with a hyphen Of course the results you get from that query might require a lot o interpretation and massaging before you can really use the phone numbers that you ve turned up in an automated dialing app or something similar 2 15 2 Complete That Phone Number Sometimes the results from a Google phone number search can be fast useful and simple to downloaded from lib ommolketab ir downloaded from lib ommotketab ir interpret Let s say you need to call somebody in your neighborhood like the local pizza parlor Let s also say that you know the pizza shop s phone number begins with the area code that s common in your neighborhood Bang a query like this
157. amp Finally add usr local lib to the etc Is so conf file which will include thespandsp fax libraries you loaded earlier Add the following lines to your var lib asterisk bash_ profile file PATH SPATH usr sbin SHOME bin downloaded from lib ommolketab ir downloaded from lib ommotketab ir export PATH export LD LIBRARY PATH usr local 1lib Then open etc rc d rc local in your favorite text editor Replace the line that currently loadsAsterisk probably something like asterisk vvv with this usr sbin amportal start Are you still reading Excellent You ve just installed AMP Now to try it out you can restart Asterisk Apache and MySQL or you can just reboot to achieve the same effect Once your reset or reboot is done point your browser to http asteriskServerAddress You ll be greeted with the Asterisk Management Portal the ultimate Asterisk GUI Now go have fun configuring If you want this page to be secured by a username and password you can use Apache s htpasswd utility For more info on this check out http www apache org downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 54 Run Asterisk Without Root for Security s Sake 54 Running a critical service as root makes a security minded sysadmin squirm But it doesn t have to be that way Asterisk doesn t need to run as the all powerful root user By default Asterisk runs as rootthe user account wi
158. and dial plan Asterisk Sipura ATA 2nd 3rd dialing by IP address dialing shortcuts for pure VolP 2nd DIDs DiffServ QoS mechanism 2nd digital telecommunications T1 digital telephones Digium TDM400P card 2nd Direct Inward Dial DirectX Display Name setting X Lite distinctive ring 2nd 3rd 4th dmesg command DNS Address setting for TSP provider X Lite DNS records for Asterisk servers DNS server Grandstream BudgeTone IP phone downloaded from lib ommolketab ir downloaded from lib eommoltketab ir Uniden IP phone Do Not Call Registry Domain Realm setting X Lite downsampling recordings of audio chats dropped packets DSL avoiding accidental disconnection voice traffic over dynamic IP addresses downloaded from lib eommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H N J K L M N O P Q R S O U M wW BI D Z echo on Ventrilo voice chat system with softphones echo command output echo test service effects eight wire CAT5 patch cable with two RJ45 connectors electrical damage to the ATA from phone company lines email routing faxes into voicemail to email functionality email integration emergency 911 calls emergency 911 service 2nd 3rd emoticons 2nd adding sound and video emoticons real time video encodings mixed voice and music file for announcement equalization SoX effect voice chat tools Ethereal 2nd 3rd
159. and phone lines to Asterisk via a media gateway is decidedly easier than using Zaptel so that s the route we re taking here The analog telephone adapter ATA pictured in Figure 1 lof Get Connected Hack 1 is a media gateway since its job is to connect an analog phone to a VoIP service provider s SIP server over the Internet In this case though we want to do the reverse of that We want to connect a telephone company line to our own server We ll do it using aClipcomm CG 200 media gateway an inexpensive Korean made gateway that supports connecting up to two phone lines the CG 400 Supports up to four When complete the phone line connected to the media gateway will be answered automatically by the Asterisk server and a greeting message will be played for the caller 4 4 1 Configure the Gateway The Clipcomm CG 200 and CG 400 are similar to other phone media gateways they provide an Ethernet interface or two and two or more FXO ports that each allow you to connect a telephone line But unlike some other gateway hardware the Clipcomm has a fantastic web based configuratior interface as shown in Figure 4 2 Figure 4 2 The Clipcomm s web based configuration interface downloaded from lib ommolketab ir downloaded from lib ommotketab ir System Information 19983 244 34 SIP Server J eoe i Server Registration E enable disable BOWwoOre ts i Lat VoIP Registrar Server FODN 192 83 934 24 Suppl
160. andards for signaling call events These events might be the beginning and end of a call an attempt to join a voice conference or looking up a phone number to discover the best way to reach a particular user on a VoIP network These three communication protocols are H 323 SIP and AX Very rarely does a single client support more than one of these protocols Firefly is an exception and providessupport for both SIP and IAX Having a basic grasp of the different protocols will help you choose aVolP client 2 4 1 1 H 323 the earliest VoIP standard An H 323 client such as Microsoft s Net Meeting really is good only in a corporate telephone system environment It was once fashionable to use H 323 to have voice conversations with buddies over the Internet but the rise of broadband firewall routerswhich break the H 323 signaling protocoland the downloaded from lib ommolketab ir downloaded from lib ommotketab ir growth of better protocols such as SIP led to a backslide inNetMeeting s popularity as a personal VoIP tool Microsoft has since replaced much of the functionality of NetMeeting in its Windows Messenger IM software So unless you need a softphone that works with your H 323 based PBX system like an early model Nortel PBX or Cisco media gateway you re probably best served by foregoing H 323 based software GnomeMeeting is a very NetMeeting like application for Linux 2 4 1 2 SIP the dominant VoIP standard SIP has
161. aph pingtimes_daily gif v milliseconds title Latency over 1 day s now ld w 875 h 475 DEF providerl voiphacks rrd providerl MAX DEF provider2 voiphacks rrd provider2 MAX DEF provider3 voiphacks rrd provider3 MAX LINE2 provider1 882222 Provider 1 LINE2 provider2 228822 Provider 2 LINE2 provider3 222288 Provider 3 downloaded from lib ommolketab ir downloaded from lib ommotketab ir This will produce a graph of a full day s results taken at five minute intervals Figure 6 1 If you d like to graph a longer period of time you can change the s option to something like now 1w for a week or now 1m for the results graphed over a month The manpage forrrdgraph has some more examples that show how to tweak the output Figure 6 1 A graph of latency and jitter across a VolP link Latency over 1 ary d m d o ee oe a W em b m E amm ske b um I zpr e pm gy i g i i a E oS ME ESE F HJE i a OM f p ei itimecenis Fr sp E j M LAAR a SS NA DG a AS SPY Lat E biaa RS a a AA ON ee AE DO sia N E AA AA A A A a Se E g nae 12 00 141d tk Ob ii di ats 0 sei 0 Go oo iad je 4 Oo Os Gap Ora Oo ic te E frevider i frosider 2 fF trestaer 3 This graphs latency measured over time To determine jitter you can examine the variation in the line on the graph for each host If it s a consistent 80 ms flat line there s little
162. applications like Microsoft Outlook Eudora and Apple Mail But in Voice over IP clients can be standalone devices like IP phones and interface boxes ATA like those described in Chapter 1 or desktop applications like softophones orinstant messaging apps The information in this hack will help you decide which VolP client is right for you Some VoIP clients use well known standards such as the Session Initiation Protocol SIP and are designed for use with your choice of VoIP service providers Others are designed specifically to attact only to a certain service such as AOL Instant Messenger AIM Still others are builtusing open Standards but are hard wired to work with only certain services Yahoo Messenger uses SIP but works only with the Yahoo service That is you can t use the VoIP features of Yahoo Messenger with your own choice of VoIP service providers Some VoIP clients are quite functional out of the box such asSkype which provides a user friendly wizard to sign you up for Skype service and get you logged in With others such asX Lite and GnoPhonewhich are designed for use with your choice of service providers or even with your own VolP serveryou really need to know what you re doing to get much use out of them Since X Lite and GnoPhone aren t officially sanctioned for use with a particular provider you ve got to knowhow to configure them yourself 2 4 1 Meet H 323 SIP and IAX VoIP clients and servers use three common st
163. ard too Though it s optional you should upload some kind of portrait the Q Card looks a bit bare without a picture 3 6 2 Make Jyve a Buddy You might be wondering how J yve is able to keep tabs on your Skype status since only Skype downloaded from lib ommolketab ir downloaded from lib ommotketab ir buddies can see your Skype presence information You re completely rightonly Skype buddies can sex if you re Available or Hacking your Volkswagen So to share this info with people who view your Jyve Q Card you ll need to make J yve a buddy in yourbuddy list To do this click the Add button and type jyve01 for the buddy s Skype name or type what J yve instructs you to type on your account page 3 6 3 Add Jyve s HTML to Your Web Page On your account page click the Get HTML button and you should get atext box with some HTML that looks like this lt a href Javascript void 0 onClick window open http jyvesolutions Jyve com Qcard your_name htm QCard menubar no scrollbars no height 500 width 800 gt lt img src http jyvepresencel com qzoxy your_name png bordert 0 gt lt 7a gt This bit of HTML when placed in the source of your own web page will put your portrait on the web page with a link to your Q Card which will open in a new window when clicked Figure 3 5shows you an example of a Q Card embedded in a web page using the preceding HTML 3 6 4 Start Jyving The Q Card in Figure 3 5 i
164. ary Dial Phone with VoIP Chapter 6 Navigate the VoIP Network Section 6 1 Hacks 7287 Introduction Hack 72 Monitor VoIP Devices Hack 73 Inspect the SIP Message Structure Hack 74 Audit a Network s QoS Capabilities Hack 75 Graph Latency and Jitter Hack 76 Explore NAT Traversal Hack 77 Shape Network Traffic to Improve Quality of Service Hack 78 Create a Premium Class of Service Hack 79 Build a 100 PSTN Gateway in 10 Minutes or Less Hack 80 Make IP Phone Configuration a Trivial Matter Hack 81 Peek Inside of SIP Packets Hack 82 Dig into SDP Hack 83 Sniff Out Jittery Calls with Ethereal Hack 84 Log VoIP Traffic Hack 85 Secretly Record VoIP Calls _ Hack 86 Log and Record VoIP Streams Hack 87 Intercept and Record a VoIP Call _Chapter 7 Hard Core Voice Section 7 1 Hacks 88100 Introduction Hack 88 Build a Killer Telephony Server Hack 89 Build an H 323 Gatekeeper Using OpenH323 Hack 90 Turn Your Linux Box into a Fax Machine Hack 91 Build an Inbound Fax to Email Gateway Hack 92 Teach Your Asterisk Box to Speak Hack 93 Build a Mac PBX Hack 94 Monitor Asterisk from Your Perl Scripts Hack 95 Build a SoftPBX with No Hard Drive Hack 96 Build a Standalone Voicemail Server in Less Than a Half Hour Hack 97 Automate Your Voicemail Greeting Hack 98 Connect Asterisk to the Skype Network Hack 99 Forward Your Home Phone Calls to Skype Hack 100 Get Starte
165. as before still apply just make sure that you are sending callers into the default context when they ring in on Asterisk s Zaptel channels And there you have it a state of the art voicemail system using all open source components done in 30 minutes or less Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 97 Automate Your Voicemail Greeting 97 Use an AGI script with Asterisk to update your voicemail greeting automatically In the business world people often update their voicemail greeting on a daily basis For example yo might call a co worker and be greeted with You ve reached the desk of Bob Smith Today is Tuesday August 16 and I am in the office today You can imagine Bob s routine when he gets into the office in the morning he verifies the current date rehearses his new message a couple of times and then calls into his voicemail and updates it He probably stumbles over the words a couple of times so he probably has to start over at least once This takes far more effort and time than am willing to commit just to update the current date in my voicemail greeting I ve got more fun things to work on than that Here s how you can use an AGI script with Asterisk and some home automation to keep yourvoicemail greeting up to date automatically without lifting a finger 7 11 1 Create the Sound Files You can always use Asterisk s built in text to speech engine to spe
166. asy to email that TIFF file to somebody s inbox as it is to print it Here is a dialing plan that will do just that exten gt fax 1 SetVar TIFFILE var spool faxes thisfax S CALLERIDNUM tif exten gt fax 2 rxfax S TIFFILE email the FAX file to the receptionist and then delete it exten gt fax 3 SetVar EMAIL receptionist oreilly com exten gt fax 3 System mewencode e S TIFFILE mail s FAX S EMAIL exten gt fax 4 System rm S TIFFILE This configuration receives the fax MIME encodes it using themewencode command a standard part of most Linux distributions and emails it to the email address stored in thes EMAIL variable This is a catchall solution it sends every fax that s received to the same recipient who can then forward it and screen it if necessary to the appropriate person based on the content of each fax message 7 5 1 Automatic Fax Routing To have the Linux server automatically route each fax to the right recipient instead of having a certain email user doing it we must have a way of associating each fax with the correct recipient We ll have to associate a certain line or a certain DID with each user so that whenever a fax is received on that line or DID we ll Know where to route it Each phone line s number or each DID number if we re using a primary rate interface PRI will become a single user s fax number To associate a DID with a particular user s email address we can use anLDA
167. at downloaded from lib ommolketab ir downloaded from lib ommotketab ir the user dials as seven digits lt 124 gt 279 el ee Lea ee oe ee OL a ee ee ee The following configuration will work on the SPA 3000 only 4S lis sGowl gt xe lt Cowl S 129 eee ow la a eee Co EE Iioc eeoa Pow Le 16 8 G60 Sexes ew e ee a a ee OT a 9 The following list explains what this dial plan does for you e Calls to 411 and 911 go to the PSTN via the POTS line e xx e g 69 goes out via POTS e Seven digit and ten digit calls go out via the POTS line e Toll free calls go out via POTS e Three and four digit extensions are sent to the first SIP server defined e Eleven digit long distance numbers are sent to the first SIP server e International dialing is sent via SIP as well Here is yet one more advanced dial plan 45 di seo 2 9 aa a ea ee Lee TSH mee se 7 oe ee lt ra i This is very similar to the previous plan however any calls prefixed with 9 that are longer than three digits will be sent via the POTS line These are limited examples of what you can do with the Sipura line of ATAs After more experimentation you will quickly realize how much fun you can have with a 70 ATA Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 63 Build a Bat Phone HACK 63 Do you think Bruce Wayne uses VolP to receive emergency call
168. at the configuration for ExTIF still applies as usual to your first Ethernet interface etno 7 9 9 Wrap Up After the system boots you should verify IP connectivity You can do this by using theping command to attempt to reach a remote system So try typing ping www google com You should see ping replies If you do not you might be having Internet issues or you might have to configure a static IF address If the ping is successful you have correctly set up AstLinux Feel free to log into the system through the console or SSH and take a look around Explore the web interface as a lot of neat things are happening there If you have any questions you can always go to the AstLinux Users mailing list at http lists kriscompanies com Enjoy Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 96 Build a Standalone Voicemail Server in Less Than a Half Hour 96 If you re a decent typist it might take you only 15 minutes Asterisk comprises many quality applications and voicemail is one of them In fact Asterisk is perhaps best known for the feature set of its voicemail system In this section will demonstrate how you can harness Asterisk s extremely powerful voicemail application in 30 minutes or less This way everybody in your house or your office can have a customized voicemail greeting and message recorder even if they don t have a desktop phone Think of this as
169. at you can use to connect two analog phones to the same jacksuch as the one on your ATAbut this isn t an ideal solution Who wants telephone patch cables Snaking across the floor anyway Emergency 911 service is required on all VoIP lines sold in the U S But since VolP TSPs handle emergency call routing differently than the old Bell system it s best to check with your TSP to determine how they handle 911 calls This way you ll Know what to expect should you need to dial 911 Fortunately you already have all the wiring you need throughout your house to share a single VoIP provider s service with multiple analog phones The phone wiring in most homes is a two wire or four wire cable that runs from the telephone company s point of entry called thedemarc to various rooms in the house In these rooms a standard modular phone jack provides a place to connect a phone using an RJ 11 equipped telephone patch cable Modular jacks can support up to two phone lines since analog residential telephony requires only two wires per line The vast majority of telephone company subscribers use only a single phone line though The analog wiring in the home provides a single loop parallel circuit which means that you can piggyback modular jacks off each other If you need to connect a phone in a new room you just locate the nearest modular jack and run the wiring to it instead of running the wiring from the new room all the way to the demarc In the same way
170. atabase for storing the CDRs and AMP s replica of the Asterisk configuration To set this up blow the dust off your latent MySQL skills and issue the following commands usr bin mysql_install_db downloaded from lib ommolketab ir downloaded from lib ommolketab ir etc init d mysqld start or etc init d mysql start mysqladmin u root password db_root_pwd mysqladmin create asteriskcdrdb p mysql user root password db_root_pwd asteriskcdrdb lt usr src AMP SQL cdr_mysgql_table sql mysqladmin create asterisk p mysql user root p asterisk lt usr src AMP SQL newinstall sql H H H The text files directed to MySQL s standard input are provided as a part of the AMP distribution and they contain all the queries needed to set up AMP s database Now launch the MySQL client mysql user root p Once you get to the mysql prompt you can begin entering the access privileges for the database mysql gt GRANT ALL PRIVILEGES ON asteriskcdrdb TO asteriskuser localhost IDENTIFIED BY amp109 Query OK O rows affected 0 00 sec mysql gt GRANT ALL PRIVILEGES ON asterisk TO asteriskuser localhost IDENTIFIED BY amp109 Query OK O rows affected 0 00 sec mysql gt quit 4 14 2 4 Run AMP s install script and finish up About the only conventional part of this configuration is AMP s shell script for installing its standard files which you need to run now usr src AMP install_
171. atisfaction respectively We called theseemotion indicators emoticons or simply emotes And unless you ve never chatted online never talked to an Internet user or never seen the film You ve Got Mail with Meg Ryan and Tom Hanks or one of its several knockoffs chances are good you know what emoticons are Today with voice chat augmenting text chat and video chat and with webcams becoming commonplace the old fashioned emoticon seems plain out of place It s Small monolithic and well underwhelming Emoticons just don t convey as much emotion as a Speaking voice or the human face which is what you hear or see when you Skype folks using voice o video facilities You re about to use Skype add ons to enable audio emotes 3 D avatars and video chat making old school text emotes seem downright archaic The tools I ve looked at are all built around the Skype API for Windows though so if you ve got a Mac for which theSkype API has only just been introduced you ll be watching the action from the sidelines 3 13 1 Adding Sound and Video Emoticons Sound emotes and 3 D avatars are two really cool add ons for Skype that can enhance the social aspect of the Skype experience Sound emotes are essentially just prerecorded sounds that you trigger as a part of your normal sound transmission so that the person on the other end of the conversation can hear them As with old fashioned textemoticons sound emotes can be just the thing you need to lighten
172. ave the Internet Phone Wizard s Skype client log in as Ted2 while go about my normal business logging in as Ted1 In essence my home phone calls will be forwarded from Skype user Ted2 to Skype user Ted1 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 100 Get Started with sipX HACK 100 Asterisk like Cisco CallManager and other softPBX platforms implements SIP as a method of supporting SIP phones and trunks but does not employ the SIP design philosophy Yet SIP and SIP alone can replace your entire PBX system Enter sipX Like Asterisk the sipX project implements a call management server for Linux implements a voicemail server with message waiting indicators and allows you to build a voice network of SIP phones Unlike Asterisk sipX does it exclusively using SIP This means that external interface gateways must be used to communicate betweensipX and non SIP networks the PSTN H 323 etc In a minute at least if you follow this little outline you ll beinstalling sipXpbx a comprehensive SIP PBX server SipXpbx brings some cool functionality to the table including a built in web based administration tool two SIP softphones sipXPhone and sipXez Phone and a suite of interoperability testing tools Awesome stuff Perhaps most important sipX implements the following components of a SIP network according to the official IETF SIP specifications unlike Asterisk which only implements certain
173. b ir downloaded from lib ommoltketab ir cp etc proxy ini etc proxy ini is far more permissive than the default configuration file and it will allow you to register unauthenticated i e passwordless endpoints Now you can rungnugk with the config file in etc by issuing the following gnugk c etc gnugk ini 7 3 3 Register an H 323 Softphone Using OhPhoneX If you re using a Windows PC chances are you already have Microsoft NetMeeting This is a very capable softphone and it works well withOpenH323 In fact the next section describes how to set it up But since the OpenH323 project produces a phone too we ll use it That phone is called OhPhone and it s distributed as an executable for Linux Windows fittp www openh323 org and Mac http xmeeting sourceforge net These examples use screen grabs from the Mac OS X version The Linux and Windows versions have only a text based UI but for those platforms GnomeMeeting and MS NetMeeting make great alternatives The first thing you ll need to do with OhPhonexX is access its Preferences menu option The Gatekeeper tab of the Preferences window will allow you to specify a gatekeeper username password alias and E 164 address phone number as shown inFigure 7 1 In Figure 7 1 the address of the gatekeeper is 10 1 1 10 The ID is a superficial freeform ID used like caller ID The User Alias ID is required only ifgnugk is configured for authenticating regi
174. b ommolketab ir downloaded from lib ommoltketab ir Skype Options e Play sound when somebody is calling me oa py Privacy Vl Pes ssound wehariT cal primar e Play sound when other party is busy AL General P Notifications i o Sound Alerts e Play sound when I put someone on hold l 7 Biay sound when a held cal srme LA Sound Devices e Play sound when the other party hangs up eae tae A ens vi gt Connection _ Play sound when contacts come online Figure 3 14 Skype for Mac OS X Events options dialog When the preferences window appears click Sound Alerts Windows or Events Mac Here you can browse to the sound file you d like to assign as your ringtone The Mac version even allows you to have Skype use OS X s speech synthesizer to announce the name of the incoming caller if you like downloaded from lib ommolketab ir downloaded from lib ommotketab ir It s not just the incoming call event that you can disassociate from the default alert sound either Thumb through the event list to check out all the possible combinations downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Hack 39 Emote by Sight and Sound with Skype P39 In this modern age of voice and video emoticons are not forgotten and have become equally modern In the early days of text based chat we developedtext expressions like and to show signs of happiness and diss
175. b ommolketab ir downloaded from lib ommotketab ir e Festoon http www festooninc com e video4lM http www video4im com At the moment both work only with the Windows version of Skype but considering the recent release of the Skype API for the Mac suspect we ll see some video goodies for Mac Skype very soon To be seen by your Skype buddies you ll need a webcam connected Logitech Microsoft Sony and Creative all sell USB webcams that are suitable for Skype video You don t need a webcam to see the video from your buddy s webcam of course downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 40 Skype with Your Home Phone 40 Combine high tech VolP with a low tech analog phone to make VolP palatable to even your technology phobic spouse If you ve worked with Skype for very long you ve probably become accustomed to its mostly good sound quality and friendly interface However always have a hard time with how it feels to be Speaking to a computer Instead of the secure feeling of an old school phone receiver am uncomfortable speaking into a USB headset or worse still soeaking into my PowerBook s built in microphone For once just wish had a good old fashioned analog phone to slideup next to my earyes even for Skype calls Fortunately this is now possible Using theActiontec Internet Phone Wizard a USB device for Windows PCs you can integrate Skype with yo
176. beverage 7 3 2 Set Up the GNU Gatekeeper Once the OpenH323 build is finished you need to downloadand compile the OpenH323 Gatekeeper gnugk software Don t confuse this with the opengk software that comes as a part of the OpenH323 distribution This gatekeeper comes from a different source altogether but is built using the same libraries as opengk The big difference is that gnugk is a much more complete implementation of a gatekeeper and opengk is a reference example and is not very useful yet First download and save the gnugk source code from http www gnugk org h323download html into root It will be named gnugk 2 0 8 tgz or something similar After the download is finished build the gnugk package tar xvzf gnugk 2 0 8 tgz cd openh323gk make opt Now issuing the gnugk command will launch the gnugk gatekeeper If you receive an error indicating Shared libraries cannot be located make sure you ve got those environment path variables set in your login profile If you run into compiler errors try grabbing the x86 Linux executable from the gnugk site Regardless of whether you compile it yourself copy the contents of the package sbin directory into usr sbin and the contents of its etc directory into etc as follows cd openh323gk cp bin usr sbin cp etc gnugk ini etc To install a sample config file that allows any endpoint to register with the gatekeeper copy etc proxy ini also downloaded from lib ommolketa
177. bound Fax to Email Gateway Hack 91 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 52 Get Your Daily Weather Forecast from Your Telephone 1452 The Weather Channel has Local on the 8s every 10 minutes but why wait 10 minutes for your forecast when you can be listening to it on your IP phone right now Aside from cataloging sea species and running a really great tsunami readiness web site theNational Oceanic and Atmospheric Administration NOAA also operates the National Weather Service Those are the guys from whom excitable TV meteorologists get their severe weather warning and watch information But TV weather guys don t have an exclusive onNOAA s weather data feeds At NOAA s web site http www noaa gov localized weather data is published in text file feeds that are updated regularly Your Asterisk server can grab these feeds and thanks to Festival Hack 92 read you a weather report based on their contents Have a look at this example exten gt 50 1 AnsSwer exten gt 50 2 System usr bin curl s ftp weather noaa gov data forecasts city oh cleveland txt text2wave weather feed wav exten gt 50 3 Wait 1 exten gt 50 4 Playback tmp weather feed wav exten gt 50 5 System rm tmp weather feed wav f exten gt 50 6 Hangup The extension 50 grabs the text feed for Cleveland Ohio using thecurl application and immediately converts it using text2wave
178. button on the Equalization panel to flatten or reset the equalizer Then monitor your speech to hear how you sound You should have the familiar convincing tone of a half machine Sith lord Now fire up your Yahoo chat client or AlMand surf on over to the closest chat room Since AVC passes the modified audio through in real time you can chat live as Darth ot you can raise the pitch shift to sound like a chipmunk And don t discount the immaturity factor if you have kids who chat with their buddies online this could be a lot of fun downloaded from lib ommolketab ir downloaded from lib ommotketab ir Voice Alterations on a Mac If you re a Mac user and you want to achieve the same voice alterations that Audio Voice Cloak makes possible for Windows users you ll need to get your hands on a tool for Mac OS X called Soundflower This awesome piece of software allows you to pipe audio intoand out ofapplications in realtime The pipes carrying the audio are logical OS X sound devices so you can use them with any audio apps that support Core Audio the standard sound framework on OS X You can create a pipe to carry your raw audio into Pro Tools Free or Logic Express run it through whatever real time transformations you like including pitch shift and flange for the Vader effect and then send it out to your softphone or chat application using another pipe For more information on Soundflower check out http www macupdate
179. call while another context describes what behavior is caused by dialing 1 at some point thereafter These contextual behaviors are defined in etc asterisk extensions conf You ll need to make one more quick change toAsterisk s sample configs change the zap g2 definition for St ruUNK In extensions conf to Zap 1 This step might not be necessary with earlier versions of the Sample config This will allow outbound dialing to be directed to the correct channel Zap 1 the one that represents the connection to the PSTN Now since you ve added a new hardware interface you must restartAsterisk Once you ve done that try calling the POTS line you ve connected to the X100P using a second phone line or your cell phone After a few rings assuming you haven t changed the configuration Asterisk will answer and you should hear the familiar demo greeting that you heard in Turn YourLinux Box into a PBX downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 41 If you examine Asterisk s console output during this demo you ll see something like this Starting Simple switch on Zaps i Executing Wait Zap 1 1 1 in new stack Executing Answer Zap 1 1 1 in new stack Playing demo abouttotry Executing Dial Zap 1 1 IAX guest misery digium com s default in new stack Through the console output you can trace every step Asterisk took to recognize answer and process the incoming
180. call ext in var Spool asterisk monitor The m argument causes the call to be mixed automatically so that caller and receiver can both be heard in the same file Without them Monitor would just create two different files most recent call in ext and most recent call out ext ext represents the extension that the caller dialed to trigger thismonitor to begin with SoX must be installed for them option to work Without Sox Asterisk cannot output automatically mixed call recordings Most of the major Linux distributions provide a SOX package as an installation option 4 12 1 Hacking the Hack If you want to keep every call you record without overwriting already recorded WAV files you ll need to come up with an automatic way of uniquely naming every file thatmonitor creates The best way to do this is probably to base the filename off of the current system date and time Not only does thi make them unique but it also affords you an easy way to find files by date and time later on when you need them This example uses thes DATETIME variable to produce a file whose name Is something like 112205 09 45 42 40 downloaded from lib ommolketab ir downloaded from lib ommotketab ir exten gt 40 1 Answer exten gt 40 2 Monitor wav DATETIME M Once the files are recorded you can use cron to automatically archive them withgzip or even use the mail command to send them to an email address much as you did with faxes in Build an In
181. call out credits Gizmo Call411 application caller ID controlling for Asterisk PBX when using PSTN trunks having Asterisk server to announce identifying calls without in Phlink AppleScripts pop up caller ID notification Phlink prompting user to enter Windows software for calls handling with Windows software 2nd 3rd 4th recording VoIP calls on Windows PC recording VoIP calls with Gizmo CDRs call detail records 2nd 3rd cell phones using for 911 service CG 200 and CG 400 media gateways channel bank 2nd channels FXO 2nd legacy interface ports on TDM400P card chorus effect SoX Cisco IP phones downloaded from lib eommolketab ir downloaded from lib eommolketab ir customizing boot logo powering with standard inline power Cisco switches classes Click 2 Call Vonage clients 2nd communication protocols comparison of features and compatibility Gizmo H 323 clients IAX Inter Asterisk Exchange understanding features Clipcomm CG 200 and CG 400 media gateways CM15A controller codecs compatibility for SIP phones 2nd 3rd 4th 5th G 711 PreferredCodec SIP setting processor power and selection supported by Uniden IP phone command line interface CLI command line interface CLI CompactFlash based PC 2nd 3rd compatibility compression compromise solutions 911 emergency service POTS line speed dialing with VoIP device emergency 911 service POTS line conference calls Skype Skype users
182. ce com Tocsip 1805 SIP Status SIP Status SIP Request SIP From sip 646 sip broadvoice com To sip 1805 SIP Status SIP Status SIP Status SIP Request RTP Num packets 176 Duration9 522s ssre 1143032709 RTP Num packets 620 Duration 12 400s ssro 210 1425955 SIP Request SIP Status ip broadwoice downloaded from lib ommotketab ir Hack 82 Dig into SDP fxs No SI P based telephone call happens without Session Description Protocol and there s an SDP message inside every SIP Invite Knowing what SDP messages look like will help you identify the cause of failed calls that are rooted ir incompatible phones That s right if the phones or softphones calling each other with SIP don t have or aren t configured to use the same media codecs they won t be able to talk to each other Of course the phone isn t going to say Hey dude don t have the right codec It s more than likel just going to give you a busy signal So you ll need to dig into SDP to find out if a mismatch of media capabilities is to blame SDP is an essential part of SIP call signaling Its elements are text tokens sent inSIP packets with the SDP content type header These tokens advertise the capabilities and requirements of each endpoint according to the parameters of the application be it a telephone call instant message or something else During call setup specifically during thesip INVITE method the SDP payload is sent from one endpoint t
183. certain number of rings But first the FXO connection must become a namedAsterisk channel Each voice channel in Asterisk has a number This number consistently represents the same channel throughout all of Asterisk s configuration files and in its logging output The numbering of voice channelsespecially those that require a dedicated piece of interface hardware in the serveris determined by the order in which their drivers are loaded and the order in which they are identified i the PC s PCI bus Figuring out which card is whichsay in a situation where you have just installed three or four X100P cards each with its own POTS linecan require a bit of trial and error In this project we re using only one card and one line so it should be a breeze The voice channel we re going to create will be called Zap 1 1 Asterisk follows a similar convention when naming all voice channels even if they aren t analog phone line channels The channel name is divided into two pieces The first piece Zap 1 refers to the physicalZaptel interface channel which is either an FXO FXS interface or a PRI channel The second piece 1 refers to theline number more on multiple line interfaces later For Zaptel interfaces that support only a single line you can refer to them without the line numberi e Zap 1 and not Zap 1 1 Assuming you haven t touched the Asterisk configuration files since runningmake samples in the first Asterisk hack you ll have to make only t
184. ch 19 5 16 05 20 13 5 16 05 20 13 5 16 05 20 14 Si Fa Dial IAX2 199 0m icratech 19 16 05 20 14 57 16 05 20 14 57 16 05 20 14 468 ele IAX2 199 0m icrotech 19 5716 05 20 29 57 16 05 20 29 6 16 05 20 29 489 Dial IAX2 1990m icrotech 19 57 16 05 20 30 6 16 05 20 30 5 16 05 20 30 Bee Dial JAX2 199 microtech 19 5 16 05 20 31 6 16 05 20 31 5 16 05 20 31 CEFE Dial AX2 1990 m icrotech 19 5 16 05 21 21 5 16 05 21 21 5 16 05 21 22 Cire Dial AX2 199 m icrotech 19 5 30 05 15 00 5 30 05 15 00 CEES Dial PAR2 1 99 aimicrotech 19 5 30 05 15 00 5 30 05 15 00 keai Hangup 6 14 05 10 31 6 14 05 10 31 6 14 05 10 33 LEEI Woloe Ma lM in 200 6 14 05 19 32 6 14 05 19 32 6 14 05 19 32 6 ts a lMa in 200 6 14 05 19 32 6 14 05 19 32 6 14 05 19 33 Pere Hangup 6 14 05 19 31 6 14 05 19 31 6 14 05 19 34 P 498 volilce Ma liMa in 200 6 14 05 19 34 6 14 05 19 34 6 14 05 19 34 wi 499 Dial SIP 203ASIP 2016SIP 2 6 16 05 17 35 6 16 05 17 35 6 16 05 17 38 ctr Dial SIP 2038S1P 2018SIP 2 6 16 05 18 09 6 16 05 18 09 6 16 05 18 12 FP 8 Hangup 6 21 05 21 52 6 21 05 21 52 6 21 05 21 55 E Mas ber DRUcCSs CU E B Sal Sum 7 6329E 12 pa p OSCRL OCAP 4 Figure 4 7 A pivot table toolbar with Asterisk CDR fields Now you can drag those column names from the pivot table toolbar to the left and right columns of the blank Pivot Table Report worksheet Dragging to the left pivot table column treats the data from that CDR column as a group label If you
185. ch a phone line to the Ethernet network These devices provide a signaling proxy that allows analog phones and lines to be used with VoIP servers like thatMac mini PBX we re about to set up The Clipcomm CG 410 referenced elsewhere in this chapter is one such devicean FXO gateway device This hack works on any Mac with OS X 10 2 or higher To get started let s download an installation package for a Mac compatible distribution of Asterisk The one like was built by Benjamin Kowarsch a Mac telephony hacker You can download it from http www astmasters net or http www macvoip com You ll probably want to grab the latest version available from one of those sites Unpack it Stufflt should prompt you to unpack it as soon as you download it and launch the Asterisk pkg file This package looks like most Mac installer packages Step through the Introduction Read Me and License screens agree to the license and select the volume where you want to install Asterisk Figure 7 6 Your boot volume is the only place you can install Asterisk so don t bother yourself with trying to figure out how to install it elsewhere Besides unless you like time consuming nondefault settings that require the constant attention of a Unix snob the boot volume should be downloaded from lib ommolketab ir downloaded from lib ommolketab ir adequate The quickest way to launch Asterisk on any system once it s installed is to issue this command sudo
186. chapter 3 1 1 How Skype Works Unlike centralized voice networks like Yahoo Chat and MSN Messenger Skype uses a P2P network This means that call routing is handled by a collective serverless group of PCs running the Skype client software As peers on the same network each Skype node is responsible for routing calls on behalf of other nearby nodes This gives its proprietors advantages over traditional SIP based VoIP downloaded from lib ommolketab ir downloaded from lib ommotketab ir telephony service providers TSPs For instance Skype has less centralized infrastructure to maintain compared to a VoIP TSP like Vonage which has to have server capacity dedicated to every call it handles There are some centralized features in the global Skype network of course The contact search function wouldn t work so hot if it weren t able to query a global database of user information Centralized functionality like this is clustered around Skypesupernodes which are actually just PCs like yours that are running Skype Like a P2P file sharing network centralized search functions are facilitated using certain member PCs that are elected to have specific duties like cataloging user dati for searches and facilitating the logon process 3 1 2 What It Does and Doesn t Do Skype is largely feature complete on all the platforms it officially supports which makes it preferable for voice chat to something like Yahoo Chat which is a web ba
187. check the IP address you dialed check the phone s configuration to make sure it is listening on thedefault port for SIP5060and make sure SIP registration is turned off These options concern the Grandstream s use with a PBX server which isn a factor in this case 5 3 2 Mounting the Grandstream on the Wall For practical day to day use Grandstream shown in Figure 5 1 has a few shortcomings At the top downloaded from lib ommolketab ir downloaded from lib ommotketab ir of my bug list for the Grandstream 101 is its half baked support for being hung on the wall in addition to sitting flat on the desktop sayhalf baked because Grandstream provides screw holes for hanging the phone on the wall but it doesn t provide a notch to keep the handset on the phone when it s hanging So the handset just slips off the phone when you attempt to set it upright Figure 5 1 The Grandstream BudgeTone is a great cheap SIP phone This won t do I ve envisioned two ways to deal with this problem First you can go the Velcro route Apply about a square centimeter of Velcro adhesive hook strip to the handset at the point where a normal wall hanging handset s notch would be At the corresponding position on the phone itself put the same amount of Velcro latch strip so that when you hang up the handset it actually stays in place The second way to deal with the wall hanging problem which is probably a longer lasting or more downloa
188. clusive phone specific config files whose filenames contain the names of their corresponding phone s MAChardware address control the firmware and hotkey setup of that particular IP phone I ll step you through a Sample Uniden phone specific config file as it might appear on your TFTP server AutoFirmwareUpdate YES FirmwareFileName uip200_455enc pac FirmwareVersion BS4 55 Enabling AutoFirmwareUpdate with a yes will cause the phone to attempt a firmware patch automatically when it boots It will try to grab and install the firmware package specified by FirmwareFileName from the TFTP server The desired firmware version is specified by FirmwareVersion and the phone will grab the firmware file you specify only if the version is different from the version currently running on the phone MyLcdDisplay Maddie s Phone MyDialNumber 1138 DisplayName Madelyn UserNameForProxy 1138 PasswordForProxy uniden UserNameForRegistrar 1138 PasswordForRegistrar uniden MyLcdDisplay determines what greeting to display on the phone when it is waiting to call or be called and MyDialNumber determines what number to display DisplayName attempts to set the caller ID name to be used on outgoing calls if the softPBX supports this UserNameForProxy PasswordForProxy UserNameForRegistrar and PasswordForRegistrar establish the login credentials to be used when the phone logs into the SIP servers that handle its calls proxies and registrars are often hosted on the sa
189. col it can be quite useful You can even use thisscript to monitor remote SIP handsets and softphone applications because all SIP hosts if implemented correctly respond in the same manner to the request this script sends Brian Degenharat downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 73 Inspect the SIP Message Structure HACK 73 SIP is the predominant signaling standard among Vol P carriers and VolP enabled PBX systems so it might be a good idea to know something about it beyond what the acronym stands for SIP is a conversational connectionless signaling protocol In English that means that SIP uses a two way data conversation generally using a UDP socket Its message structure is similar to that of Simple Mail Transfer Protocol SMTP or HTTP messages which also contain headers and a payload SIP serves many purposes in a telephony environment including setting up and tearing down VolP phone calls Poking around the VoIP network with Perl is a great way to learn about SIP s message structure Besides monitoring the availability of hosts you can use the script from Monitor VolP Devices Hack 72 as an investigation tool for understanding how the SIP protocol works Using the v Switch you can see the full output of a SIP interaction sip_ping pl v 192 168 0 123 The preceding command sends the following SIP message to the specified host OPTIONS sip 192 168 0 123 SIP 2 0
190. congestion as source of jitter buffers Journal feature Outlook Jyve 2nd 3rd creating account HTML to embed Q Card in your web page making a Skype buddy web browser plug in downloaded from lib ommolketab ir downloaded from lib ommotketab ir Index SYMBOL A B C D E F G H M J K L M N O P Q R S W U M W BI M 4 Kewlstart signaling FXS foreign exchange station configuring for FXO interface card Kewlstart signaling FXS 2nd keydisk AstLinux 2nd downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H M J K L M N O P Q R S W U M W1 RI D 4 language setting LARTC latency 2nd graphing over time with RRDtool measuring on a route LDAP inquiry to associate DIDs with email address 2nd 3rd LDAPget package libdnet library libevent library libpri module libtiff line number Asterisk voice channel linking serveral PBXs over the Internet Asterisk adding remote locations configuring dial plan Linksys BEFSR81 broadband router Linux finding TFTP server NetFilter firewall 2nd 3rd PBX that communicates with the PSTN 2nd 3rd 4th 5th Skype turning into an Asterisk PBX local public safety dispatcher logging configuring for Asterisk VolP traffic 2nd 3rd 4th 5th downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A
191. ction 1 8 2 2 Program your VoIP device with speed dial to mimic 911 If you absolutely can t keep aPOTS line around or you prefer not to bear the expense of one merely for 911 dialing you might be able to get your VolP equipment to somewhat mimic the real thing e Program speed dial buttons or key combinations on your IP phone or softphone that will auto dial the local fire department or dispatch center via its regular non 911 number You should be able to obtain the local 10 digit phone number for the emergency dispatcher by contacting the administrative office of your local fire department Ask them to give you the phone number of the line where 911 calls are answered If you get lucky the person you ask will Know what Vol F is and will understand why you re asking but don t count on it e f that s a dead end you can program speed dial buttons or key combinations maybe even 9 1 1 itself into your IP phone or softphone as a shortcut for calling a trusted neighbor or family member This isn t exactly emergency dispatching but it s better than nothing 1 8 2 3 Use a cell phone for 911 Like a POTS line a cell phone can often be used effectively to reach the 911 dispatcher but check with your cell phone carrier to make sure 911 service is available and reliable in your service area J ust because wireless 911 service has been mandated by the Federal Communications Commission doesn t mean it works everywhere so check with your carr
192. ctly and you should find the appropriate lines manually instead Next you ll need to install the Asterisk Perl modules for Asterisk like this downloaded from lib ommolketab ir downloaded from lib ommotketab ir wget http asterisk gnuinter net files asterisk perl 0 08 tar gz tar xvgf asterisk perl 0 08 tar gz cd asterisk perl 0 0 8 perl Makefile PL make all make install H SH HOH Then grab a couple more Perl modules from CPAN and install them these enable the forwarding of faxes received if you want AMP to handle faxes perl MCPAN e install IPC Signal perl MCPAN e install Proc WaitStat To make sure that AMP s email integration works correctly grab a copy of the MIME Construct package from Roderick Schertler http Ssearch cpan org src ROSCH mime construct 1 9 and unpack it to root or usr src whichever you prefer Then from the directory where it s been unpacked install it as follows make Configure PL make install To add fax receiving support to AMP install the soandsp package per the instructions in Turn Your Linux Box into a Fax Machine Hack 90 Then you ll need to set up the MySQLCDR interface for Asterisk When you downloaded the Asterisk CVS this was downloaded to usr src asterisk addons cd usr src asterisk addons make clean make make install OS 4 14 2 3 Configure the MySQL database Now you re getting to the meat of the hack the MySQL d
193. d file just use SOX Hack 24 But to add the header a little Perl magic is needed 5 4 1 The Code This script was written by Tony Mountifield and its purpose is to create aGrandstream compatible ringtone file usr bin perl Sfilename shift or die need output filename n undef slurp whole file at once audio lt gt e dake this Sfilesize 512 length Saudio if Sfilesize amp 1 l length odd adda zero byte should never happen Paeudio chr 0 die Audio file too large n if S filesize gt 65536 this is the format for the header sneaderimt n nn C4 nm CC a224 n 216 n nm 36 azo get the current date and time Smin Shour Sday Smonth Syear localtime 1 5 Syear 1900 smontch 1 downloaded from lib ommolketab ir downloaded from lib ommolketab ir create the header with zero for the checksum Sheader pack Sheaderfmt O 0000 Sfilesize 2 O put checksum in later 1 0 0 1 version Syear Smonth Sday Shour Smin Sfilename oF 0000 or OOC8 why 256 0100 Sfilesize 2 Grandstream standard music ring Sanity check Sheaderlen length Sheader die header length wrong Sheaderlen n unless Sheaderlen 512 add the audio Sheader Saudio compute the checksum Schecksum unpack 1l6n Sheader printf checksum before 04x n Schecksum insert it in the correct place substr Sheader 4 2 pack n Sc
194. d from lib ommolketab ir downloaded from lib ommotketab ir Here I ll the second class of service the economy one as shown earlier tc class add dev SDEV parent 1 1 classid 1 20 htb rate S LINKSPEED kbit burst 6k prio 2 Finally I ll added the default class as shown earlier This is where undefined traffic will falli e anything not on the ports specified in the previous section tc class add dev SDEV parent 1 1 classid 1 30 htb rate S 9 SLINKSPEED 10 kbit burst 6k prio 3 The following command makes IP packets that have therr tos header set to 0x19 match Class 1 our premium class Remember that if you re prioritizing by port number you might have no needto prioritize by the ToS header So you might be able to skip this line especially if you have no control over your ToS headers as discussed earlier te filter add dey DEV parent 120 protocol ip 1 prio 10 u32 match ip tos 0x19 Oxff flowid 1 10 The following line says that any packets with anip tos header equal to 0x18 will match class two the economy class This isn t foolproof either not all packets even have a ToS header value inserted Starting to get the idea Discriminating by port numbers is more consistent and easier to manage than discriminating by ToS bits but there it is so you can see how it s done to filter add dev Sey parent 1 0 protocol ip 1 Drio 20 mos matoh ip toe UXlo OKIL Ilowid 1 20 The simple loop shown in the following snippet makes
195. d with sipX Colophon downloaded from lib ommotketab ir downloaded from lib ommotketab ir Index downloaded from lib ommolketab ir downloaded from lib ommotketab ir Copyright 2006 O Reilly Media Inc All rights reserved Printed in the United States of America Published by O Reilly Media Inc 1005 Gravenstein Highway North Sebastopol CA 95472 O Reilly books may be purchased for educational business or sales promotional use Online editions are also available for most titles Safari oreilly com For more information contact our corporate institutional sales department 800 998 9938 orcorporate oreilly com Editor David Brickner Production Editor Sanders Kleinfeld Series Editor Rael Dornfest Cover Designer Marcia Friedman Executive Editor Dale Dougherty Interior Designer David Futato Printing History December 2005 First Edition Nutshell Handbook the Nutshell Handbook logo and the O Reilly logo are registered trademarks of O Reilly Media Inc The Hacks series designations VolP Hacks the image of a Morse code tapper and related trade dress are trademarks of O Reilly Media Inc Many of the designations used by manufacturers and sellers to distinguish their products are claimed as trademarks Where those designations appear in this book and O Reilly Media Inc was aware of a trademark claim the designations have been printed in caps or initial caps W
196. dband Internet connection and they are normally supplied by your VolP TSP when you sigr up for their service In addition to an ATA some TSPs permit you to placeVolP calls using the following An IP phone These telephones connect directly to an Ethernet network using a patch cable or wireless link They have an IP address as a PC would and they communicate with the VolP TSP s data center over your Internet broadband link A softphone These are software programs that run on a PC and permit telephone style communication using your broadband link They appear on your Windows Linux or Mac desktop with graphica user interfaces that often resemble a telephone and they require that your PC have a microphone and speakers downloaded from lib ommolketab ir downloaded from lib ommotketab ir For this hack I ll concentrate on connecting to a TSP that provides an ATA allowing you to use an analog phone to place and receive calls via the Internet Table 1 1 lists domestic U S and Canada TSPs that provide broadband VolP calling Bring your own device means the TSP allows you to make phone calls across its VolP network usinc your choice of equipment such as an IP phone a PC or your own ATA TSPs that don t allow you to bring your own device will provide an ATA to make the connection Table 1 1 Vol P TSPs Company Web site Bring your own device AT amp T CallVantage http www att com No BroadVoice http ww
197. dd reverb to a live sound input before adding it to the podcast Or you can run it through EQ compress it or do a host of other cool stuff to it to whip it into shape for your own radio show Great for sound effects GarageBand in concert with Soundflower could be the makings of the ultimate radio drama podcast You can apply reverb echo and pitch shifting to sound inputs in GarageBand taking your dramatic podcast to the Grand Canyon or to a village of squeaky talking munchkins as in The Wizard of Oz 3 10 3 Windows Podcasting Tools A great place to start on your quest to make the ultimate Skypecast isTotal Recorder Standard Edition a Windows shareware tool that you can grab fromhttp www highcriteria com Like its Mac counterpart WireTap Total Recorder lets you intercept audio from one or more channels like the sounds from a Skype conversation and save it to a file that you can integrate into your Skypecast As you can see in Figure 3 9 the creators of Total Recorder were thoughtful enough to create the Record also input stream option which automatically mixes the microphone input channel with the sound output you re recording simplifying the task of recording your Skypecast that is unless you don t want your voice to be heard during the interview Once you ve saved a WAV file with your interview you can edit it into the rest of your podcast using Windows Sound Recorder or your favorite sound editor Then if your editor d
198. debar Download and install AVC from http www goldsoftware com downloads5903 html Launch it and after the shareware commercial you ll be able to click the All Controls button to reveal all of the sound altering controls available to you The program uses the default microphone input so if you re using a nonstandard microphone channel for your telephony or online chat you ll need to click the Recording Source button and select the right input Figure 2 2 Audio Voice Cloak s main interface downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Blaze Audio Voice Cloak Trial Fie Effects Presets SoundEffects ew ntemet Help Hilda Se art Monitor Your Voice Tittany Bark Bobby Belch Recording Source Big Bad Bob Glass Break Robby Robot Center E Echo Off Fiange Or 5 Cherus Off Roesetvaied Off E gd Equalize G SAVE Curent ettect settings 28 Restore factory default settings for While you tinker with AVC s settings you can monitor yourself with the aptly titled Monitor Your Voic button Beware you d better put on a pair of headphones or you ll get feedback To get the most authentic Vader imitation short of hiring Ben Burtt the famed sound effects guru from Lucasfilm you ll want a slightly southerly pitch shift drag the pitch slider down a notch or two and a flange effect click the Flange Off button to toggle it on Finally click the Center
199. ded from lib ommolketab ir downloaded from lib ommotketab ir durable approach is to drill a small hole in thephone base at the point where the wall hanging notch Should be The hole should be about one third of an inch to three quarters of an inch in diameter Then again at the corresponding spot on the handset screw into the plastic casing a very short round headed screw The head of the screw if small enough to fit into the hole you drilled should keep the handset firmly latched onto the phone s base Not pretty but it works downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 61 Build a Custom Ringtone for Your Grandstream Phone HACK 61 Sure your cell phone has a custom ringtone but does your IP phone With a little help from Perl you ll able to load any sound you like onto the Grandstream phone If you carry a cell phone you ve no doubt changed your ringtone once or twice From a sample of a vintage mechanical ringer to a recording of a C 3PO line fromStar Wars ringtones have become central to pop culture communication So why can t you customize the ringtone on aGrandstream IP phone one of the cheapest and most popular SIP hardphones available Well since you asked youcan It just takes a little hack job The Grandstream s firmware stores the ringtone in its own odd format auLaw sound file with a custom header at the beginning of it It s simple enough to make auLaw soun
200. ded from lib ommotketab ir 1 6 3 Make the Call When the X Lite phone has successfully registered with the TSP s proxy its main window will display a message like Logged InEnter a Phone Number Now you should be able to type in a valid public telephone network number try your cell phone for an easy test if you have one The service shoulc function at least as well as it would via an ATA and analog phone with one possible exceptioecho Echo is common with softphones if you re using your PC speakers to listen to the person on the other end of the call If you experience echo when you speak use a pair of headphones to cancel the acoustic feedback loop downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 5 Prioritize Packets to Improve Quality HACK 9 Voice traffic competes for available bandwidth on your broadband connection If there is not enough bandwidth packets get dropped VolP media streams require a constant uninterrupted data flow This data flow is composed of UDP packets that each carry between 10 and 30 milliseconds of sound information Ideally each packet ir a media stream is evenly spaced and of the same size In a perfect world a packet never arrives out of sequence or gets dropped Voice over IP media packets are framed in a highly precise performance sensitive way described in more detail inSwitching to VolP O Reilly Dropped packets and packet jitter packets arriving out of
201. ded from lib ommotketab ir and Cisco and they come in a variety of hardware flavors too PCI cards rack mountable enclosures and tiny little single line converter boxes that are reminiscent of Ethernetmedia converters Though these interfaces are self contained standalone devices they tend to be called media gateways or just gateways For IP phones and Internet based connections to the phone company there is no FXO FXS vernacular and no legacy signaling involved at all When there s no legacy signaling it s calledpure VoIP 4 1 1 2 And Then There Was T1 Digital circuits that employ the T1 carrier the most widespread type of digital telecommunications connectivity can also be used to connect legacy phones to the Asterisk server and to connect the Asterisk server to central office switches Depending upon your needs and on what is available from your phone service provider you might employ aprimary rate interface PRI to hook up to 23 phone lines at a time to an Asterisk PBX all on a single T1 Likewise you can connect a T1 to an FXO interface box a media gateway to connect analog phones to the server or you can connect a T1 to a device called a channel bank to connect 24 legacy analog phones or analog phone lines 1 In Europe T1s are called E1s use a different voice codec and have 30 phones lines I ve chosen hacks that will let you experiment with Asterisk while avoiding the relatively high cost and management ov
202. different buil systems and methodologies found and stuck with a wonderful combination of crosstool and PTXdist After some serious time and effort AstLinux was reborn and this time it came in at just under 27 MBsmall enough to fit on a 32MB CF card This hack will show you how to use AstLinux to create asoftPBX system that doesn t require a hard drive Read the next section to find out the kinds of features you can fit into such a tiny system 7 9 1 Current Features of AstLinux As of this writing AstLinux has the following features e DHCP server client e File transfer protocol FTP server e IFIP server e Asterisk with zaptel and libpri e Sangoma WANRouter with voice time division multiplexing TDM support e Web server with HTTPS downloaded from lib ommolketab ir downloaded from lib ommotketab ir e Administration via console serial console SSH or web graphical user interface GUI e Network time protocol NTP client server e VPN support IPSEC IKE and OpenVPN e SPI firewall iptables with my astfw script e Quality of Service QoS my AstShape script e NFS client server e Linux 2 6 kernel e Caching DNS proxy server dnsmasq Additionally AstLinux now runs on everything from the Soekris net4801 PC Engines WRAP series of Single Board Computers SBCs to Dell rack mount gear Pretty much any modern machine using PC hardware is now supported by the AstLinux i586 image 7 9 2 AstLinux s Keydisk
203. downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 43 Connect a Phone Line Using an FXO Gateway 1x43 The easiest way to interface Asterisk to a standard phone linelike the traditional phone line in most homesis by using a telephone media gateway There are essentially two ways to connect a traditional non IP phone devicebe it an analog phone a digital phone interface or a telephone lineto the Asterisk system The first way is via theZaptel telephony framework a driver standard that permits telephony interfaces to be used with Asterisk Oddly Zaptel s inventor Jim Dixon named his creation after the early 20th century Mexican revolutionary Emilano Zapata Zaptel engineering is a very deep subject that warrants its own hack Hack 44 and even its own book such as O Reilly s Asterisk The Future of Telephony So we re going to start with the other way of connecting non IP phone devices to Asterisk via media gateways Once you ve got this down you can move on to the wonderful world of Zaptel A media gateway is a device that offloads the responsibility of hardware interfacing from the server It converts non IP signaling into VoIP signaling and vice versa Media gateways don t need driver frameworks like Zaptel to support connecting phone lines or other legacy technology They come ready to install on the network with no software to compile Just plug configure and go Connectinc legacy phones
204. dware If none of the previous solutions turns your crank there s yet another way you can use your classic rotary dial phone with modern telephony services like VolP and your tone dialing only phone company If you have Yahoo Widgets formerly Konfabulator you can actually do tone dialing via your rotary phone with the help of your Windows PC or Mac Download and install Harry Whitfield s totally leet DTMF Dial widget http www widgetgallery com view php widget 35922 By taking your old school phone off the hook and holding its mouthpiece up to your computer s Speakers while dialing on the widget the DTMF tones this widget generates will be sent through the phone line to the phone company or to your VolP ATA if you use a VoIP service J ust make sure the volume on your computer is up high enough for the mouthpiece to pick up the sound of the tones Joel Sisko downloaded from lib ommolketab ir downloaded from lib eommoltketab ir Chapter 6 Navigate the VoIP Network Section 6 1 Hacks 7287 Introduction Hack 72 Monitor VoIP Devices Hack 73 Inspect the SIP Message Structure Hack 74 Audit a Network s QoS Capabilities Hack 75 Graph Latency and Jitter Hack 76 Explore NAT Traversal Hack 77 Shape Network Traffic to Improve Quality of Service Hack 78 Create a Premium Class of Service Hack 79 Build a 100 PSTN Gateway in 10 Minutes or Less Hack 80 Make IP Phone Configuration a Trivial Matter Hack 81 Peek I
205. dwidth making it a good choice for lag sensitive gamers Teamspeak offers clients and servers for Windows and Linux They re fully interoperable with each other Like Ventrilo Teamspeak requires DirectX 8 1 or later and its designers insist on a pair of headphones to reduce echo Also offered is a hosted pay for play service based on the software Though not officially sanctioned by the designers of Teamspeak a great Mac client called Teamspeex has been developed You can download it fromhttp www savvy nl blog download 2 14 3 Skype Perhaps the easiest way to VoIP while fragging is withSkype This desktop voice chat package Supports conference calls with five participantsperfect for maintaining open communications for a marauding patrol of Halo warriors behind enemy lines Skype has several things going for it it has handsfree operation making it ideal for gaming it s fully cross platform even Linux has a client and it s stable But since only five conference participants at a time are permitted in Skype four plus the person hosting the conference you aren t going to accomplish your entire 500 personEverQuest guild meetings using Skype That kind of scalability is something you d need Teamspeak for For small conferences Skype is adequate Perhaps it s no coincidence that the maximum size of a quest group in World of Warcraft is five membersideal for a Skype conference J ust be sure the host PC has plenty of horsepower and a sol
206. dy to proceed Surf over to http www kriscompanies com Go to Downloads and then Asterisk and then locate AstShape Provider Download it to your machine place it somewhere in yourspathH like usr local sbin make it executable chmod x astshape provider and optionally change the name to something that you will remember Let s take a look at the script shall we 6 8 1 Get Started with AstShape Provider Open AstShape Provider in your favorite text editor If you have ever seen AstShape you will notice that AstShape Provider is actually smaller and simpler That s because we are assuming that all this router will handle is VolP traffic There are no provisions for handling other types of traffic and as the script says you will want to block this traffic withiptables or some other firewall There are four possible knobs to turn and they look conspicuously like those in plain vanilla AstShape This is the speed in kilobits of your Internet connection This value can be best determined by testing and testing often This will be the hardest part LINKSPEED 1000 This is the wide area network WAN interface on which to do QoS DEV ethl What you have here is a list of ports separated by spaces that will be placed in Class 1 This is the premium more important class of service I ve chosen 5000 and 5001 for mypremium class s ports Class 1 priority ports CLASSIPORTS 5000 5001 downloaded from lib ommolketab ir
207. e don t recommend either of these approaches however because neither o them provides a straightforward way of telling Asterisk where to send the fax from the outside application or script that s handling the scanning and packaging Without a lot of hacking Asterisk just doesn t make a good day to day occasional use fax server for outbound fax transmittals There are better solutions to this need already One of them isHylaFAX a freely available fax server for Linux and BSD operating systems HylaFAX can use standard fax modems which also makes it cheaper to implement than Asterisk with comparatively expensive Digium voice cards You can obtain HylaFAX fromhttp www hylafax org downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 91 Build an Inbound Fax to Email Gateway ot Once you start faxing with your Linux box why stop there This hack shows you how to route faxes automatically into emails and PDF files In the previous hack you built a configuration to direct all incomingfaxes from Zaptel channels to a file which in turn you could automatically print But if the server were working on behalf of many possible fax recipients you would have to rely on the incoming fax s cover sheet to know which recipient it s destined for Worse still someone would have to go to the printer pick up the fax and hand deliver it to the correct person There s a better way of course email It s just as e
208. e dimensional emoticons involving a cartoon character that has several characteristics that you can manipulatehair shirt and pants color and gender though it appears you ll be stuck with red shoes no matter what The coolest part about 3D Avatar Messenger is how it displays your character in the Same window as your conversation partner s character allowing you to interact with him The application is limited to two participants at a time of course and it s the most fun to use during a Skype voice conversation To install it unpack the zip file you ll find at the URL mentioned earlier and execute itsun bat file either by double clicking or executing at the command prompt This will launch the J ava interpreter and allow the program to run Once it s up and running you ll need to find a partner who also has it installeda husband or wife will work well hey it works for me Otherwise you ll be emoting by yourself as in Figure 3 15 Figure 3 15 Skype 3D Avatar Messenger enhances VolP calls with an animated alter ego 3 13 2 Sometimes There s No Substitute for Video If you want to use your webcam to enhance your Skype calls withreal time video just like the video viewer screens on Star Trekl l only lower resolution point your browser to two of the coolest video on Skype plug ins 1 Star Wars holograms look cooler than Star Trek video screens but I m afraid there aren t any hologram plug ins for Skype yet downloaded from li
209. e electrical circuits provide the power If a circuit blows or the electrical supply fails you won t be able to make any calls This would also be the case if your Internet connectivity failed or experienced a VolP prohibitive downloaded from lib ommolketab ir downloaded from lib ommotketab ir traffic jam You wouldn t be able to make calls or you might not be able to hear or be heard Neither would be acceptable in anemergency calling situation yet broadband VoIP TSPs can t prescribe a solution to this problem This is because the TSP doesn t control the traffic between your Vol Pdevice your ISP and the rest of the Net that provides the data transport between your VoIP device and the TSP Unfortunately there aren t many solutions to these issues 1 8 2 Hack a Compromise Solution In the event of an emergency you re going to want to know you can pick up the phone and reach help quickly You can do a few things to ensure this 1 8 2 1 Keep a Plain Old Telephone Service POTS line for 911 calls By keeping a traditional phone line hooked up you ensure that you can reach 911 using the old phone and you provide a line that your VolP ATA might be able to use for 911 dialing Many VoIP ATAs and VolP integrated broadband routers such as the Zoom X5V and V3 routers allow you to connect a standard POTS line that 911 calls can be routed to in case of an emergency Check with your VoIP TSP to see if it Supports this kind of conne
210. e Using Zaptel Hack 45 Forward Your Home Calls to Your Cell Phone Hack 46 Selectively Forward Calls Hack 47 Report Telephone Activity with Excel Hack 48 Kindly Introduce Telemarketers to Mr Privac Hack 49 Build a Four Line Phone Server Hack 50 Master Music on Hold Hack 51 Record Calls Hack 52 Get Your Daily Weather Forecast from Your Telephone Hack 53 Put a Happy Face on Asterisk Using AMP Hack 54 Run Asterisk Without Root for Security s Sake Hack 55 Link Two Asterisk Servers with PSTN Hack 56 Link Several PBXs over the Internet Hack 57 Route Calls Using Distinctive Rin Hack 58 Tune Up Your Asterisk Logs Chapter 5 Telephony Hardware Hacks Section 5 1 Hacks 5971 Introduction Hack 59 Record Calls the Old Fashioned Way downloaded from lib ommolketab ir downloaded from lib eommoltketab ir Hack 60 Make IP to IP Phone Calls with a Grandstream BudgeTone Hack 61 Build a Custom Ringtone for Your Grandstream Phone Hack 62 Tweak Your Sipura ATA Hack 63 Build a Bat Phone Hack 64 Brew Your Own Zaptel Interface Card Hack 65 Build a Speed Dial Service on Cisco IP Phones Hack 66 Power Cisco Phones with Standard Inline Power Hack 67 Customize Your Cisco IP Phone s Boot Logo Hack 68 Configure Multiple IP Phones at One Time Hack 69 Customize Uniden IP Phones from TFTP Hack 70 Control the Lights Using Your IP Phone Hack 71 Use a Rot
211. e converted all your sound files create a directory calledvm sounds in var lib asterisk Ssounds and copy the files into it 7 11 2 Motion Detection Code set up a motion detector right under my desk in my office use an excellent home automation package called MisterHouse http www misterhouse net to monitor the motion detector know that if there is no motion in my office between 8 a m and 9 a m am probably not going to be in the office that day Here s the code file give to MisterHouse to write a file if I m in the office betwee those times Soffice movement new X10 _Item Al Al is the X10 code my Soffice_ presence _start 8 00 AM my Soffice_presence_end 9 00 AM my Soffice_presence_file office presence txt if time_now Soffice_presence_start reset the file unlink Soffice_presence_file if state_now Soffice_movement eq ON and time_greater_than Soffice_presence_start and time_less_than Soffice_presence_end open PRESENCE gt Soffice_presence_file print PRESENCE time close PRESENCE Save the previous code in a file and place it in the MisterHouse code directory You ll need to reload MisterHouse to have it start using your code 7 11 3 Dialing Greeting Code Here is the code use to control the creation of my dialing greeting Note that it requires the Asterisk AGI Perl module usr bin perl use strict use Asterisk AGI downloaded from lib ommolketab ir
212. e entering digits On the Sipura line of ATAs this is controlled by two more parameters that you should be familiar with They are called I nterdigit Short Timer and Interdigit Long Timer and you can find them on the Regional tab Interdigit Short Time specifies the delay in seconds for sending numbers that match a string found in thedial plan Interdigit Long Timer specifies the delay also in seconds for sending numbers that do not match the dial plan Here is an example Line 1 Dial Plan 7xxx Interdigit short time 3 seconds downloaded from lib ommolketab ir downloaded from lib ommotketab ir This means that when dial 7104 theSipura will send that number to the remote SIP server 3 seconds after press 4 If were to dial 2627638123 the Sipura would send that number to the remote SIP server 10 seconds after entered 3 because there is no pattern matching that number Let s take a look at a more complete example Line 1 Dial Plan 2 ee a ee eee a ee a ee a ee Pa ee ee Interdigit short time 3 seconds Interdigit long time 10 seconds This example matches NANPA 7 digit 10 digit and 11 digit dialing It also includes NANPA internationa dialing as well as matches for three and four digit extensions beginning with 7 This way most Standard dialing as well as extension dialing will be covered by this dial plan thus matching the Interdigit Short Timer of 3 should point out that if you want a number dialed
213. e facts when most IP phones boot they look forconfiguration files on a nearby TFTP server and download them to configure the phone further Hack 80 The configuration files allow the Specification of a logo along with other tweakable goodies By editing or adding thelogo_url setting in a phone s configuration file you can dictate which logo the phone should use The storage or location for this logo varies depending on the version of the firmware that s loaded on your phone but you should be able to specify a standard HTTP URL to point the phone toward its logo This kind of hack is also possible on other phones so look it up Here s the specific setting for a Cisco configuration file logo_url http domain cisco logo bmp As you can see the URL points to a bitmap picture logo bmp So far this looks to be a simple hack and with a few notes it will stay that way The image that the phone downloads is on a server so it needs to find the server somehow In other words make sure the phone s DNS server setting is right so that it can resolve the hostname you provide in place of thedomain place holder in the URL The image size and color are also important If you use an image with the wrong size or aspect it wi come out looking a bit funky on your Cisco s LCD The default size for the display on the Cisco 7960 133 x 65 pixels so that would be a good place to start The image should be monochrome at least for the 7960 Color is Suppor
214. e good ways people are hacking and pass the hacker ethic of creative participation on to the uninitiated Seeing how others approach systems and problems is often the quickest way to learn about a new technology Since it is based in software VoIP is overflowing with hack potential If you love to tinker and optimize this technology offers a cornucopia of exciting things to tweak and customize As in the heyday of the World Wide Web fortunes will be made in the nascent VoIP industry and lots of fun will be had by voice hackers like you and me downloaded from lib ommolketab ir downloaded from lib ommotketab ir How This Book Is Organized You can read this book from cover to cover if you like but each hack stands on its own so feel free to browse and jump to the different sections that interest you most If there s a prerequisite that you need to know about a cross reference will guide you to the right hack The book is divided into seven chapters organized by subject Chapter 1 Broadband VolP Services In this chapter you ll be introduced to some Internet based VoIP phone service providers who can help you replace your traditional phone line with a cost saving feature rich VoIP line Chapter 2 Desktop Telephony Since VoIP is rooted in software it has some wonderful uses on your desktop PC or Mac In thi chapter you ll learn how to customize and maximize productivity enhancing telephony applications Chapter 3 S
215. e handset waiting for your contact to answer 2 2 2 BroadVoice Users Can Use a Web Based Tool to Place and Manipulate Calls If you re a BroadVoice subscriber you ve got some really coolweb based call management tools at your disposal Thanks to BroadVoice Call Manager a web based tool that BroadVoice gives you access to when you sign up you can use a web page to control your voicemail enable and disable call forwarding and even tell BroadVoice how to handle your incoming calls based on their caller IDsmaybe you want to forward certain callers to one number while allowing your BroadVoice voicemail to handle other callers Nifty eh 2 2 3 You Can Automatically Dump Unwanted Girlfriends and Boyfriends Using a VoIP Based Service Sad but true Hey if you can get a date using the Web why not dump people the Internet way too VoicePulse a broadband VolP carrier provides the VoIP network framework for a service that will downloaded from lib ommolketab ir downloaded from lib ommotketab ir help you handle unwanted advances like a dating champ You don t have to be a VoicePulse subscriber to use the service though Any phone userVolP traditional or cellcan dump somebody the high tech way Let s say you re at a party and some doofus asks you for your phone number Give the doofus the local number you find at RejectionHotline com http www rejectionhotline com numbers and _ cities php rather than your real number Whe
216. e identification password recovery reconnaissance and literally dozens of other intriguing tasks Cain amp Abel is literally a Swis army knife of handy networking goodies Not least among these goodies is a VolPcall sniffer recorder that s slicker than a wet rock It provides a sortable date and timestamped list view that logs any VoIP calls it picks up during a sniff It assumes that any RIP traffic is VolP and attempts to decode it and record it into a WAV file According to the Cain amp Abel web site http www oxid it cain html the program can decode calls in uLaw aLaw ADPCM LPC GSM iLBC and a host of other codecs Of course it can t interpret any streams that are encrypted so it s still nearly impossible to record a Skype call from another host Cain amp Abel has a ton of password cracking and networking snooping stuff built in SO be sure to abide by the local policy of the network you re working on or you could end up in a heap of trouble 6 16 1 The Easy Way to Intercept Calls To record a call from the local computer where Cain amp Abel is runningthat s the easiest wayinstall the program on a machine with X Lite or a comparable softphone that can place calls in one of Cain amp Abel s supported codecs Of course this technique will only allow calls placed to and fromthis machine It will not sniff out calls between other computers or IP phones Fire up Cain amp Abel Then select the Configuration menu option in
217. e needs to support SIP and Real time Transport Protocol RTP This means sticking with a highly compatible VolP platform like Asterisk or sipX or carefully evaluating a commercial solution 7 2 2 Size and Select a Voice Server If you ve chosen a commercial softPBX like the Avaya Media Server or Cisco CallManager you re pretty much pinned to the sizing guidelines provided by the manufacturer When you build it yourselfon Linux or BSDyou ve got total control over scalability With that said there s really no hardand fast rule for determining how much processor power your server needs I ve run small workgroup 5 to 10 user Asterisk servers on Pentium II machines A downloaded from lib ommolketab ir downloaded from lib ommotketab ir large workgroup with hundreds of phones connected would certainly need a much beefier computer VoIP can be a very light load on a PC or an immense one If you have five SIP phones connected to a single Asterisk server all using the simplest codec uLaw a Linux server with a slower processor Pentium Il or newer and 256 MB of RAM is probably just fine But if you have 50 SIP phones using three or four different codecs and attaching to the Public Switched Telephone Network PSTN you ll need at least a 2 GHz Pentium 4 or equivalent with 1 GB of RAM S P to SIP calls are much less processor intensive than SIP to Zaptel calls Conference calls are more processor intensive than normal two party calls
218. e not set up like this it will probably be to your advantage to renumber As you will see this method has one big advantage unless every extension is globally unique it s more difficult for each Asterisk server to route calls The beginning 8 signifies an internal extension begin with this to standardize on four digit extensions so that an internal extension is readily recognized as being a free internal call as opposec to an outside call to the pizza place This prefix helps Asterisk figure out where to route the callto ar internal user or to the phone company Using the same numbering convention around the world will make your life easier When you bring that new office in Stockholm online you just have to assign it the 84XX range and update your Asterisk servers and the phones around the world will automaticall recognize it as a valid range If you have not already done so let s set up some basicDNS records for this system We are going tc create several A records in our existing DNS zone twidgets com These A records are going to be called chicago twidgets com tokyo twidgets com and london twidgets com They should each point to the static IP address of the Asterisk server at each respective location Once DNS is set up properly verify basic IP connectivity byusing the ping command to each location from each location Ping Tokyo from Chicago and London Ping London from Tokyo You get the drift This is what you should have so far
219. e that doesn t require a complete restart like a change to a certain phone extension you can just US reload at the Asterisk prompt 4 2 2 Linux Specific Start and Stop Scripts Depending on your particular flavor of Linux be it Fedora Debian SUSE or something else you ll find your system s normal startup scripts in a place that s unique to each flavor Fortunately Asterisk s Makefile has an option that lets you automatically generate start and stop scripts that are specific to your flavor of Linux In your Asterisksrc directory just issue the command make config and the scripts will be installed These scripts start and stop not only Asterisk but also the Zaptel drivers if you ve compiled them As it stands at this point your Asterisk server won t be especially useful You ll be able to explore the Asterisk command prompt with asteriskr but the truly fun stuff like hooking up phone lines and phones is still to come To try out Asterisk s cool demonstration routineslike interactive voice response IVR and an Internet based VolP callyou ve got to configure a phone of one sort or anothe to access the Asterisk server Keep reading downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 42 Attach a SIP Phone to Asterisk HACK 42 Asterisk is a phone system But it won t do you much good without some phones connected You re about to use a SIP telephone to access the de facto auto attendant gr
220. ecting to FXO interface card power injector power over Ethernet PoE power supplies PowerDsine presence primary rate interface PRI prioritizing packets 2nd 3rd on Linksys broadband router received packets and RTP traffic privacy check for your phone number PrivacyManager command proprietary signaling protocol Provider DNS Address setting X Lite proxy servers 2nd PSAP Public Safety Answering Point PSTN Public Switched Telephone Network avoiding with pure VoIP dialing interface hardware interfacing your phone line with VoIP network 2nd 3rd linking Asterisk servers with dial plan configuration 2nd Linux PBX that communicates with 2nd downloaded from lib ommolketab ir downloaded from lib eommolketab ir placing receiving calls to and from Skype network PSTNgw Public Safety Answering Point PSAP pulse dialing support by Digium IAXy FXS gateway support by Digium Wildcard TDM400 pure or native Voice over IP pure VoIP pure VoIP dialing between TSPs 2nd 3rd PWLib libraries downloaded from lib eommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M wW K M K QoS Quality of Service auditing for VolP network 2nd 3rd 4th graphing latency and jitter using pathping 2nd 3rd DiffServ implementing where jitter originates improving by shaping network traffic 2nd 3rd 4th 5th 6th prioritizing some types of
221. ed from lib ommotketab ir To have voice communication while blasting the competition to bits you re going to needheadphones and a microphone The mic can be built into the headset or it can be freestanding But definitely us headphones They ll cancel the acoustic feedback you would get if you were using regular speakers and they ll substantially reduce annoying echo Your gamer buddies will thank you There s nothing quite aS unnerving as a VolP enabled four player round of Warcraft III when one of the players is echoing like crazy Friends don t let friends frag without headphones downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 21 Google for Telephony Info iE Harness the world s most Knowledgeable search database for your own voice purposes Near the end of the dotcom boom a littlesearch engine startup called Google was born Today Google dominates search on the Internet Though Google has moved into the realm of VoIP with Google Talk its new IM client the company s best offering totelephony is still its famously useful search engine Google com 2 15 1 Mine for Phone Numbers If you re looking for a particular phone number or for a group of phone numbers to be used in telemarketing or fundraising applications a great place to start is with Google And a really smart next step is the National Do Not Call Registry http www donotcall gov if you plan to solicit the folks you re c
222. ed to a particular port a typical managed Ethernet switch allows you to listen in on traffic on the other portslike the ports where a VoIP call participant is connected On a nonswitched network like a hub you can use the easy way to monitor any device that s connected to the hub Cisco switches use a technique calledPort SPAN to mirror the packets sent or received on one port to another port In this manner the switch administrator can inconspicuously capture all traffic on any port he chooses To record a VoIP call you ll need to set up port spanning between your PC s port and the target VoIP device s port For the moment I m going to assume you re eloquent enough witt Cisco configuration that you can at least get into your switch s command prompt and Enable mode you ve no idea what this means you might want to invest in James Boney s insightfulCisco 10S ina Nutshell O Reilly Let s say the VoIP device we want to record packets from is connected to port 5 on the switch Use this command to mirror packets into what Cisco calls a SPAN Session a place we can retrieve them from on another port Switch config monitor session 1 source interface fastethernet 5 1 Now traffic to and from port 5 is mirrored to SPAN Session 1 Next we need to reflect that traffic to downloaded from lib ommolketab ir downloaded from lib ommotketab ir the port where the sniffing PC is connectedsay port 4 Switch config monitor s
223. eed to dial the full international number GD What phone number are you dialling Enter the full area code and phone number just as you would when dialing with a regular phone 14403281414 About SkypeOut 2 What country is the number in Buy Now Select the country below Dialing Wizard acting Hel amp Dial the number like this in Skype Skypeln BETA Always dial a full international number in Skype Skype Voicemail BETA 49 144032814144 eee L The telephone number skype is not a International dialing for Germany telephony replacement service and cannot be used for emergency dialing Rates for dialing this number Rates listed here are for reference only View your call list after your call for exact charges 0 0 17 per minute i iT rr To convert from euros to the currency of your choice you ve got a couple of options You can go the web based route using a site like http www xe com ucc the Universal Currency Converter Or if you re a Mac OS X 10 4 Tiger user you can launch the handy Unit Converter widget in the Dashboard For more information about the Dashboard and other cool Mac goodies pick up a copy of Mac OS X Tiger The Missing Manual O Reilly Whether it s currency conversion for SkypeOut or just checking the weather forecast widgets are really nifty Windows and Mac users alike should take a look at Yahoo s really cool widget framework formerly Known as Konfabulator at http
224. eeting and to access a brief demonstration of an Inter Asterisk Exchange IAX trunk over the Internet Sound like too much Don t worry most of this is already configured withAsterisk out of the box The toughest part for a VoIP beginner will be making sure Asterisk is willing to answer SIP callsand that s pretty easy and an Internet connection A You won t need a regular phone line for this hackjust aSIP phone Asterisk SIP is one of several standards that allow IP voice endpoints and application servers such as Asterisk to establish monitor and tear down media sessions across the network Asterisk uses SIP to facilitate calls on behalf of SIP based IP phones such as theBudgeTone 101 the Cisco SIP IP Phone 7960 and the Avaya 4602 I ve chosen the BudgeTone 101 hardware because it s cheap but you car go even cheaper and apply this hack using a softphone like the X Lite Hack 4 which is free The Sipura SPA 841 is another excellent low costSIP phone 4 3 1 Configure a Grandstream BudgeTone 101 IP Phone The BudgeTone 101 phone has a Menu key an LCD display and two arrow keys that you use to navigate its configuration menu options DHCP IP Address Subnet Mask Router Address DNS Server Address TFTP Server Address Codec Selection Order SIP Server Address and Firmware Versions called Code Rel on the phone s screen When you get to the option you want you press the Menu key to select it and then you enter the numeric data req
225. eive SIP error messages when trying to dial to or through the Asterisk server This is Asterisk s way of ignoring what it sees as an unauthorized endpoint Unlike traditional PBXs which tend to give network access to any phone connected on an active port SIP servers tend to enforce some securityusually in the form of password authentication So tell the Asterisk server to stop ignoring requests from yourIP phone Asterisk the softPBX refers to IP phones and other SIP devices as channels SIP channels or peers if you like are defined in Asterisk s configuration file etc asterisk sip conf To enable the phone as configured inFigure 4 1 add the following to the end of this file defaultsip type friend context default username 103 fromuser SIP Phone callerid 103 host 10 1 1 103 nat no Canreinvite yes dt fmode info disallow all allow ulaw downloaded from lib ommolketab ir downloaded from lib ommotketab ir The preceding configuration settings add the 10 1 1 103 IP phone that matches the configuration of the Grandstream BudgeTone Take note of the username callerid and host values which resemble each other 103 in this case They don t need to resemble each other however because there s no relation between a phone IP address and its SIP username or caller ID These can all be completely different One of the biggest differences between SIP and its predecessor H 323 is that SIP identifies its phone endpoints or
226. elephone calls from one IP endpoint directly to another without the need for a VoIP call management server This is known asIP to IP calling Since each IP phone has a unique identification characteristic within the scope of the networkan IP addressone phone can call the other by IP address as if it were a phone number Now to dial by IP address All IP addresses are 12 decimal digits long even if preceding zeros aren t visible when notated Conversely the dots that are normally included in a notated IP address are not dialed So on the BudgeTone phones 10 1 1 103 Is dialed as 010 001 U01 103 To dial take the phone off the hook so that you hear a dial tone and then press the Menu key dial the address of your second phone according to the convention just shown and press the Send or Redial button Of course nobody would want to dial 12 digit IP addresses to place phone calls all the time call management servers like Session Initiation Protocol GIP registrars provide more elegant dialing conventions However dialing by IP address does allow you to circumvent call management and make a direct VolP connection between two endpoints When the receiving phone rings have somebody answer the call If you can hear them talk through your IP phone s handset you ve just made your first successful VolP phone callsort of the IP equivalent of Bell and Watson s first phone call back in 1876 If the receiving phone doesn t ring you might have to
227. elephony service providers TSPs get your voice onto the Net allow you to make and receive phone calls just like traditional phone companies and tend to shrink your phone bill to boot Some of these service providers give you a basic free service that enables you to call other users over the Internet Others allow you to make toll free calls free of charge but charge for local and long distance calls TSPs that allow you to call traditional telephone service subscribers do so by connecting your standard home phone to the Net Some TSPs also let you use a special piece of software called a softphone to place calls with your PC To get connected to a TSP you need a broadband Internet router configured as a DHCP server a spare Ethernet port either on your router or on a nearby switch and a good old fashioned analog telephone TSPs are data centers with telephony servers that route calls to and from your home network or broadband VolP device The real time packets that carry each call s sound over your broadband link use IP and User Datagram Protocol UDP protocols and the TSP communicates key moments in the calllike dialing connecting and hanging upusingsignaling protocols that are similar in some ways to the ones your browser uses to surf the Web The VolP device that most TSPs provide to connect your home phone is known as ananalog telephone adapter or ATA These little boxes allow you to connect a residential style analog phone tc your broa
228. ementary Function a Outbound Proxy FODN i gq 83 234 24 FRO Interface ie NAT Traversal OiP Se Channel User ID ___ Display Name yrit 7712 Support Tii _ 7712 SS M O O Authentication 1D 7711 if 12 Password RTP Port 15000 _ 15104 Register Expires sec 3600 SIP Local Port 5060 Voice Codec G 7Li Mu Law e DTMF Transmission peczes3 INFO C Inband e Use VAD enable disable Use Echo Canceller onable disable Jitter Buffer Size ms 50 a hi 000 Apply Changes Save save and restart Let s assume you ve already set up the TCP IP basics on yourAsterisk server machine and on your media gateway Make sure you can ping back and forth between them too Then accessing the VolF configuration page on the Clipcomm enter the Asterisk server s IP address into the SIP Server field Enable SIP Registration and enter the Asterisk server s IP address again into the Registrar and Outbound Proxy fields This will cause the media gateway s SIP client to register with the Asterisk server Registration is the process by which a SIP client is authenticated with the SIP server and is also the means by which the SIP server knows how to reach the endpoint in case it needs to route a call to it Just as a TCP IP device can register with a DHCP server a SIP client can register with a SIP server called a registrar Now you ll need to tell the media gateway whatcredent
229. ength 20 G711MuJitterBufferLength 10 G711ATxPacketLength 20 G711AJitterBufferLength 10 G729TxPacketLength ZO G729JitterBufferLength 10 The Uniden s three supported codecs are G 711 muLaw G 711 aLaw and G 729 The G 711 codecs are standard 64 Kbps PCM bitstream codecs that mimic the sound framing technology used by the legacy time division multiplexing TDM equipment on the public telephone network G 729 is a high compression codec that requires about half the bandwidthof G 711 These settings allow you to tweak the packet length in milliseconds of each codec Adjusting the packet length also called the packet interval changes how large each sound packet will be Shorter lengths will yield smaller packets but will require greater bandwidth because they incur more Ethernet and IP overhead For a great description of how packet intervals and overhead interact if do say so myself pick up O Reilly s Switching to VolP The jitter buffer settings tell the UIP200 how many milliseconds of sound data to record before transmitting to overcome the commonplace network instability known as jitter In all reality you might not need to touch any of these settings though it is certainly fun to toy around with the jitter buffer length if your wide area network WAN link is particularly jittery DiffServMode OFF DefaultDiffServParam 192 RTPDiffServParam 160 downloaded from lib ommolketab ir downloaded from lib ommotketab ir DiffServ
230. entswith a few gotchas One of the most popular downloads and purchases for cell phones are customizedringtones This feature is available for Skype too with none of the proprietary nonsense pushed on you by your cell phone carrier you can use just about any WAV file as a Skype ringtone On a Mac an AIFF file will suffice too But not just any WAV file will make Skype happy stereo WAVs won t work as the WAV needs to be monaural one channel Thankfully you can use a common recording tool to convert your stereo WAV file so that it can work with Skype If you ve got SOund eXchange SoX installed see From Podcasting toSkypecasting Hack 36 to find out where to get the Windows version on your Windows Mac or Linux machine you re read to convert files to mono C gt sox stereo file wav 1 mono_file wav SoX s c option knocks that stereo file down to one channel perfect for use with Skype If you d like to convert an MP3 file into a mono WAV file just specify the WAV file extension for the destination file s name C gt sox stereo_file mp3 el mono _file wav Drop the file created by SoX into a directory where you can access it from Skype and pull up Skype Options dialog To do so on Windows select Options from the Tools menu in Skype Figure 3 13 To do so on a Mac select Preferences from the Skype application menu igure 3 14 Figure 3 13 Skype for Windows Sound Alerts options dialog downloaded from li
231. ep you busy for a while New VolP projects arrive weekly at SourceForge net so this chapter represents only a partial smattering of what s out there There s no question that the open source world is a voice hacker s paradise a realm of mission critical high stakes real time applications with very little tolerance for underperformance and SourceForge is crawling with new ways to take advantage of real time converged networks for mission critical voice apps Being a voice hacker is sort of like being a network marine You ve got to train hard to spot issues that don t show up in other less loss sensitive kinds of networking You ve got to be first to fight when problems occur on a voice network because voice users will pick up on network slowdowns anc outages before plain old data users It takes thick skin and quick thinking to join the ranks of the voice hacker I ve saved some of the coolest and toughest hard core hacks for the last chapter because wanted to ease you into them You ll need a pretty good understanding of the Session Initiation Protocol SIP Asterisk and Linux to get through this chapter unscathed So if you haven t read the first six chapters it would be a good idea to do so now Have some fun with those hacks and when you re ready to get serious come on back Do you want to build and harden the ultimate PC based telephone server Do you want to master the old VoIP standard H 323 How about building a fax to ema
232. ephony management and call accounting features including voicemail Session Initiation Protocol SIP telephone support dial plan anc so on in a nutshell Asterisk is an all software PBX If you re wondering about these technical terms don t worry As you experiment with Asterisk and learn more about VolP they ll become very familiar For now just compile and install all three packages After you run the CVS download the source code for each Asterisk software component is sitting in its respective directory in usr src Let s compile each software component by issuing the following commands Again you need to compile zaptel and libpri only if you re planning on using legacy or Digium interface cards Many of the examples in this book use legacy devices so it s probably a gooc idea to compile them all right now Here is the sequence of commands cd zaptel make clean make install cd libpri make clean make install cd asterisk make clean make install H HH OH HH Do compile Zaptel before you compile Asterisk or else Zaptel features will be missing from the Asterisk build What is Zaptel you ask Keep reading It should take 20 minutes at most to complete the whole build on an average PC Once built Asterisk is ready to use But you can t race a Ferrari without a training lap on the test track and you can t really use Asterisk until you understand the basics of configuring it So it s time for driving school To get
233. er This applies to VolP very well When you are downloading a large file from a remote web server you will be dealing with fewer very large packets With VoIP you are dealing with many more much smaller packets In fact with some VolP codecs the size of the Ethernet IP UDP header is much larger than the codec payload itself G729 being a good example So VoIP is the Concorde and most everything else is the Boeing 777 What does this have to do with limiting the speed of your connection by 15 Simple By limiting the speed of your connection by 15 we are hopefully ensuring that the FIFO queues outside of your control do not fill up completely Anyone who has ever used VoIP on a cable modem or DSL line knows what happens when someone else using that connection begins downloading a very large file The user on the VolP connection experiences large gaps in audio transmission sometimes lasting several seconds This is because the FIFO queues on your cable DSL modemand on your Internet Service Provider s ISP s CMTS DSLAMfill completely with web traffic and your tiny little VolP packet is at the end of the line Because we can t control these FIFOs like we can our Ethernet interface we have to place a hard limit on the amount of traffic However not all hope is lost If you are anADSL subscriber using Linux you should look into the S518 ADSL board from Sangoma Technologies For around 115 USD you can have an internal PCI form factor ADSL
234. er To access X Lite s configuration settings click the button to the right of the CLEAR buttor on X Lite s main window as shown inFigure 1 2 Figure 1 2 X Lite s main window looks a bit like a cellular phone downloaded from lib ommolketab ir downloaded from lib eommolketab ir When the configuration window appears double click System Settings and then double click Network This to bring up the network configurations Figure 1 3 Find the Provider DNS Address setting and change its value to the DNS server provided by your VoIP TSP not your Internet Service Provider or ISP Your VolP TSP might require the use of its own DNS because itsSIP resources might be on a private domain that cannot be resolved through the public DNS system If your VoIP TSP didn t provide a DNS address you can leave this setting blank Figure 1 3 X Lite s network configuration window downloaded from lib eommolketab ir downloaded from lib eommolketab ir Click the Back button in the lower left corner of the screen to get back to the prior window Here you ll need to double click SIP Proxy to open the SIP Proxy Settings window Double click Default and you ll be able to configure thesoftphone to use a SIP proxy server which is located at your VolP TSP and routes your softphone s calls The X Lite softohone can use more than one SIP proxy but in most situations you ll need to use only one This list describes the setting
235. er of flipping some configuration switches 1 7 1 Prioritize Packets on a Linksys Broadband Router ToS is a feature of Ethernet switches that permits packets tagged as high priority to be handled first maximizing their QoS 802 1p is a similar concept but tends to hang around on routers not switches The Linksys BEFSR81 broadband router is a device that supports 802 1p It sells for less than 100 USD online and you can probably find one secondhand on eBay for even less In fact setting up priorities on this router is a snap thanks to Linksys s usual snazzy web based interface Once you get the router unboxed and hooked up use the web interface to locate the QoS screen You ll see it after you click on the Advanced Configuration button and the QoS tab downloaded from lib ommolketab ir downloaded from lib ommotketab ir The QoS screen contains two sections one that allows you to establish queuing priorities for packets depending on their TCP UDP port numbers and one that allows you to alter the queuing priority depending upon which Ethernet switch port the traffic originated from That is since this router has lt built in switch you can prioritize some of its eight Ethernet ports using the lower half of the QoS screen 1 7 1 1 Prioritize RTP traffic Most VoIP media streams are carried by Real time Transport Protocol RTP packets To raise the priority of RTP traffic enter the port numbers 5004 and 5005 each on it
236. er s server the first time they are powered up Don t interrupt this process If the initial registration is interrupted it could render your ATA useless and the TSP might need to exchange it for a new one Some ATAs wil download firmware patches during the initial registration too Refer to your ATA s instructions for indications on when this process is complete making it safe to power off the ATA Usually if you can hear a dial tone on the connected phone the process is complete and it s safe to place a call or powe down the ATA If you can t hear a dial tone on the connected phone check that it is connected to the appropriate port on the ATA Make sure your broadband router is configured as aDHCP server Without DHCP running on your network the ATA will be unable to obtain an IP address crippling it Many VoIP calling plans require that you dial the full 11 digit phone number even if you re just calling your next door neighbor So if you make a lot of local calls get used to dialing your own area code a lot Once you hear a dial tone it s probably best to investigate any features that are included with your calling planvoicemail especially Then try calling a buddy to see if you can hear any difference between a traditional call and a VoIP one Chances are that the person on the other end won t notice the difference unless you tell him you re on a VoIP call Then he might say he suspected you were ol a cell phone The sound quality on
237. ered with the gatekeeper you can call between them using their E 164 numbers since they re on the same zone Now if you like download OpenAM from theOpenH323 project to set up an H 323 based personal message recorder Figure 7 5 NetMeeting ships with a selection of five codecs including G 711 uLaw aLaw and G 726 ADPCM Advanced Compression Settings z Manually configure compression settings Prefered codec for audio compression Microsoft G 723 1 8 kHz Mono 6400 Bit s Microsolt G 23 1 8 kHz Mono 6400 Bitis Microsolt G 723 1 8 KHz Mono 5333 Bits CCITT u Law 8 000 KHz 8 Bt Mono CCITT A Law 8 000 kHz 8 Bit Mono u F ij j downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 90 Turn Your Linux Box into a Fax Machine 90 Have you ever wished you could handle fax traffic with your Linux machine Asterisk offers a built in fax detection mechanism This allows you to handlefaxes that are sent to your Asterisk box on a Plain Old Telephone Service POTS line connected via a Zaptel interface It s Asterisk s Answer Command that triggers the fax detection If an incoming fax is detected Asterisk automatically transfers the call to the special extension calledfax if it exists To use this special extension you ll need to compile and install thespandsp package Download the latest version from ftp ftp opencall org and unzip the file into usr src spandsp To compile it is
238. erhead of T1s though don t have a T1 in my home or in my business test lab and don t expect you to either Fortunately though lots of other great sources of information about T1 are available For starters check out T1 A Survival Guide O Reilly Then when you re ready to integrate legacy digital telecom into Asterisk check out Switching to VoIP O Reilly and Asterisk The Future of Telephony O Reilly for the details You ll also get a much deeper exploration of Asterisk and enterprise telephony to boot Now let s get hacking shall we downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 41 Turn Your Linux Box into a PBX reat Install and test the Asterisk open source telephony server on your Linux PC Some RPM packages are available to simplify Asterisk s installation but manual compilation is relatively easy So I m going to show you how to download compile and install Asterisk the old fashioned way The development branch you ll download from is stable though once you get comfortable with Asterisk you ll want to jump out on the bleeding edge and try the developer releases too Each release tends to introduce something new and worthwhile even if it s not in the Stable branch yet The easiest place to download the Asterisk software is theCVS repository at Digium the company responsible for Asterisk and some of the hardware components that work with it To access the CVS reposit
239. ers By taking the rport and received addresses of the preceding example we can rewrite the headers correctly to make it appear as though our packet is coming from the externa interface of the NAT firewall sip ping pl v s 66 27 57 228 p 14328 proxy01 sipphone com grep Via Via SIP 2 0 UDP 66 27 57 228 14328 branch z 9hG4bK1116737949 Via SIP Z 07UDP 66 2 57 2262143526 brancn z9nG4bK1 1167379049 downloaded from lib ommolketab ir downloaded from lib ommotketab ir This now specifies the correct external IP address and port in the Via header By specifying the public IP and port we can now traverse NAT even when contacting remote hosts that do not support RFC 3581 This is a great example of how we can get around the NAT problem with SIP Now if only our SIP clientthe phonewere smart enough to employ this technique itself our SIP phone would appear as though it were outside the firewall to the remote SIP devices it calls That s where the STUN protocol comes into play 6 6 1 Get STUNned STUN is an IETF recommendation that allows SIP clients to connect to aSTUN server on the Internet to determine what address and port the connecting client appears to have from behind aNAT host This way the client can send the appropriate SIP Via headers and at least in theory the SIP device it calls will be able to respond by sending packets to the client sNAT firewall Once a SIP device sends response packets to the NAT firewall it
240. erships and geographic moves 7 2 1 1 Creating stability To provide the utmost stability a server s environment must be kept cool with a room temperature of 65 to 75 degrees Fahrenheit It should be kept dry duh and connected to a protected power source with a battery backup Large environments should consider a backup power generator with ar automatic transfer switch The switches and routers that provide connectivity to the server should be on protected power too The server should also be well hardened against potential denial of service attacks which you ll do Shortly 7 2 1 2 Creating high availability To assure that the server is always available and that its voice services aren t affected by internal hardware failures like failed disk drives andpower supplies you should design redundancy into each server A RAID 5 disk array with four or more hard disks provides a hot spare so that you can swap out a failed drive without having to shut down the server you can also consider building a server with no hard disks Hack 95 Redundant power supplies allow the same hot swap ability in case a power supply bites the dust There are other techniques for high availability too like clustering 7 2 1 3 Building in compatibility If you want to future proof your voice server don t bother building it on software that doesn t Support the open standards needed to make it interoperable with other servers Specifically your softwar
241. es To allow this SIP phone to dial out using your newly connected phone lines you ll need to make sure there is a context inextensions conf that looks something like this private phones exten gt _NXXXXXX 1 Dial Zap gl1 EXTEN exten gt _NXXXXXX 2 Congestion exten gt _1NXXNXXXXXX 1 Dial Zap gl1 EXTEN exten gt _1NXXNXXXXXxX 2 Congestion exten gt 911 1 Dial Zap gl1 911 exten gt 911 2 Congestion The string patterns _Nxxxxxx and _1NXXNXXXXXX are actually masks designed to identify phone numbers that are 7 and 11 digits long respectively This way if the dialed number is 7 or 11 digits long Asterisk knows it must dial the number represented by the variables ExTEN using the group of four phones Zap g1 you previously defined in zapata conf The 911 extension performs call routing to the phone company s Public Safety Answering Point PSAP via the standard 911 phone number In countries other than the United States local jurisdictions will use different numbers for this purpose so check with your localemergency dispatch authority to find out what number to use Add a _011x extension to enable international dialing Of course none of this is going to work until the drivers are loaded and the dial plan is reread by Asterisk so give your machine a reboot or load the modules and restart Asterisk manually Then call and be calledon the cheap The coolest thing about the PBX you ve just built is its cost
242. ession 1 destination interface fastethernet 4 1 So now traffic from port 5 will also occur on port 4 where theCain amp Abel PC can sniff it Don t forget a write mem if you want to keep the switch configured this way permanently Now you can use Cain amp Abel and vomit for that matter to record calls that traverse your switched network even if you can t install a recorder on one of the participating VoIP devices downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 87 Intercept and Record a VoIP Call 87 This ts the ultimate sneaky way to intercept and record VolP calls l m going to demonstrate why people say VoIP isn t secure If you ve got a laptop and a patch cable you can record calls from a Cisco CallManager IP telephony network or even from a Vonage subscriber More specifically you ll be doing so without the need for port spanning on the switch Hack 86 and without installing a recorder or sniffer on the device you re trying to record Hack 85 Instead you ll be resorting to a tactic that s well unnatural In fact if you do this outside the test lab it could be considered unethical too so be careful am about to teach you how to secretly listen in on other users VoIP calls without having any direct contact with their VolP phones or PCs and without being the administrator of the local network Be advised though I m not recommending that you ever do this in the field I
243. etab ir downloaded from lib ommotketab ir exten gt 100 1 MusicOnHold 30 exten gt 100 2 Hangup The idea here is that when you dial 100 in this context you ll get 30 seconds of hold music before th server disconnects your call Save the changes to etc extensions conf and go ahead and reboot your Linux box or modprobe ztdummy and restart Asterisk 4 11 1 Hacking the Hack You can assign different groups of phones and phone lines to their ownmusic on hold classes classes define selections of recordings that you can assign to groups of peers so that they hear different music A group of SIP phones can be in one music on hold class and a group of Zaptel connected phone lines can be in another Add as many classes as you like Such aSdefault aS shown earlier to the musiconhold conf file and then point your various Zaptel channels and SIP phones at those classes For Zaptel channels you configure this in zapata conf The first two Zaptel channels are pointed at thestevie Ray class and the second two are in the BB King Class channels language en context default signalling fxs_ks usecallerid yes hidecallerid no callwaiting yes callwaitingcallerid yes threewaycalling yes transfer yes callreturn yes group 1 musiconhold Stevie Ray Channel gt 1 2 musiconhold BB King Channel gt 3 4 4 11 2 Hacking the Hack Some More If you d like to use a streaming MP3 Internet radio station instead of a group of MP
244. etab ir chown chown chown chmod chmod chmod chmod chmod H HHH H H SH OH chmod R asterisk asterisk var run asterisk R asterisk asterisk var spool asterisk R asterisk asterisk dev zap R u rwX g rX o var lib asterisk R u rwX g rX o var log asterisk R u rwX g rX o var run asterisk R u rwX g rX o var spool asterisk R u rwX g rX o dev zap chown R root asterisk etc asterisk R u rwX g rX o etc asterisk You can now launch the Asterisk server from the new user account or from root using the su command su asterisk c usr sbin safe asterisk Finally you ll need to adjust thesafe asterisk script so that it uses the new user account to launch Asterisk rather than root To do so open usr sbin safe_asterisk in your favorite text editor and add su asterisk c before each instance of an asterisk command Be sure to leave the commands unchanged aside from prefixing them with thesu command Once these steps are taken Asterisk will have only as much power as you grant theasterisk user Would be attackers might be able to crash Asterisk but in so doing they won t be able to gain acces to root s credentials downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 55 Link Two Asterisk Servers with PSTN HACK HJI You don t have to have dedicated point to point lines to link two PBX systems J ust use the PSTN phone lines that are already connected to them and y
245. f you ve come this far with Asterisk and thel nternet Phone Wizard you ve unlocked a world of wicked cool hack potential To get you primed for your journey to Aster Skype hackatopia let me show a very simple dial plan modification that will simultaneously ring your incoming phone calls on your locally connected phones as well as on your Skype phone In etc asterisk extensions conf consider the following exten gt s 1 Dial SIP 100 amp SIP 200 This extension will dial the two phones connected on SIP peers 100 and 200 and connect the call to whichever peer answers first But let s say an Internet Phone Wizard is connected to channel Zap 2 Hack 98 Now you can actually dial those two phones and a Skype speed dial alias from your buddy list exten gt s 1 Dial SIP 100 amp SIP 200 amp Zap 2 99 Now whoever is associated with speed dial number 99 in your Skype buddy list will also receive a ca through the Skype network courtesy of the Internet Phone Wizard that you ve hooked up to a Zaptel FXO port Zap 2 on the Asterisk machine So if this was your default context for incoming calls from your home phone line connected to Asterisk via another channel your incoming home phone calls would also ring on the Skype client that was logged in as the buddy in your list Ideally this buddy is a second Skype account you ve set up because Skype doesn t let you call yourself So would set up Ted1 and Ted2 as Skype buddy names and then h
246. for calls interface cards TDM400P card 2nd 3rd 4th team voice chat system Ventrilo Teamspeak for gaming telemarketer calls 2nd telephones analog phones wiring your house phones for VolP 2nd 3rd telephony hardware 2nd 3rd telephony info telephony server 2nd 3rd 4th 5th H 323 gatekeeper 2nd procesor power needed selecting and hardening an OS 2nd 3rd 4th three areas of focus telephony ready sound files Telnet interface 2nd text expressions for happiness and dissatisfaction text message to answer Skype calls text messages TFTP Desktop TFTP servers configuration files for IP phones IP phone configuration 2nd mass configuring Uniden IP phones setting address for SIPDefaults cnf file TIFF file tiff2ps utility 2nd time division multiplexing TDM time shifting effect SoX Time HiRes Perl module TiVo boxes downloaded from lib ommolketab ir downloaded from lib eommoltketab ir ToS Type of Service 2nd using to prioritize class of voice tratfic 2nd TOSC Total Recorder adjusting sound resolution and output format Total Recorder Standard Edition TR16A phone controller traceroute 2nd traditional phone service traffic shaping script AstShape 2nd 3rd 4th transducer pickup Trillian instant messaging client TSPs telephony service providers SIP proxies S IP based using softohone with 2nd tuning up logs Asterisk changing default storage location size of logfiles tweaking Sipura A
247. from lib ommolketab ir downloaded from lib ommotketab ir In its default configuration Asterisk has an auto attendant that can route calls To try it out take the IP phone off the hook and dial 2 Then dial the BudgeTone s Send button You will hear a friendly voice saying Asterisk is an open source fully featured PBX and IVR platform Try this demo while watching the call progress on Asterisk s console by issuing asterisk vvvvvr at a Unix shell before beginning the call While listening to the automated attendant greeting dial 500 This will cause the Asterisk server to greet you connect you to a server at Digium Inc using the Internet and allow you to listen to another automated greetingthe one being played back by a production Asterisk PBX at Digium s office This connection does not use the Public Switched Telephone Network PSTN at all but rather a Voice over IP trunk that is set up on the fly by Asterisk The Voice over Internet demo requires User Datagram Protocol UDP port 4569 If you re using a firewall or NAT device be sure it permits outbound traffic on this port Most broadband routers will permit this type of traffic by default You can also perform an echo test by dialing 600 and you can access Asterisk s built in voicemail service by dialing 8500 This will give you at least some idea of how your voice sounds when it s beer processed and played back for the person on the other end of a call
248. fusing statements and typos that you find anywhere in this book Please also let us know what we can do to make this book more useful to you We take your comments seriously and we will try to incorporate reasonable suggestions into future editions You can write to us at O Reilly Media Inc 1005 Gravenstein Highway North downloaded from lib ommolketab ir downloaded from lib ommotketab ir Sebastopol CA 95472 800 998 9938 in the U S or Canada 707 829 0515 international local 707 829 0104 fax To ask technical questions or to comment on the book send email to bookquestions oreilly com The web site for VoIP Hacks lists examples errata and plans for future editions You can find this page at http www oreilly com catalog voiphks For more information about this book and others see the O Reilly web site http www oreilly com To reach the author of this book Ted Wallingford you can send an email to ted macvoip com Got a Hack To explore Hacks books online or to contribute a hack for future titles visit http hacks oreilly com downloaded from lib ommolketab ir downloaded from lib ommolketab ir Chapter 1 Broadband VoIP Services Section 1 1 Hacks 17 Introduction Section 1 2 VolP Based Phone Service Providers Hack 1 Get Connected Hack 2 Use Pure VolP Dialing with Your TSP Hack 3 Wire Your House Phones for VolP Hack 4 Use a Softphone with a VolP TSP Hack 5 Pr
249. g it all baby The log prefix options allows you to specify what will appear at the beginning of the log entry for eacli packet That way when you comb through lengthy databases of these entries you can find downloaded from lib ommolketab ir downloaded from lib ommotketab ir specifically what you re looking for The following rule is very broad it captures any and all SIP traffi going through the firewall FoRWARD chain and logs it iptables A FORWARD p udp dest port 5060 5061 j LOG log prefix SIP Let s say that you are operating a SIP proxy that facilitates VolP calling via SIP directly to two other proxies Let s also say that all three SIP proxies are in the same organization and that a site to site VPN is used to connect them all The three proxies support three VolP LANs at separate offices The LANs they support have the network addresses 10 1 0 0 16 10 2 0 0 16 and 10 3 0 0 16 aS in Figure 6 13 The configuration examples given in this section are assumed to be running on the firewall in the 10 1 0 0 network Figure 6 13 Three physically separate softPBXs connected by an Internet VPN Assuming a VoIP WAN like the one in Figure 6 13 it s possible to do some interesting logging and prefixing Say you want to log SIP traffic by remote network You can use the following commands tc tag inbound and outbound traffic iptables A FORWARD P udp s 10 2 0 0 16 dest port 5060 j LOG log prefix FromDetroit
250. g to other formats as opposed to GSM is as simple as changing the extension on your final filename downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 26 Sound Like a Pro Announcer 26 Do you want to have a deep commercial sounding voice on your greeting messages Here s how You might think it takes a ton of natural talent to sound like one of those professional deep voiced commercial announcers that you hear tn the on hold messages of big companies But all it really takes is adherence to a few simple guidelines for clear soeech and maybe a little hacking talent First the speech guidelines Commercial announcers usually speak much more slowly and concisely than would be appropriate for normal conversation As a result they are very easy to understand and follow Radio announcers often place almost lopsided stress on the words they want to emphasize too So if your recorded announcements are sales pitches you ll want to speak concisely slightly more slowly than normal and with great stress on the words you d like to emphasize You ll also want to smile while you record yourself It seems odd at first but a smiling announcer conveys a different attitude than a blank faced one A high school or college speech textbook might be a good source if you really want to hack your speaking ability Now on to the hack By now you ve probably used SoX to convert files from one format to another but did y
251. ge and tomorrow s global voice network is the Internet This book contains only a small subset of VolP knowledgeenough to serve as an introduction to the world of VolP and teach you how to use it to save money be more productive or just impress your friends My friends love my on hold music when they call my house love that when people call my house the call is connected to my notebook PC via Skype no matter where am in the world You ll learn how to do all of this and more hope this book gets your mental gears turning and that your VolP hacks are as enjoyable to implement and customize as they were for me to write For more VoIP theory and detailed reference information about Voice over IP check out these great O Reilly titles e Switching to VoIP Skype Hacks Talk Is Cheap Asterisk The Future of Telephony Practical VolP Using Vocal downloaded from lib ommolketab ir downloaded from lib ommotketab ir Why VoIP Hacks The term hacker has a bad reputation in the press They use it to refer to someone who breaks into systems or wreaks havoc with computers as their weapon Among people who write code though the term hack refers to a quick and dirty solution to a problem or a clever way to get something done And the term hacker is taken very much as a compliment referring to someone as being creative having the technical chops to get things done The Hacks series is an attempt to reclaim the word document th
252. gestion is usually signaled to the user as what Is called fast busy If you have ever left a POTS phone off the hook for too long you have heard a fast busy 4 17 2 Adding the Remote Locations Save extensions conf and reload Asterisk with asterisk rx reload If the servers in Tokyo and London have been set up with those extensions go ahead and try calling them They won t be able t call you back yet but you should at least be able to verify that you now have direct dial around the world for free You should now repeat this process on yourAsterisk servers in London and Tokyo For brevity s sake will give abridged versions of the preceding instructions for the Tokyo and London offices For Tokyo edit etc asterisk extensions conf and add the following to your internal context downloaded from lib ommolketab ir downloaded from lib ommotketab ir exten gt _81XX 1 Dial IAX2 guest chicago twidgets com EXTEN 20 exten gt _81XX 2 Congestion exten gt _83XX 1 Dial IAX2 guest london twidgets com S EXTEN 20 exten gt _83XX 2 Congestion Save extensions conf and reload Asterisk with asterisk rx reload For London open etc asterisk extensions conf and add the following to your internal context exten gt _81XX 1 Dial IAX2 guest chicago twidgets com EXTEN 20 exten gt _81XX 2 Congestion exten gt _82XX 1 Dial IAX2 guest tokyo twidgets com EXTEN 20 exten gt _82XX 2 Congestion
253. gnaling in Asterisk slang Hence our switch will be pretending to be a phone by using FXS signaling on its FXO interfaces Confused yet No worriesonce your server is up and running and your TDM400P is functioning as planned you ll probably never need to mess with it again Now that you ve saved zaptel conf bring up etc asterisk zapata conf in a text editor and make sure it resembles this example precisely channels language en context default signalling fxs_ks usecallerid yes hidecallerid no callwaiting yes callwaitingcallerid yes threewaycalling yes downloaded from lib ommolketab ir downloaded from lib ommotketab ir transfer yes callreturn yes group 1 channel gt channel gt channel gt A w N e Channel gt This block of config fine tunes the settings of the four Zaptel channels that are provided by the card for a concise description of what all of these settings do check out Chapter 17 of O Reilly s really amazing book Switching to VoIP Save this file before proceeding 4 10 1 Set Up Incoming Calls Now pay a visit to etc asterisk extensions conf Here you ll need to adjust the default context section of the file so that incoming calls on the four lines can be handled appropriately Take a look at this sample default context section inextensions conf which deals with incoming calls from the TDM400P connected phone lines and from any other channels that point to the default context
254. go headphones if you keep your transmissions brief so as to discourage echo 2 14 1 Ventrilo Ventrilo from Flagship Industries http www ventrilo com is a team voice chat system that uses the Global System for Mobile GSM codeca very bandwidth conservative codec that s excellent for use with games you don t want your voice traffic to create in game lag so a codec like GSM is perfect Ventrilo has client and server components The client runs on Windows and Mac OS X and the server runs on Mac OS X Windows and Linux A version of the client for Linux is said to be in development To run the Windows client you ll need DirectX 8 1 or later available from Microsoft and standard with Windows XP and above The Mac client requires OS X Version 10 3 2 or higher You ll also need a microphone and a pair of headphones the headphones are superior to using freestanding Speakers because ambient noise from the speakers will spill into the microphone creating really annoying echo for your game playing buddies 2 14 2 Teamspeak Teamspeak http www goteamspeak com is similar in purpose to Ventrilo though its web based downloaded from lib ommolketab ir downloaded from lib ommotketab ir chat room administration tools are more advanced and its bent towardgaming is a lot more obvious Ventrilo professes to be useful for other things in addition to gaming Teamspeak uses theSpeex codec which like GSM is very lean on ban
255. group of enterprise VolP servers or just as a gateway router for a segment where VolP is used a lot of VolP related events can be monitored and logged Logging from the firewall is useful for the security minded but it s important for other reasons too It lets you get a feel for which remote networks and hosts are communicating with you VoIP services and how often they are doing this This will improve your understanding of bandwidth consumption and traffic patterns on your network besides giving you a keener awareness of security 6 14 1 Logging with NetFilter NetFilter s default configuration provides for no logging If you want a particular type of packet loggedsay from a specific network or on a specific portyou must tellNetFilter to log it When a packet is logged its pertinent information is sent tosyslog to be stored syslog is the system wide logging daemon that is a staple in most Unix variant operating systems Logging packets using NetFilter doesn t save the contents of the packets ust information from the packets headers If you want to capture packets you ll need other software like Network Associates Sniffer or the open source tool Snort To enable logging you must set up a rule that specifies which packets you want to log The following rule says to log all packets sentto the machine running the NetFilter firewall keep in mind that this will eat up tons of storage space fast iptables A INPUT j LOG log prefix Lo
256. gt lt Name gt Larrys Latkes lt Name gt lt Telephone gt 18005551212 lt Telephone gt lt DirectoryEntry gt lt DirectoryEntry gt lt Name gt Timbos Nacho World lt Name gt lt Telephone gt 18665551212 lt Telephone gt downloaded from lib ommolketab ir downloaded from lib ommotketab ir lt DirectoryEntry gt lt DirectoryEntry gt lt Name gt Drunken Cow Steak House lt Name gt lt Telephone gt 18775551212 lt Telephone gt lt DirecgtoryEntry gt lt CiscoIPPhoneDirectory gt Look at all that s changed You are now using the lt CiscoIPPhoneDirectory gt tag to show the dial and other buttons at the bottom of the screen You also are now uSing lt DirectoryEntry gt rather than lt MenuItem gt and lt Telephone gt rather than lt URL gt To complete the hack edit the work xml and play xml files Andrew Latham downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 66 Power Cisco Phones with Standard Inline Power HACK 66 To avoid lock in with Cisco only phones and switches learn how to power Cisco phones from non Cisco switches IP phones can be powered through their Ethernet connections The standard for this inline power is called 802 3af and many equipment manufacturers support itexcept for Cisco which uses its own proprietary inline power method Because of this you can match Cisco IP phones only with Cisco powered switches unless you use Cisc
257. guration we are creating a range of extensions that will map into a range of voice mailboxes At this point that range is 80008999 8xxx If this does not match what you have or need change it now as we will be usinc this variable throughout this hack Underneath default we are telling Asterisk to include the separate section vmserv We are also defining what to do when a call goes to an invalid extension or times out hang up on them The vmserv context is where the magic happens We are using the variables vmBasE to create a range of extensions We are also telling Asterisk that when we get a call for one of those extensions we should put that call into the voice mailbox of that extension which has the same number We will play back the unavailable greeting from that mailbox and hang up on the caller when he is finished leaving a message downloaded from lib ommolketab ir downloaded from lib ommotketab ir So how do we retrieve these voicemails Simple all you have to do is call into the Asterisk system and add a 9 before your mailbox number So if your mailbox number is 8000 extension 8000 will allow callers to leave a message in mailbox 8000 To check mailbox 8000 you will call extension 98000 There are many ways to do this and suggest that you look into Asterisk substrings and extensions conf to get a better idea But for now save extensions conf because we are done here 7 10 1 Create the Voice Mailboxes N
258. gure 7 7 Here you ll want to enter your friendly organization name your server s DNS domain name a simple authentication realm name of your own choosing and an alphanumeric PIN that will later serve as your administrator password on sipX Click Submit and after a few moments you should be looking at the standard login prompt Use the username superadmin and the password you established on the PIN prompt in the preceding page Click Submit and watch as sipXconfig loads the administrative GUI This is where extra RAM and a fast processor will really come in handy Unlike Asterisk and the Asterisk Management Portal which use Perl and PHP and run swiftly on a minimally configured machine sipXconfig is a highly sophisticated set of Java applications that really call for heavy duty server hardware A Pentium 4 PC with at least 512 MB of RAM should be sufficient to run sipXconfig at an acceptable pace Figure 7 7 sipX s initial configuration screen downloaded from lib ommolketab ir downloaded from lib eommoltketab ir x e600 Configuration Server lt a J R fat http 10 1 1 10 8080 pds ui install OG z HowTo install sipX on Fedora Cor Configuration Server sipXconfig n f Sat Aug 20 22 59 Configuration Server Install Organization Organization Name Kelly Systems DNSDomainfpbx CS Authorization Realm pbx PIN i _ Confirm PIN peee Fil
259. h Now you re ready to label the column headings according to their purposes Insert a blank row at the top of the spreadsheet and you can label them as outlined in Table 4 1 Figure 4 5 The second step of the Text to Columns Wizard breaks up the CDR log text into meaningful cells of data downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Convert Text to Columns Wizard Step 2 of 3 This screen lets you set the delimiters your data contains You can see how your text is affected in the preview below Delimiters Treat consecutive delimiters as one Tab Semicolon Comma Text qualifier 7 5 C Space C Other gi2167711111 default 164 SIP 1019 2p S12162782111 default 104 SIPI zop 1 1 hoe ITED 7A Zop i 1 SIP ied home T Hal ingfind lt 449 Sedii oh SIPG default 162 SIPG SIPOH i default 163 SIP IG3 5123 SIPOH Table 4 1 Asterisk CDR default fields CDR field Description Account code A tag that can be used in billing and analysis Source The unique identifier of the endpoint placing thecall Destination The unique identifier of the endpoint receiving the call Context The dial plan context of the call more on this later The calling party identification signals supplied by the calling Caller ID endpoint Incoming Channel The voice channel that routes to the caller Destination Channel The voice channel that routes to the receiver
260. h cable with two RJ11 connectors FWD Free World Dialup FXO foreign exchange office channel Digium X100P FXO card gateway connecting phone line to Asterisk 2nd 3rd 4th gateway device connecting analog phones to Mac mini PBX interface card 2nd modifying FXO channel for pulse dialing modules on TDM400P card FXS foreign exchange station Digium IAXy FXS gateway downloaded from lib eommolketab ir downloaded from lib ommotketab ir interfaces downloaded from lib ommolketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S W U M wW K M K G 711 codec ames GarageBand gateways Gauge It Gizmo Project genkd script Gizmo Project 2nd 3rd 4th 5th 6th conferencing limitations downloading and installing client extra features placing a call support for SIP Global System for Mobile GnoPhone 2nd GNU Gatekeeper gnugk GnuGK Google images Google Groups Google Talk Gotolf command Gotta Go widget Grandstream BudgeTone making IP to IP_ phone calls 2nd mounting on the wall Grandstream IP phones 2nd 3rd graphical user interfaces GUIs Asterisk web based sipX GUI raphing jitter and latency data RRDtool 2nd greeting messages different message each day mixing different sound files for voicemail 2nd greeting script 2nd GSM Global System for Mobile codec 2nd 3rd downloaded from lib ommolketab ir
261. happening again the customer asked I m glad you asked she replied You can download the latestfirmware patch for our ATA which Should make the ATA automatically reregister with our server whenever it loses communication That would be the best thing to do downloaded from lib ommolketab ir downloaded from lib ommotketab ir Is there anything else can do If you d like to hack a solution you could build a system that can perform a regular timed reboot of your ATA Or you could cron a shell script that dials into aDataprobe AutoPALthis is a really cool device that lets you remotely reboot thingsor you could But he cut her off OK I think I ll just download the firmware patch Can I get it from your web site he asked Of course Is there anything else can do for you today don t think so he said Thanks for calling Ownage sir Have a good day she said and they both hung up Satisfied the customer picked up a pen and jotted down the entire conversation in the hopes of someday publishing it in a book about VoIP You can find out more about the Dataprobe AutoPAL at http www dataprobe com power auto_ pal html 1 9 1 The Hack Updating ATA firmware is a great way to stay on top of known performance issuesand it might allow you to take advantage of new telephony features introduced by your TSP Most telephony hardware vendors tend to make their systems more stable with each release
262. he Public Switched Telephone Network PSTN has been the dominant global voice network for a century Quality of Service VoIP is a wonderful technology It enables all kinds of features and portability options that are not available with traditional telephony technologies However unlike traditional telephony VoIP has some inherent quality issues By the time you finish reading this hack you ll Know how to tackle the quality problem with the VoIP engineer s best weapon the Linux kernel s built in router On the traditional telephone network every single call has adedicated time slot using a technology called time division multiplexing TDM With TDM a circuit is divided into several time slots each with its own dedicated slice of bandwidth This is what ensures that your call is the only call tn that time slot and that after all of the time slots are used the circuit is at top capacity and no further calls will be allowed With VoIP your call is converted into thousands of small datagrams or packets if you please Thes packets are then queued up on a device your computer analog telephone adapter ATA router etc and thrown out over the wire with no guarantee that they will even reach their ultimate destination wherever that might be You can see how this might cause problems with voice quality especially when other data traffic on that same link is vying for that link s limited capacity or bandwidth depending on your prefe
263. he arrows to find Phone Settings and press Menu Then it s the arrows again until you find Auto Config Press Menu and then press the up arrow to set Auto Config to Enabled Press Menu press Cancel and then press Menu twice to reboot the phone On the next boot the phone will look to your IP address to find its SIP and telephony configuration 5 11 3 Build a Uniden Configuration File The best way to learn Uniden SIP configuration is to step through the Uniden configuration files But before we do so let me go over the basic structure a Uniden UIP200 phone looks for when booting u and searching for its config on a TFIP server First the phone expects to find a file called unidencom txt which describes the operational characteristics that apply to all phonesthings like site wide audio settings addresses of SIP proxies softPBX servers that is DNS servers and the like The phone will also look for a file calleduniden lt mac gt txt where lt mac gt is the MAC hardware address of the phone So there is a single unidencom txt file and there are many uniden lt mac gt txt files l Il step you through the key settings found inunidencom txt Proxyserver gE cdl sal meal ProxyserverPort 0 OuLboundProxyl 10 11 10 OutboundProxylPort 0 The proxy settings tell the phone which SIP server to deal with when resolving dialed numbers and attempting to connect calls ProxyServerPort allows you to override the default SIP UDP port of 5060 if your
264. he audio packets for a short period of time to cushion against any delayed or out of order packets As the packets arrive the buffer grows and shrinks to accommodate variations in their latency thus smoothing out their perceived latency While jitter buffers are an excellent tool for improving audio quality on VolP traffic they still come with the cost of added latency in the call It is preferable to eliminate the jitter on the network altogether if possible One big source of jitter and latency is network congestion Let s suppose that you re on an important VoIP call when somebody on your network decides to download the latest movie trailer instantly using up all of your bandwidth Suddenly your VoIP packets have to wait in line behind the movie trailer packets causing a change in latencyi e jitter One solution to this problem is to use QoS policies on your router This is the practice of prioritizing some types of network traffic ahead of others QoS works well because some network traffic like downloading movie trailers or checking your email is not affected by small changes in latency or jitter and other services like VolP traffic are This uncovers a flaw in our use oftraceroute and ping to measure latency Because ping traffic is not the same as VolP traffic some routers QoS policies might treat them differently Hence the need for our Perl SIP ping utility Hack 72 As it uses SIP it provides a much better estimate for measuri
265. hecksum ensure the new checksum is Zzero Schecksum unpack 1l6n Sheader printf checksum after 04x n Schecksum die checksum failed n unless Schecksum 0 write the file open F gt Sfilename or die can t open output file filename n print F Sheader close F 5 4 2 Running the Code To use this program save it aS makering pl make it executable chmod 755 makering pl and pipe a uLaw sound file into it in a shell like so sox my_sound r 8000 c 1 t ul rate makering pl ringl bin In this example the file my sound will be resampled to 8000 Hz and will be piped in uLaw format to downloaded from lib ommolketab ir downloaded from lib ommoltketab ir the standard input of makering pl which is the Perl script shown earlier The enhanced output is then saved as ringl bin Upload this file to the tftpboot directory of your Grandstream s TFTP server and then reboot your Grandstream For some tips on setting up a TFTP server see Make IP Phone Configuration a Trivial Matter Hack 80 With a fun newringtone your IP phone is now as cool as your cell phone downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 62 Tweak Your Sipura ATA HACK 62 If you own a Sipura ATA you ve got a veritable softPBX hiding in that slick plastic enclosure If only you knew how to set it up Sipura Technology now a division of Cisco makes some very powerfu
266. hentic redbat phonelLl is the following HI I ve since learned that true bat phones not only are red but must also reside in a glass cake cover when not in use 1 Find a cheap old rotary dial phone at a garage sale or in the attic Remove the dial wheel and discard it Carefully remove the electromechanical guts of the phone and set them aside S Use some bright red plastic friendly Krylon Fusionspray paint to turn that vintage monster red like a tomato Allow it to dry of course 5 Put the phone guts back into the newly blushing enclosure reconnect to your Sipurabat phone ATAs and use that hotline to your heart s content Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 64 Brew Your Own Zaptel Interface Card HACK 64 With a little tweaking a very common fax modem card can become a clone of the single line X1LOOP interface card The Digium X1OOP foreign exchange office FXO card used to connect a single phone company line to an Asterisk server is actually an Intel V 92 Data Fax Voice modem card One visual comparison between an Intel V 92 Winmodem PCI card and an official X100P and it s obvious that the two cards are identical So using the less expensivemodem card in place of an X100P card is not only possible it s downright easy The Intel 537EP chipset is aV 92 PCI modem chip family Many modems are built on the Intel 537EP chipset but this h
267. hile every precaution has been taken in the preparation of this book the publisher and author assume no responsibility for errors or omissions or for damages resulting from the use of the information contained herein Small print The technologies discussed in this publication the limitations on these technologies that technology and content owners seek to impose and the laws actually limiting the use of these technologies are constantly changing Thus some of the hacks described in this publication may not work may cause unintended harm to systems on which they are used or may not be consistent with applicable user agreements Your use of these hacks is at your own risk and O Reilly Media Inc disclaims responsibility for any damage or expense resulting from their use In any event you shoulc take care that your use of these hacks does not violate any applicable laws including copyright laws ISBN 0 596 10133 3 M downloaded from lib ommolketab ir downloaded from lib ommotketab ir Credits About the Author Contributors Acknowledgments downloaded from lib ommolketab ir downloaded from lib ommotketab ir About the Author Ted Wallingford is a senior network engineer with LCG Technologies Corp in Elyria Ohio and the author of Switching to VoIP O Reilly Ted has led installations of VolP technology in the construction manufacturing and networking industries A periodic contributor toMacworld magazi
268. his being sure to include enabled yes and to add a section like the hansolo one to define a username and password with which to access the Asterisk Manager later on Asterisk Call Management support general enabled yes port 5038 bindaddr 0 0 0 0 hansolo secret falcon deny 0 0 0 0 0 0 0 0 permit 127 0 0 1 255 255 255 0 read system call log verbose command agent user write system call log verbose command agent user You ll need to restart Asterisk run asterisk rx as root at the shell prompt to commit these config changes Next you ll need to install the Asterisk Perl module Download compile and install it as follows wget http www netdomination org mirror asterisk gnuinter net files asterisk perl 0 08 tar gz downloaded from lib ommolketab ir downloaded from lib ommotketab ir tar xvzf asterisk perl 0 08 tar gz cd asterisk perl 0 08 perl Makefile PL make all make install 4H H Now pop into your asterisk perl source directory and check out the Asterisk Manager example in the examples directory It s called manager test pl and it demonstrates how to poke Asterisk with Perl For it to work though it will need to be authenticated as a legitimate Asterisk Manager API user anc that means adjusting the beginning of the script to match the username and password you put in etc asterisk manager conf Open manager test pl and make the username and password match Sast
269. his creates a context within Asterisk for incoming calls to arrive Now open up extensions conf so that you can create a dial plana set of instructions that tell Asterisk what to do in this incoming contextto correspond to peer 7711 Sincoming context setting incoming exten gt s 1 Dial SIP 7712 S CELL_ PHONE 30 exten gt s 2 Playback abandon all hope exten gt s 3 Hangup Since sip conf indicates that all incoming calls from SIP peer 7711 should enter the incoming context we ve created that context in the dial plan as shown in the previous code snippet Using the special s extension whose purpose is to incorporate into the context incoming calls that haven t been triggered by a user dialing an extension number calls like those incoming from the outside world we can specify three steps to deal with the call 1 Dial the cell phone number on the second SIP peer 7712 You can specify the numberfor instance 1 440 864 8604instead of using thes cELL_PHONE variable since we haven t really covered variables yet The 30 specified in the first command says to attempt to bridge conference the call on the two lines for up to 30 seconds before giving up 2 If the call isn t bridged because theDial command times out the Playback command will play a greeting In this case I ve specified a greeting calledabandon all hope 3 If either of the previous steps cannot be completed the caller will be disconnected So when an inc
270. how application LDAPget The show application command will confirm that the module is installed and loaded by showing you a brief description of the LDAPget dial plan command Now you can set up the LDAP inquiry your dial plan will use to get email addresses based on the DID provided bys EexTEN To set up this query open etc asterisk Idap conf It might not exist yet since you ve only just compiled the LDAP module Create an entry like this in ldap conf maLitlfromdid host ldap oreilly com user cn root ou People o oreilly com pass jarsflood base ou Addressbook o oreilly com filter amp objectClass person fax s attribute email Convert UTF 6 150 3659 1 downloaded from lib ommolketab ir downloaded from lib ommotketab ir This configuration will cause an LDAP inquiry to ldap oreilly com asking for an object of the person class with the attribute fax equal to the value of the 3s token which will be replaced with the DID at runtime The attribute setting tells the inquiry which attribute from the object to return as a value tc the dial plan s variable This might seem confusing right now but it should be clearer once you see howthe LDAPget Command is used in the dial plan In the context where your incoming PSTN calls begin specified inzapata conf you can capture the DID from the ExTEN variable and use it to supply an argument to an LDAP inquiry If the inquiry is successful Asterisk s LD
271. ial Service on Cisco IP Phones Hack 66 Power Cisco Phones with Standard Inline Power Hack 67 Customize Your Cisco IP Phone s Boot Logo Hack 68 Configure Multiple IP Phones at One Time Hack 69 Customize Uniden IP Phones from TFTP Hack 70 Control the Lights Using Your IP Phone Hack 71 Use a Rotary Dial Phone with VolP downloaded from lib eommolketab ir downloaded from lib ommotketab ir 5 1 Hacks 5971 Introduction One of the reasons VoIP is such a positive evolutionary step for telecommunications is that it employ a highly distributed software centric design philosophy It has the extensibility and programmability of the Internet putting telephony power back into the hands of the users not the phone companies It is programmabilitysoftwarethat makes IP telephony such a killer application Yet critical parts of voice telecommunications are entirely in the domain ofhardware This chapter focuses mainly on hardware hacks projects with a decidedly piquant earthy flavor projects that deal with analog telephone adapters ATAs phone line gateways and bat phones You won t need much in the way of I33t coding skills but you ll use some basic Perl Have some Ethernet patch cables and Velcro handy Oh and it might help to have a bottle of XML on ice if the mood is right In this chapter you re going to be looking at VoIP and legacytelephony hardwareeverything from State of the art Internet Protocol IP phones to vint
272. ials username and password to use for each phone line you re going to be passing through to the Asterisk server If you re justconnecting a single line you need only establish credentials for VolP1 VoIP2 VolP3 and VolP4 correspond to the second third and fourth phone lines you can connect The User ID tends to be a phone number When this User ID is dialed by a caller it signifies that this SIP endpoint should be called This differs from theAuthentication ID which is used to register the SIP endpoint with the Asterisk server Authentication ID and User ID needn t be the same but they often are In this case we ve chosen 7711 for both This is the number we ll use later with Asterisk ti handle calls to and from the phone line that s connected to this mediagateway Click the Save and Restart command button to reboot the media gateway Then click Supplementary Function This will pull up a page similar to the one in Figure 4 3 downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Figure 4 3 The Clipcomm s supplementary function configuration page VoIP Supplementary Function Cees J VolP FEO Interface EERE E NAT Traversal Stage FX0 Gateway e upport Use FXO PIN Code E FRO PIN Code 4 digits pood Call Forwarding to VoIP E Call Forwarding to PSTN te Call Forwarding Number Channel PSTNI j 2 Stage FXO Gateway E Use FXO PIN Code E FXO PIN Code 4 digits oota
273. icates that O is the reference number that RTCP will later use to refer to this G 711 uLaw at 8000Hz capability The other capabilities are advertised with other numbers These numbers are reserved like commonly used port numbers in TCP IP and they can be overridden In Figure 6 11 you can see that the 200 ox response sent by the receiver to the sender has an SDP payload that presents no audio codecs at all in its media attributes This is because they have been purposefully disabled of course Packet 10 is the customary SIP ack method acknowledging receipt of the 200 ok and giving the go ahead for RTP to begin But without any matching SDP media attributes to establish the RTP media channel the receiver selects attribute reference number 101 using SDP s m token 101 means that no valid capabilities match RTCP will report to the calling endpoint a few seconds later that no media channel exists and the receiver hangs up with a SIPByvE method in packet 12 Figure 6 11 The bottom pane of the capture window shows the media attribute list SDP s listing that advertises the capabilities of the receiving endpoint in this case Asterisk downloaded from lib ommolketab ir downloaded from lib ommoltketab ir amp Untitled Ethereal Sklic File Edt View Go Capture Analyse Statistics Help No Time Source Desbnation Protocol Info 7 4 249763 10 1 1 202 10 1 1 10 SIP SOP Request INVITE sip 20181
274. ick Advanced You should see a wealth of new options appear Move over to the PSTN Line tab Table 6 1 shows you the values to fill in for this page Table 6 1 Values to place in the PSTN Line configuration Field name Value Proxy IP address of Asterisk server Username spa3k Display Name spa3k Password spa3k Register Yes Make Call without Register io Ans Call without Register No Dial Plan 8 s0 lt 1000 PSTN Ringthrough iNo PSTN Default DP 8 PSTN Answer Delay 8 After you have entered these changes click Submit all changes The Sipura will reset and once it reboots you should have a fully functioning SIP PSTN gateway connecting calls between your Asterisk server and the PSTN Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir C PREY Hack 80 Make IP Phone Configuration a Trivial Matter fiso Trivial File Transfer Protocol TFTP servers are simple stripped down file storage servers that play a role in Vol P networks that s anything but trivial When an IP phone is first powered up it goes through a boot up sequence that s similar to that of a PC While most PCs boot up from a functional configuration that s stored locally an IP phone s configuration can be controlled remotely by an administrator who can store each configuration in a central repository IP phones use the TFTP protocol to retrieve updated configurations fro
275. id broadband Internet connection Skype for Windows runs on Windows 98 and up while Skype for Mac runs on Mac OS X 10 2 and up For more Skype details sift through the delicious goodies inChapter 3 2 14 4 The Skype Alternatives Gizmo Project which is a lot like Skype but uses industry standard SIP for call signaling is as of Version 1 0 very limited in terms of conferencing In fact it doesn t have any Vol P based conferencing built in at all Conference calling using SIP requires a centralized conference mixing server a complexity that makes Skype preferable to Gizmo for in game conferencing Google Talk http talk google com is another Skype alternative Like Gizmo Project Google Talk which has two party voice calling features supports a well known standard for call signaling called Jabber And like Gizmo Project Google Talk is free iChat can be configured for use with the Google Talk Jabber network as can various other IM clients such as Trillian and Adium So you aren t confined to using Google s Windows only client if you want into the network But that s where the pros end and the cons begin Like Gizmo there s no way to do conference calls And worse still voice chat between official Google Talk clients and non Google clients such as iChat doesn t work at all So Google Talk s usefulness as an in game voice conferencing tool is well nonexistent 2 14 5 The Hardware downloaded from lib ommolketab ir download
276. idencom txt covers the networking configuration of the phones specifically SIP proxies So if you wanted a particular config to affect all the phones unless overridden in the phone specific configs you would place that config in unidencom txt Cisco Grandstream and others use a very similar model to Uniden For some tips on Cisco and Uniden TFTP configuration check out the hacks inChapter 5 The unidenBase txt file is downloaded from lib ommolketab ir downloaded from lib ommoltketab ir there just as a template from which to generate future phone specific config files downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 81 Peek Inside of SIP Packets rst When the going gets tough the tough sniff packets Adding VoIP to any network can be a daunting challenge but accomplishing the task can seem particularly impossible when problems begin to arise To help troubleshoot any network problems being proficient with the use of network analysis tools can provide some restful nights Ethereal is a network analysis tool that allows for the sniffing or capturing of data packets on the network Ethereal has some VolP specific features too By digging deep into VoIP signaling conversations wit Ethereal and assessing SIP traffic problems using a conventional call flow graph you ll reveal the source of many problems you re likely to encounter Because of the dominance of SIP this exercise wi
277. ier to be Sure downloaded from lib ommolketab ir downloaded from lib ommotketab ir 1 8 2 4 Use a good old fashioned permanent marker If all else fails using a felt tip permanent marker write the full 10 digit phone number of thelocal public safety dispatcher on every phone in your house that uses your VoIP service Don t write it on tape or sticky labels adhered to the phone because they will eventually peel off and you never know when you ll need that important phone number downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 7 Update Your VoIP ATA Firmware WEG An ATA with up to date firmware will have fewer problems Yesterday made phone calls through my VoIP TSP all day long But today don t even hear a dial tone when pick up the phone grumbled the frustrated consumer regretting having replaced his local telephone service with a slickly advertised VolP service from a California company called Ownage This was the third or fourth time his VoIP service had quit working So he grabbed his cell phone and frantically called the Support Department at Ownage The tech who answered wasn t especially helpful She listened to the customer describe his recurring problem and then told him the same thing Support had been telling him ever since the first time the dial tone disappeared Sir can you reboot your analog telephony adapter by removing the power cord and then plugging the p
278. il gateway How about building a PBX server with no hard disks or bridging a SIP network with the Skype network I m willing to bet you ll rise to the challenge downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 88 Build a Killer Telephony Server HACK 88 Using any old PC with Linux is great way to experiment and learn about I nternet Protocol IP based telephony but to implement a production server you ll need some slightly bigger iron and you ll need it hardened In my travels as a networking consultant get to visit a lot of enterprise data centers These range from meager stuffy 100 degree closets crammed with desktop PCs that accidentally became servers Surrounded by a spaghetti pile of crummy cabling to the 2 000 square foot raised floor uncomfortably cool server rooms with halon fire prevention systems and row after row of racks filled with quad processor servers When have an opportunity to work in a modern decked out environment I m thankful that I m not crammed in an undersized overheated closet searching in vain for a free port on an incorrectly labeled Ethernet switch where can plug in my PowerBook It never ceases to amaze me just how little some folks seem to care about the environmental state of their critical data and communication equipment As long as things keep running Suppose they aren t likely to balk at the sorry state of their servers But with a critical
279. il is following the logfile call each number that causes distinctive rings on your POTS lines When the POTS interface senses the ring pattern a log entry will appear containing Asterisk s representation of it a string of digits made up of three values separated by commas Each value represents a duration of ringing such that each ring pattern could have up to three rings of varying length in a one or two second time span The pattern repeats at regular intervals until the call is answered This string supplies a value to thedring argument in zapata conf Repeat this process until you ve identified the strings needed for each phone number associated with your POTS line Here s a sample config in zapata conf that describes two distinctive ring signals and assigns them different contexts in the dial plan usedistinctiveringdetection yes Gray l 3259 95 0 dring2 95 0 0 dringcontexti TedsCalls dringcontext2 JakesCalls Channel gt 1 Distinctive ring features outside North America can use caller ID signaling instead of ring pattern signaling to indicate which phone number is being called Check with your telephone company to see how theysupport distinctive ring The Zaptel channel s configuration will tellAsterisk the context into which distinctively rung calls are sent In this example we ve used a POTS line with two ring signals and two corresponding contexts Now we ve got to create those contexts in the dial plan Here s a sample that acc
280. illy books does require permission Answering a question by citing this book and quoting example code does not require permission Incorporating a significant amount of example code from this book into your product s documentationdoes require permission We appreciate but do not require attribution An attribution usually includes the title author publisher and ISBN For example VolP Hacks by Ted Wallingford Copyright 2006 O Reilly Media Inc 0 596 10133 3 If you feel your use of code examples falls outside fair use or the permission given above feel free tc contact us at permissions oreilly com Safari Enabled When you see a Safari Enabled icon on the cover of your favorite technology book that means the book is available online through the O Reilly Network Safari Bookshelf Safari offers a solution that s better than e books It s a virtual library that lets you easily search thousands of top tech books cut and paste code samples download chapters and find quick answer when you need the most accurate current information Try it for free athttp safari oreilly com How to Contact Us We have tested and verified the information in this book to the best of our ability but you may find that features have changed or even that we have made mistakes As a reader of this book you can help us to improve future editions by sending us your feedback Please let us know about any errors inaccuracies bugs misleading or con
281. in binary and source code forms for both Linux and Windows though a crafty hacker should be able to get it running on a BSD ish OS too This project will allow a Microsoft NetMeeting H 323 softphone and an OpenH3230hPhone softphone to place calls through an H 323 gatekeeper running on a Linux computer In this example I ll use Microsoft NetMeeting on Windows XP and OhPhone on Mac OS X Although OpenH323 provides a framework of tools for developing H 323 servers and endpoints it also natively implements a complete H 323 gateway MCU and endpoint Here s a partial list of software packages that accompany OpenH323 OpenGK A simple H 323 gatekeeper server example OhPhone An H 323 softphone for Linux and Windows OhPhoneX is the Macintosh version OpenMCU An H 323 conference bridge server PSTNgw An H 323 gateway server Each requires the base distributions of OpenH323 and its prerequisite PWLib a project specific class library downloaded from lib ommolketab ir downloaded from lib ommotketab ir 7 3 1 Installing OpenH323 A Pentium III clocked at 600 MHz should be sufficient to handle the role of a small scaleH 323 gatekeeper The PC should be running Linux though H 323 is also Windows compatible and can be the same PC that runs Asterisk if you like The best place to get OpenH323 Is from its maintainer s web site http www openh323 org code html Compiling all of these elements is straightforward on Linux
282. in the dial plan configuration on the SIP server Once configured the SIP softphone will automatically register with the SIP registrar as soon as you close the Configuration menu If registration was successful you ll see Logged In in the UI display a Shown in Figure 6 2 If it wasn t make sure the SIP proxy profile called default is enabled and is configured to match a SIP account on the server 6 11 2 Configure Ethereal Once Ethereal is installed launch it Next begin a capture by selecting Capture Start This will show you the Capture Options dialog shown in Figure 6 4 To limit the kind of traffic that Ethereal will capture you ll need to use a filter string Ethereal has a rather sophisticated syntax for this string which instructs Ethereal what to capture and what to ignore This syntax is explored more in Managing Security with Snort and IDS Tools O Reilly Figure 6 2 The X Lite softphone s user interface downloaded from lib ommolketab ir downloaded from lib ommoltketab ir In this case our SIP server is 10 1 1 10 and the standard port for SIP traffic is UDP 5060 We want to capture traffic in both directionsthat is to the SIP server and to the softphone running on the Same host as Ethereal Here is the string that achieves this host 10 1 1 10 and udp port 5060 Check the Update packets in real time and Automatic scrolling in live capture options to see the packet capture log occur immediate
283. in the fields on this screen and press the Done P 7 14 5 Register for the Administration Guide To expand your sipX prowess beyond installation you should get your eyes on the sipX Administration Guide But to do this you ve got register with the chief commercial sponsor of the sipX project Pingtel The URL for registration is https Secure pingtel com registration registerUser jsp downloaded from lib eommolketab ir downloaded from lib ommotketab ir Colophon Our look is the result of reader comments our own experimentation and feedback from distribution channels Distinctive covers complement our distinctive approach to technical topics breathing personality and life into potentially dry subjects The tool on the cover of VoIP Hacks is a Morse code tapper Also Known as a telegraph key this electrical switching device is used to send Morse code over electrical wires The old school variety of telegraph key glamorized in many classic films was the straight key a simple contraption fashioned from a bar with a knob fastened atop one end When the knob was depressed the bar completed an electrical circuit and current flowed through the telegraph wires B rapidly forming and breaking this circuit telegraphers could transmit a series of signals conventionally known as dits and dahs or more colloquially dots and dashes which spurrec an electromagnet on the receiving end to produce clicking noises that co
284. indows this application creates a basic TFTP server with a simple graphical user interface GUI but it does so using Mac OS X s built in TFTP daemon meaning there are no limits to how much traffic it can handle You just tell TFTP Server what folder you want to share using TFTP and that s it Linux and other Unix users can probably find a fully functionalT FTP server already on their systems To be sure issue a tftpd amp at the root command line If you don t get a command not found response you ve got the TFTP daemon and you just have to figure out how to make it work Thankfully there s nothing to it Just create a directory to create your TFTP repository commonly downloaded from lib ommolketab ir downloaded from lib ommotketab ir tftpboot and launch the TFTP daemon like this mkdir tftpboot chown nobody tftpboot tftpd 1 s tftpboot amp This will launch the TFTP server which on many Linux systems runs as the phantom usemobody for security reasonshence the chown to give nobody permission to the TFTP directory Any files placed in tftproot will be accessible from a TFTP client such as the one built into IP phones 6 10 2 Understand IP Phone Configuration The syntax of the phone configuration files saved on the TFTP server varies from one make of IP phone to the next That is Cisco has a different configuration file structure than Uniden for example But they all bear a few things in common First
285. ing in your Linux box Most commercial distributions of Linux Red Hat Debian etc make sound card configuration a straightforward affair Next get your hands on the Festival source code at http www cstr ed ac uk projects festival download html Find a link to download it and save it from your browser or uSewget to grab it and its supporting Speech Tools libraries wget http www cstr ed ac uk downloads festival 1 95 festival 1 95 beta Larg wget http www cstr ed ac uk downloads festival 1 95 speech_tools 24 e a a Lal G2 wget http www cstr ed ac uk downloads festival 1 95 festlex_CMU tar gz wget http www cstr ed ac uk downloads festival 1 95 festlex_POSLEX tar gz wget http www cstr ed ac uk downloads festival 1 95 festvox_kallpc_16k bal OZ m downloaded from lib ommolketab ir downloaded from lib ommotketab ir Once you ve downloaded Festival you should unpack compile and install it with these commands tar xvzf speech_tools 1 2 95 beta tar gz tar xvzf festival 1 95 beta tar gz cd speech_tools configure make make install tar xvzf festlex_CMU tar gz tar xvzf festlex_POSLEX tar gz tar xvzf festvox_kallpcl6k tar gz cd festival patch p1 lt usr src asterisk contrib festival 1 4 3 diff cd festival configure make make install H HHH H HH H HS OH OH The make make install commands take the longestupward of five minutes each on my trusty old garage built Pentium III
286. ing just one class ofvoice traffic over another This is particularly useful when you need to segment two groups of users regardless of the applications they re running In the other hack showed you how to prioritize voice for everybody But suppose you want to prioritize it only under certain circumstances If you were going to launch a service like Skype where users can make free calls to other Skype users and pay for calls to non Skype users you would want to provide the highest possible quality for the paying users right Let s say that you have a VoIP service that allows callers to interconnect with the PSTN like Skype and with other VolP users on the Internet Let s also say that you have two pricing levels The economy pricing level does not guarantee quality and is less expensive or perhaps free but the premium pricing level does and costs more in return will show you how you can implement this using a slightly modified version of AstShape There are two ways Linux traffic control can build classes of service by port number and by ToS headers First am going to assume you ve got a PC with two Ethernet interfaces loaded up with Linux and the ability to control the IP ToS bitsor port numbers used by the VoIP devices you re going to Support be they IP phones or VoIP servers This is absolutely critical All we are going to use to separate the two levels of our traffic are the IP ToS bits or UDP TCPport numbers so withou
287. ing terms from 3 4 2 Running the Code In the AppleScript Editor save this code as an AppleScript in your user sLibrary Address Book Add ins folder Then fire up the OS X Address Book and find a contact with a phone number as shown in Figure 3 3 Figure 3 3 The Mac OS X Address Book Tristan Degenharat Aaron Bailey Company _ Tristan Degenhardt E Kristian Kielhofner E Company a Joel Sisko James Taylor Steve Totaro work tristan lbop4 com E Ted Wallingford friend N j assistant Names work 440 555 1212 Tra Now right click or Ctrl click the contact s phone number The contextual menu should contain an option labeled Call with Skype if you ve done everything correctly up until this point Clicking that option will cause Skype to attempt a call to the phone number in the contact record Note that you ll need to have SkypeOut credit to call a PSTN phone number using Skype Remember those Skype pay services mentioned at the beginning of this chapter SkypeOut is one of them It lets you call old fashioned phone numbers from your Skype client for just a few cents a minute You can buy minutes ahead of time from http www skype com downloaded from lib ommolketab ir downloaded from lib ommotketab ir Even though the Address Book s phone fields are referred to aS numeric in nature throughout the Address Book AppleScript documentation you can enter alphanumeric values
288. ink scripting visit the source ol several of these scripts http www gunsmoke com scot home_ automation phlink hAtml downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 20 VoIP While Fragging P20 This sure beats typing OWNED in an in game chat window If you re like most Ubergeeks and I say this as an admitted ubergeek there might be no pastime more satisfying to you than online gaming Indeed it s hard to beat the pure excitement of fragging your best friend with a rocket launcher in Quake or laying down the Horde smack onto a World of Warcraft nOOb Of course if you re a Ventrilo or Teamspeak user you can use Voice over IP to rub it in your opponent s face verbally when you crush him Ventrilo and Teamspeak provide hands free conference calling designed for online gaming This way teammates can coordinate their strategies verbally communicating by mouth without interrupting their in game action rather than by typed messages which can be a real distraction Nothing s a greater mood killer than having to stop to type a chat message to call for a rescue only to get hit from behind by a stray rocket while typing your plea One great feature of both Ventrilo and Teamspeak is their push to talk capability This allows you to treat them like a walkie talkiecutting out the background noise that would otherwise be transmitted if the chat were always live With this feature you can even fore
289. io 19644 RATP AVP 0 amp 101 Media Type audio Media Port 19644 Media Proto RTP AVP Media Format ITU T G 711 PEMU Media Format ITU T 711 PCMA Media Format 101 Media Attribute Ca rtomap 0 PomuU so000 Media Attribute a rtpmap amp poma sooo Fie Untied S107 bytes i P 11 D 11M 0 E E E Fm m hal Packet 5 is the authenticated INVITE method The user in this example ts calling SIP user 201 Included in packet 5 is an SDP payload Ethereal indicates this in its Protocol column in the top packets pane of the main capture window shown in Figure 6 9 Packet 6 is the 100 TRying response Packet 7 is the ox response which also includes an SDP payload If there s a codec match in the media attributes list of the SIP INVITE and the 200 ok response shown in the bottom pane all that s needed is a sip acK method sent by the caller to confirm agreement on the first matching attribute That s what packet 8 is 6 12 2 Inspect Failed Capabilities Negotiation If there was no capabilities match call setup would fail This scenario can be produced easily by temporarily crippling the capabilities of the Asterisk server To make it impossiblefor the X Lite softphone to negotiate an audio stream with the Asterisk server you can disallow all codecs Supported by X Lite and permit only Global System for Mobile GSM codecs which X Lite doesn t Support Take a look at this snippet of etc asterisk sip conf which does just that
290. ion as shown inFigure 4 12 Phones at the East office will be 30003099 phones at the West office will be 34003499 Have one SIP phone register with the East server and the other SIP phone with the West server our West user will be using the phone that registers with the West server and the East user will be using the East registered phone The following code shows the SIP peer config for 3001 at the East office East office sip conf 3001 callerid East User lt 3001 gt Canreinvite no context default host dynamic mailbox 3001 secret 3001 type friend username 3001 The following code represents the SIP peer config for 3401 at the West office West office sip conf 3401 callerid West User lt 3401 gt cCAanreilnvyice no downloaded from lib ommolketab ir downloaded from lib ommotketab ir context default host dynamic mallbox 3401 secret 3401 type friend username 3401 With these first two configs committed the SIP phones can now register with their respective Asterisk servers and place calls in their own default contexts But they still can t call each other without dialing a lengthy PSTN phone number waiting for the auto attendant and dialing the extension on the answering Asterisk system To get around that we can tell both Asterisk servers to route calls bound for the extension number range of the other office out through the PSTN and automatically dial the extension on the answering system as foll
291. ional phones and IP phones is a simple matter of analog electronics seeChapter 5 but recording softphone and instant messenger voice calls is another matter entirely Of course you can set an old fashioned tape recorder on your desk and press the Record button but come on In our digital world there s got to be a better way right Of course there is You can find a handful of useful recorder apps athttp www download com and http www downloadsquad com that can record WAV files and MP3s from any sound input or output on your Windows PC One such application is Total Recorder developed by High Criteria Figure 2 6 In its default configuration Total Recorder will record only the output the person on the other end o the call but not your voice To alleviate this click Total Recorder s Recording Source and Parameters button and then check the Record also input stream checkbox This way your recording will be sure to contain both sides of the call The Remove silence checkbox will enable a feature that doesn t save moments of silence into the recording This might be useful if you record a ton of calls and review them regularly as waiting through unneeded silence would certainly slow this process and use up more hard disk space A real time saver is found by checking Convert using Recording Parameters specified below and then clicking the Change button In the dialog window that appears you can adjust thesound resolution and the
292. ioritize Packets to Improve Quality Hack 6 Got 911 Hack 7 Update Your VoIP ATA Firmware downloaded from lib ommolketab ir downloaded from lib ommotketab ir 1 1 Hacks 17 Introduction Voice over IP or VoIP for short is a technology that allows Internet Protocol IP networks like the Internet to be used to enable voice communication similar in some ways to a telephone Some folks call VoIP IP telephonyand the technology comes in many forms from desktop communication software to automated message recording and fax integration tools But in its simplest form IP telephony enables you to place phone calls over the Internet rather than over a traditional phone line This is a pretty big deal since no long distance charges or hefty federal access taxes are levied on Internet based phone calls Plus IP telephony lets you integrate your desktop PC your desk phone and your cell phone in ways never before imagined I m anxious to Share the details with you in this book In the tradition of O Reilly s Hacks book series you ll be using short hacks like the basic ones in this chapter to learn about Voice over IP and computer based telephony and a number of my peers in the telecommunications industry have contributed some of the most useful most educational and coolest projects to VolP Hacks Hopefully beginning right here in this chapter you ll be saying I didn t know you could do that with VolP downloaded from lib
293. is in progress the Connection Stats window shows the status of the call in excellent detail as in Figure 7 3 7 3 4 Register an H 323 Endpoint Using NetMeeting Microsoft NetMeeting is an H 323 softphone application that comes packaged with Windows Me 2000 and XP To run it on XP however you ll have to perform a slight hack to activate it Select Start Menu Run type conf and click OK Then select Put a shortcut to NetMeeting on my desktop in the wizard that follows Once this is done NetMeeting is activated on Windows XP just as it would normally be on Windows 2000 To configure NetMeeting to register with the gatekeeper inside NetMeeting click Tools Options This will display the Options dialog where you can click the Advanced Calling button The Advanced Calling Options dialog box will appear as inFigure 7 4 Check the boxes next to Use a gatekeeper to place calls and Log on using my phone number In the Phone number field enter the address of the gatekeeper as well as the E 164 address you d like to use downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Figure 7 2 OhPhonexX s console log can help you troubleshoot the registration process oks ohphone Console Log 16 gt H 261 ClF lt A Userlnputhookflash lt 17 gt 3 UserinputbasicSiring lt 18 gt Userinput dtmf lt 19 gt UserinputRFC2833 lt 20 gt Listening interfaces ALL 1720 Gatekeeper set OpenH323GK 10 1 1 10 Wa
294. is likely your email program of choice And if you re an Outlook user you can leverage Skype directly within Outlook allowing you to use your Outlook address book to contact Skype buddies All of this Office integrated goodness comes by way of the Skype Toolbar for Outlook add on a program that combines Office and Skype APIs to turn your emailer into the world s coolest softphone The hardest part of the setup process is the download which you can grab from http share skype com directory skype toolbar for outlook beta think it s safe to assume that once this software is no longer in beta you can remove the beta from that URL Once you ve downloaded the software and stepped through the installer you ll be eager to press buttons and turn knobs 3 3 1 Your New Outlook Toolbar The first thing you ll probably notice when you launch Outlook is that it has a new toolbar that looks like the one shown in Figure 3 1 The first button on the bar launches Skype and the second button provides a drop down menu that lets you change your Skype status very handy and export your Outlook contacts to Skype s buddy list Figure 3 1 The Skype Toolbar for Outlook downloaded from lib ommolketab ir downloaded from lib ommotketab ir This export process pictured in Figure 3 2 is a little shaky but it does its job of matching users from the global Skype directory to your existing Outlook contacts with passable accuracy though at a s
295. ise when you subscribe to traditionalphone service you re really just leasing a telephone line With that line you can use cordless phones corded analog phones answering machines fax machines modems and all kinds of other access devices These different analog devices all use the same electricalaccess signaling to communicate with the phone company You could think of this analog protocol as even more primitive than the Morse code It s simple but it s what allows analog phone devices to place and receive calls If legacy telephony devices are more primitive than the Morse code Session Initiation Protocol SIP the predominant VolP access signaling protocol is light years ahead of both SIP is a suite of media Signaling software specs that define how streaming media devices and applications should interact The most significant of modern streaming media apps is IP telephony of course which brings me to my point Unlike old fashioned telephone signaling which isPlug and Play PnP using a softphone is a bit more involved To understand how a softphone works or an ATA or IP phone for that matter you must have a simple grasp of SIP Although SIP is a sprawling specification with dozens of proposed spinoff and major revisions you need to Know only a few things to get by with a SIP softphone SIP is a lot like Simple Mail Transfer Protocol SMTP If you re comfortable with that SIP will make z lot of sense to you Like SMTP SI
296. it the phone directory off as its own service available through the Directory button rather than the Services button but that shouldn t stop you from making your own directory with this hack The menus are simple to create but you ll need to become familiar with the strange tags If you re not familiar with XML the lt ciscoIPPhoneMenu gt tag which is used to enclose Cisco phone menu structures might seem a little strange to work with would have just used lt Menu gt but that might not work in newer firmware Menus contain one or more menu items denoted with the tag lt MenuItem gt A softkey type of menu that uses the lt SoftKeyItem gt tag also is available Softkey menus use the buttons on the side of some of the phones the 7960 for example has six on the right and four at the bottom of the display Some of the older Cisco phones do not have the extra buttons so for compatibility with all of Cisco s IP phones stickto simple menus Inside lt MenuItem gt are easy tags such aS lt Name gt and lt URL gt These are casesensitive on some current firmware versions lt Name gt is what shows up as the item s name and lt uRL gt is what it will request if the user selects the menu option Notice that thehttp part is in the URL but FTP URLs will work too Now on to your first menu Start the following example and adjust it to suit Let s say you use the extension of the phone for the title to allow you and your friends or use
297. ith these guysthey are established knowledge leaders in this new industry I d also like to thank Mike Loukides who recommended that writeVolP Hacks or perhaps he merely succumbed to my nagging He edited Switching to VoIP and he is the author or co author of several excellent O Reilly volumes including the highly usefulUnix Power Tools and one of O Reilly s earliest technical books System Performance Tuning He s a pretty amazing pianist too VolP Hacks was edited by David Brickner David s editing is pragmatic politically incorrect and to the point love that I ll give David all the credit for anything good aboutVolP Hacks This book survived the criticisms of several tech reviewers including Kristian Kielhofner Leif Madsen and Jim Van Meggelen Thank you for looking over my work your expertise added much to the book s accuracy must heartily acknowledge my hometown crowd too My wife Kelly and my friends at Pathway have given me plenty of much needed encouragement The crew at LCG Technologies is a great bunch too just barely squeaked this book out thanks to my new workloadkeep up the great work LCG Thanks to Brian Downey of The Linux Fix for his expert Linux Support as well downloaded from lib ommolketab ir downloaded from lib ommotketab ir Preface Voice over IP or VoIP is a family of technologies that enables voice applications and telephony to be carried over an Internet Protocol IP network such
298. iting for incoming calls for 2112 Command ClearAll 3 fe Figure 7 3 OhPhoneX s Connection Statistics window tells you which codec your call has selected and how much bandwidth it s using downloaded from lib ommolketab ir downloaded from lib ommotketab ir F e x OOO Connection Statistics in Call With Evil 10 1 1 109 Call Duration 0 25 Type Cadec_ Bitrate Toral Packets Total Bytes Audio In G 71l uLaw 64k isw 63 3 kbit s 825 193 4 KB Audio Out G 71l uLaw 64kisw 47 9 kbit s 552 1295 4 KE Video In 0 0 kbit s 0 O Bytes Video Out 0 0 kbit s 0 Bytes Connection Quality Packets Lost 0 0 0554 Round Trip Delay Oms Packets Late 0 Packets Out of Order 0 Figure 7 4 The NetMeeting Advanced Calling Options dialog box allows you to configure gatekeeper registration downloaded from lib ommolketab ir downloaded from lib ommotketab ir Microsoft NetMeeting is a very worthwhileH 323 softphone and it s quite customizable It allows video calling as well as audio calling and it has a built in T 120 whiteboard and instant messaging text chat applications You can tweak the codec selection preferences by choosing Audio from the Options dialog and then clicking Advanced The codec selection dialog is shown irFigure 7 5 If you re really looking to restrict codec selection most compliant gatekeepers allow you to do it centrally 7 3 5 Make the Call Once both phones are regist
299. k greeting 30 6 x tri phlink cronjobs Friday sh 30 6 2 K mon phlink cronjobs Monday sh 30 6 tue phlink cronjobs Tuesday sh 30 6 i wed phlink cronjobs Wednesday sh 30 6 i tnu phlink cronjobs Thursday sh Now your Phlink setup will have a different greeting depending upon the day of the week downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 18 Use Caller IDs in AppleScripts HACK 18 One of Phlink s AppleScript hooks occurs when incoming calls arrive which means you can create actions to handle how those calls are handled If Phlink didn t have AppleScript support it wouldn t be nearly as cool as it is In fact when I first fired up the Phlink application looked at the minimal interface and thought to myself Is thatt The fact is that Phlink s most awesome functionality is in its AppleScript object model By using Phlink s functions in tandem with other AppleScript aware applications you can do some very cool telephony automation from music on hold to greetingcallers with the Mac s speech synthesis in interactive stages Anything you can retrieve into an AppleScript variable from other Mac apps you can pass into Phlink functions for interaction with callers The only limit then is your imagination1 H For an unrelated affirmation of the limitlessness of the human spirit visit http Awww zombo com As got into setting up these voice AppleScripts was reminded of
300. kype and Skyping Skype the ubiquitous desktop voice calling application is one of the most hackable desktop telephony tools and therefore is worthy of an entire chapter of hacks Chapter 4 Asterisk Just as VoIP enables desktop telephony it also enables enterprise telephony In this chapter you ll learn how to install configure and hack Asterisk an open source PBX Chapter 5 Telephony Hardware Hacks VolP is rooted in software but it is used with lots of different kinds of hardwareeverything fron next generation IP phones to old school rotary phones This chapter shows you how to add these devices to your VoIP setupand how to customize them Chapter 6 Navigate the VolP Network Voice over IP is carried over the network using packets just like traditional data With the advice in this chapter you can monitor VoIP and troubleshoot it using traditional admin tools downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Chapter 7 Hard Core Voice By the time you reach this chapter you will have advanced to the hallowed ground that s held by a very exclusive crowd the community of hard core voice hackers Conventions Used in This Book The following is a list of the typographical conventions used in this book Italics Used to indicate URLs filenames filename extensions and directory folder names For example a path in the filesystem will appear as Developer Applications Constant width
301. l Select the host you want to snoop on the left and then select one or more of the destinations relatec to that host on the right This way you ll be poisoning ARP requests for the snooped hosts only if they re sent from certain IP addresses don t recommend that you ARP poison a large groupthis could cause difficulties on the network including a rather comprehensive denial of serviceso stick to one IP address on the left and one on the right until you ve got the hang of It Ideally the address on the right will be the host on the other end of the call but it doesn t have to be If the VolP call is to a user on the Internet somewhere via the default gateway you would choose the IP address of your Internet router on the right side In fact if you wanted to sniff your Vonage calls you would pick the address of your Vonage ATA on the left and the address of your Internet router on the right Once you ve got a pair of addresses selected for monitoring click OK Now start the sniffer and the ARP poison router by clicking the Start Stop Sniffer icon and the Start Stop APR icon which looks like a radiation symbol Wait for a VoIP call to be placed on the targeted hostor place one yourself on that hostand watch the call list in the VoIP tab In a moment an entry will pop up indicating that the call is in progress and is being recorded into a WAV file by Cain amp Abel as shown in Figure 6 18 Figure 6 18 A call has been logged and reco
302. l It will probably take two rings before caller ID information is transmitted from the phone company it tends to come between rings but Phlink calls thering script until it returns false as in the previous example An even cooler use of thering script is to retrieve the caller s Address Book entry based on the caller ID signals received on incoming_call given callername theName if the Name is return true else tell application Address Book set selectedPerson to the name of the person whose name contains theName end tell end if return callAgain end incoming_call downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 19 Control iTunes from Phlink HACK 19 If you have a ton of iTunes tracks just sitting there on your hard drive why not put them to work in Phlink One of the coolest things about Phlink is its AppleScript abilities Much like PhoneValetand other desktop telephony packages custom scripting is where all the fun lies Sure letting folks record their voicemail onto your computer is fun but integrating the other stuff on your computer with the phonethat s even better You ve seen how to do some basic database interaction between Phlink and the Address Book Hack 18 That s a great starting point for this hack because it introduces the events that can trigger scripts within Phlink If you haven t been there already check it out come back and l Il be waiting here with this iT
303. l libdnet and libevent downloaded from lib ommolketab ir downloaded from lib ommotketab ir To download compile and install thelibdnet and libevent libraries required by vomit log in as root and use these commands H HHH HEH HE HH HH HCH cd usr sre wget http ufpr dl sourceforge net sourceforge libdnet libdnet 1 10 tar gz tar xvfz libdnet 1 10 tar gz cd libdnet 1 10 configure make make install cd wget http www monkey org provos libevent 1 la tar gz tar vzxf libevent 1 la tar gz cd libevent 1 la configure make make install Obviously this is just a sequence of commands to fetch the libraries open the archives and compile the source code within 6 15 1 2 Compile and install vomit Next grab the vomit tarball and compile it on the same machine again as root HH SH HS HOH cd usr sre wget http vomit xtdnet nl vomit 0 2c tar gz tar zvfx vomit 0 2c tar gz cd vomit 0 2c configure make make install 6 15 1 3 tcodump some packets When I did this hack did it on my Asterisk server running on Linux This simplified the capture process since all had to do was set up an extension on the Asterisk server that answered the call immediately and produced some audio For a refresher on doing this flip back to Attach a SIP Phone to Asterisk Hack 42 Once the extension was in place startedtcodump like this tcpdump w test file downloaded from lib ommolke
304. lP knowledge enough to serve as an introduction to the world of VolP and teach you how to use it to save money be more productive or just impress your friends If you love to tinker and optimize this is the one technology and the one book you must investigate downloaded from lib ommolketab ir downloaded from lib eommolketab ir VoIP Hacks By Theodore Wallingford Publisher O Reilly Pub Date December 2005 ISBN 0 596 10133 3 Pages 326 Table of Contents Index Copyright Credits About the Author Contributors Acknowledgments Preface Why VoIP Hacks How This Book Is Organized Using Code Examples Chapter 1 Broadband VoIP Services Section 1 1 Hacks 17 Introduction Section 1 2 VolP Based Phone Service Providers Hack 1 Get Connected Hack 2 Use Pure VoIP Dialing with Your TSP Hack 3 Wire Your House Phones for VoIP Hack 4 Use a Softphone with a VoIP TSP Hack 5 Prioritize Packets to Improve Quality Hack 6 Got 911 Hack 7 Update Your VoIP ATA Firmware Chapter 2 Desktop Telephony Section 2 1 Hacks 827 Introduction Hack 9 Track Vonage Account Info on Your Deskto Hack 10 Pick a Desktop VoIP Client Hack 11 Sound Like Darth Vader While You VoIP _ Hack 12 Grow Your Social Network with Gizmo Hack 13 Record VolP Calls on Your Windows PC Hack 14 Handle Calls with Windows Software _ Hack 15 Let Your Mac Answer and Log Your Calls Hack 16
305. lib mv usr local lib libid3tag so usr 1lib Once madplay executes without any error notices you re ready to go on to the next step You ve got to tell Asterisk s voicemail module that you want it to usemadplay as its preferred player Comment out the default line in the etc asterisk musiconhold conf file and add an entry like this in its place default gt custom var lib asterisk mohmp3 usr local bin madplay mono R 8000 output raw This tells Asterisk to use the madplay application to stream random MP3 files from Var lib asterisk mohmp3 in mono at a forced sample playback rate of 8 MHz perfect for telephony Though you don t need the Zaptel driver or card for a SIP only setup music on hold bridging is dependent on the Zaptel driver framework s built in timing code and you won t hear much music on hold unless you load either a real Zaptel driver for a real Zaptel card or the Zaptelztdummy driver which is meant to fill in on machines that don t have an actual Zaptel board installed Lucky for you when you compiled the Zaptel drivers Hack 41 you also unwittingly compiled ztdummy How convenient Put these commands in etc rc d rc local before Asterisk loads if you have no Zaptel card installed modprobe zaptel modprobe ztdummy Next make a test extension that lets you listen to some on hold music Place an entry like this in etc asterisk extensions conf in the most appropriate context downloaded from lib ommolk
306. lication that implements its own network and signaling protocol There s a reason for this proprietary characteristic of Skype s design despite all the great open VolP Standards such as Session Initiation Protocol SIP Since Skype uses its own P2P network and proprietary signaling protocol it can get around the biggest problems facing the open standard SIP protocol which often breaks when used on phones that have to connect to the Internet through a broadband firewall router SIP was designed withoutfirewalls in mind so SIP based VoIP can prove frustrating for home users who just want to call a friend without a lot of technical hassle By solving this problem Skype has become the most widely used desktop VoIP application in the world This might be why Skype s official slogan is it just works Yet this is also why Skype inspires some controversy VolP advocates want to leverage Skype s ubiquity to advance VoIP s popularity but to do this it needs to support the openstandard for VolP Signaling SIP Perhaps at some point Skype will provide for SIP compatibility or open the proprietar Skype signaling standard so that VolP hackers can bridge the gaps betweenSIP based apps and the Skype network Perhaps Skype s coolest feature isn t a feature at all The Skype application programming interface API is a development framework that allows programmers to buildubercool add ons for the Skype network I ve pointed out a few of the coolest ones in this
307. likelihood of jitter If it bounces all over the place it s more likely that you ll experience jitter on a call Because this script takes a sample only every five minutes it doesn t provide the truest possible measurement of the type of jitter that will affect a VoIP call But since it s very difficult to trigger UDI datagrams at a real world rate of 20 to 50 per second using thesip_ ping pl and RRDtool tools measuring latency over time will have to suffice Plus with that many samples you d have a ton of numbers to crunch to figure out the utilization trends over a long span of time like days weeks or years By compromising short term accuracy and taking samples only every five minutes it s easy tc look at latency over long spans of time Brian Degenharat downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 76 Explore NAT Traversal 76 Network Address Translation NAT poses a fundamental problem for VolP But like so many problems the first step toward a solution is realizing you ve got a problem Have you ever tried a Voice over IP application that purports to let you talk to your buddies over the Net only to fire it up and not be able to call them or worse yet to hear only dead silence on the other end Chances are the VoIP application is being broken by your firewall router NAT is most often used to permit computers that are on private networks behind firewall routers to access services
308. lketab ir downloaded from lib ommotketab ir Asterisk after the Festival patch has been applied The Festival installation instructions provided earlier show how to apply this patch In the Asterisk extensions conf file the Festival command simplifies the playing of synthetic speech for callers exten gt s l Answer exten gt s 2 Festival Hello caller My name is Mr Synthetic To greet a caller by name use Asterisk s built in caller ID variables exten gt s l Answer exten gt s 2 Festival Hello S CALLERIDNAME My name is Mr Synthetic The Festival Command allows you to send speech to callers but not to the sound card To do that you ll need to pump some output from Asterisk to the Festival application at the appropriate time in the dial plan Using Asterisk s System Command you can trigger all kinds of activity in the Unix Shell from within Asteriskincluding of course Festival activity Take a look at this Festival shell command which simply speaks the quoted text through the PC s sound card echo Hello world festival tts This causes the text output of the echo command to flow to Festival s standard input It is then spoken using the audio output of the sound card Commands like this really giveAsterisk some cool speech abilities Say you wanted to have your Asterisk server announce the caller ID of each incoming call exten gt s l System echo You are receiving a call from SCALLERIDNAME
309. ll you ll literally hear crickets chirping how appropriate 2 6 1 Extra Gizmo Features Gizmo comes with a few extra features not available with a default install of Skype 2 6 1 1 Map It downloaded from lib ommolketab ir downloaded from lib ommotketab ir Have you ever wanted to know where the person you re talking to is located When you re in a call click the Map It icon and you ll see a very nice satellite photo with lines drawn between the estimated locations of the call s participants as inFigure 2 4 Figure 2 4 Gizmo s Map It function 2 6 1 2 Record It Podcasters rejoice Folks who ve been looking for an easy way to record theirVolP calls from Skype and other softphone apps need look no further than Gizmo Project which has the built in ability to record all calls without the need for any other software As a call is in progress if you want torecord it just click the circular record button at the top of the conversation pop out window that appears at the outset of each call Recorded calls are saved in a WAV audio file on thedesktop by default you can change this location in Gizmo s preferences 2 6 1 3 Gauge It At the bottom of Gizmo s main window is an icon that looks like the signal strength icon you might bi familiar with from your cell phonea row of vertical bars that indicate thequality of the connection to the phone network In Gizmo s case the bars represent the quality of the voice pathway
310. ll concentrate on using only the features of the Ethereal sniffer that support analyzing this protocol rather than any of the protocols that came before it MGCP H 323 etc Feel free to learn those other protocols if you like but know that almost every commercial VolP provider uses SIP when Say almost mean 95 or more You can use Ethereal to inspect network traffic from the Ethernet layer all the way up to the application layer Packets are captured in a buffer and are displayed on the screen Filters can be applied to restrict the capture to packets matching a certain source destination size protocol or service To obtain Ethereal download it from http ethereal com and install it on your Windows Mac or Linux box The screenshots and examples here assume the Windows version On a switched network a nonadministrative user can only capture packets being sent to or from his own machine So to keep things simple this hack monitors traffic for the SIP softohone known as X Lite Hack 4 Both X Lite and Ethereal will need to be running on the same machine If you re using a nonswitched network like a hub Ethereal can observe packets not bound for or originated from your PC which means you ll be able to monitor all VoIP traffic without restrictions Butfor now l Il assume you can monitor traffic only on your own PC Another way to view packets bound for other PCs on a switched Ethernet network is to use a technique known as
311. llers can be prompted to select which voicemail box they d like to record their message in If you ve got a large household this can be a godsend Mom and Dad can have the generic family voicemail greeting and teenagers Todd and Susie can feel cool because they have their own individual voicemail greetings With distinctive ring enabled on your phone line the kids can even have their own separate phone number and Call Soft Pro will recognize their ring pattern and play their special greeting downloaded from lib ommolketab ir downloaded from lib ommotketab ir TOSC makes a scaled down version of Call Soft Pro called Call Soft which is geared more toward home users and lacks many of the automation features of the Pro version Figure 2 9 Call Soft Pro is a comprehensive message recorder and telephony package for Windows desktops ey Call Soft Pro V3 1 File View Modem Call Options Help Ga Connected F Ha AIDE brite a Answer D ahetiessane x flac 0 New Messages Listening for calls on Softk56 Data Fax Voice Speakerphone CARP Incoming Call History Phone Book Outgoing Call History Dialer Faxes To Send Mailboxes Caller Humber when Length dim amp Wicki Home phone 830 405 4 45 August 0r 10 56 Bi Gade 0 Main Mailbox 0 Dutofaea 0000000000 August 02 2 06 PM E Chris amp Aeda 936 471 3809 August 01 6 29 PM foe 1 Lisa Mom 632 239 9594 July 23 3 02 PM t MEARS L 936 448 5
312. lling other realms using SIP and generally only support calls to and from the public telephone network an outbound proxy isn t necessary But BroadVoice for one requires that you configure an outbound proxy addressand in its case it s the same address as that used for the SIP proxy setting Use Outbound Proxy This setting tells X Lite whether you want to treat all SIP requests as though they are destined for another realm This effectively circumvents the SIP proxy for any activity other than registration though if the two proxies have the same address as in BroadVoice s configuration it doesn t matter what this setting is set to The choices are Always and Never in case you were wondering Some TSPs have more than one SIP proxy and they might allow you to choose among them To determine which one to use ping them all The one with the least amount of variance from one ping packet to the next is the one you want Register This setting tells X Lite whether you want the SIP client to authenticate and register with the SIP proxy server It s very Uncommon not to register and you won t get very far with your VolP service if you don t So definitely set this one to Always For most TSPs you can leave the rest of the settings unchanged For a more detailed description of X Lite s settings you can download a PDFuser manual from CounterPath s web site http www counterpath com downloaded from lib ommolketab ir downloa
313. ltelephony devices With hundreds of options and many potential combinations literally thousands of possible configurations are available While can t cover them all here for obvious reasons can give you a few examples to get your mind working 5 5 1 Configure the Sipura by Dialing Sipura s line of products has a powerful interactive voice response IVR system built in that gives you administrative access to many of the ATA s features In fact the IVR will probably be one of your first experiences with Sipura s ATAs The IVR like the web interface has quite a few options Thankfully Sipura publishes a user guide that details all of the available options in the IVR menu as well as in the web configuration screens In fact so many options are available that the user guide was 87 pages long at the time of this writing You can find this document in the Support section of http www sipura com After unpacking the Sipura and connecting the cables you should pick up your phone and dial This will connect you to the Sipura Configuration Menu You will be asked to enter an option But what option to enter Table 5 1 will get you started Table 5 1 Configuration options for the Sipura via IVR Option name Spron Valid options Notes name DHCP status 100 None Check IP address 110 None Reads current IP Set static IP 111 IP address enter IP address using to input periods Check network a Nore mask Set network Same as
314. ly click the link to launch the conference call Try a link like the following lt script Sro http pl ugin y vya 15 7 plugin j6 gt lt ecr ip lt a href CREATECONFCALL jyvetestl garfield odie nermil gt lt img src http jyvetools jyve com sendvoicemail gif border 0 onclick setENDown gt lt a gt Just put a tilde between each contact in the link When clicked the J yve plug in will use the Skype client of the clicker to host the conference call downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 34 Carry Skype in Your Pocket P34 Turn your Pocket PC PDA into a wireless VoIP phone You can use your WiFi enabled Pocket PC as your personal Skype portable phone Well not any Pocket PC only a Pocket PC running Windows Mobile 2003 will work And it ll need a 400MHz processor too Pretty hefty specs know But to do cool things you need cool hardware right The sizzling HP iPAQ hx4700 Pocket PC has an XScale processor that cruises along at a cool 624 MHz incidentally Grab PocketSkype from http www skype com products skype pocketpc and install it on your Pocket PC device using ActiveSync Once you ve installed it you should be able to Skype people usin the Pocket PC in the cradle sharing your PC s network connection The PocketSkype user interface shown in Figure 3 7 is similar to the desktop versions Figure 3 7 Skype for Windows Mobile 2003 PocketSk
315. ly instead of waiting until the capture session is complete Then click OK and the main capture window will appear You re now ready to observe your SI Pregistration attempts 6 11 3 Observe SIP Registration Now restart X Lite It will attempt to register automatically with the SIP server upon startup By the time X Lite says you re Logged In you can stop the packet capture by clicking Ethereal s Capture Stop menu item Figure 6 3 X Lite s SIP client configuration downloaded from lib ommolketab ir downloaded from lib eommolketab ir The main capture window should be filled up with a number of packets as shown in Figure 6 5 In this instance Ethereal shows the first packet packet number 1 aS aSIP REGISTER method Newer versions of Ethereal such as the 0 10 7 version used here can parseSIP packets and tell you which methods and responses they contain Packet 4 is a second registration request the first one failed because X Lite tries anonymous registration first Packet 5 is the 100 TRying SIP response sent from the SIP server back to the softphone Packet 6 is the 200 OK SIP response sent from the SIP server back to the softphone indicating that the registration was successful Packet 7 is asIp NoTIFry method asking for username 204 at the SIP registrar Packet 8 is the 200 ox response At this point registration is complete The additional packets 9 and
316. m FXO devices and vice versa deprecated in favor of wctdm in releases of Asterisk later than 1 0 5 f R To alleviate confusion over FXS FXO kernel module naming wcfxs has been The next change you need to make is in the etc asterisk zapata conf file The sample configuration should be completely commented out comments are denoted by semicolons at the beginning of the line If it s not commented out place a semicolon at the front of each line Then add the following lines to the end of the file context default signalling fxs_kKs usecallerid yes echocancel yes callgroup 1 pickupgroup 1 immediate no channel gt 1 The first line tells Asterisk what set of assumptions to make i e what context to choose when handling calls coming in on the POTS line The second line tells Asterisk notztcfg what type of Signaling the XLOOP has been set to use The following lines turn on a few traditional telephony featurescaller ID echo cancellation and other stuff that s covered in more detail later The last line assigns all the previous settings to channel 1 The assignment of these inherited settings uses the gt assignment operator rather than just an equals sign The Asterisk configuration parser doesn t distinguish between them the convention is merely for ease of human readability Contexts are Asterisk s way of meaningfully grouping call flow scenarios A context describes what behavior is caused by dialing 1 at the outset of a
317. m just passing on some knowledge picked up while working as a networking consultant 6 17 1 Get to Know Cain amp Abel While you re here make sure you ve read Log and Record VoIP Streams Hack 86 which introduces the software tool I ll be using to make all this happen Cain amp Abel If you don t know Cair amp Abel go back a hack read it through and you ll be able to proceed with confidence Now on to the hack An Ethernet switch has anywhere from 4 to 48 ports where Ethernet devices like PCs and IP phones can connect Each device connected to the switch has a 32 bit hexadecimal address called a MAC address Ordinarily the switch knows which port to send a particular packet to because address resolution protocol ARP has been used by the sender or by the closest router to address the packet to the correct MAC address All the switch has to do to get the packet to the right destination is transmit it on the port where the device with a matching MAC address is connected The secret to recording another person s VoIP calls is in making the switch think your MAC address is a valid destination for the VoIP traffic that s bound for that person s VoIP device Specifically when the sending device uses ARP to resolve the IP address of the intended recipient your PC must respond by saying I am the holder of that IP address send the data to me This hack which rather goes against the prescribed way an Ethernet LAN is supposed to
318. m that repository known as a TFTP server With a TFTP server you can centrally store and manage an entire network s worth ofIP phone configurations Merely placing a particular phone s configuration file on the TFTP server will change the phone s functionality to match the new configuration file the next time the phone is booted Many ATAs can be configured this way too In addition firmware updates for IP phones and ATAs can be delivered by TFTP So clearly understanding a little about TFTP and learning how to set up such a server is useful for any VolP network 6 10 1 Set Up a TFTP Server TFTP servers can be hosted on Windows Mac Linux and BSD machines and on Cisco routers too and there s a host of free software and plenty of good shareware to enable a basic TFTP server for your test lab For a more robust TFTP server use the age old tftpd software that s included in most Unixes To set up a simple TFTP server on Windows download TFTP Desktop from Weird Solutions This is a limited version of Weird s commercial Windows TFTP server and it allows one transfer at a timeenough to satisfy the needs of a VolP network with a half dozen IP phones or so You can find TFTP Desktop at http www weirdsolutions com weirdSolutions pages 02products download index htm Mac users can turn to my favorite TFTP server Fabrizio La Rosa s aptly named TFTP Server available at http www macupdate com info php id 11116 Like TFTP Desktop for W
319. man gt user hansolo Sastman gt secret falcon Sastinan host localnos localhost IS used to specify the host that the Perl script will connect to in order to send messages to the Asterisk server in this case it ll connect to the same machine as the one where the Script is running Run the script like this manager test pl If you place a call to any of the channels on the Asterisk server the script will give you output like this via its connection to Asterisk Manager Event Newchannel Uniqueid 1121992811 1 Callerid lt unknown gt Channel Zap 3 1 State Ring Event Newexten Channel Zap 3 1 Context default INDE a ae E Extension s Application Answer appdata Unigqueid Priority 1 Event Newstate Callerid Cleveland OH lt 4403281441 gt Channel Zap 3 1 state Up Uniqueid 1121992811 1 downloaded from lib ommolketab ir downloaded from lib ommoltketab ir This output says that the Asterisk server has received a callfrom 440 328 1441 on channel Zap 3 assigned it a unique ID for tracing it among the other Asterisk Manager output and indicated that is being handled by extensions the default extension in the default context The State Ring bit indicates that the channel is merely ringing and hasn t yet been answered TheApplication Answer line indicates that in accordance with the dial plan in etc extensions conf this call is being handled by the Answer Command Quite a lot
320. manufacturer of your ATA to see if you can reliably connect more have two analog phones and two cordless phones which receive their power separately anyway connected to an 8x8 DTA 310 and don t experience any problems Devices that use analog modems to communicate on traditional phone lines like older TiVo boxes and fax machines can t be used with the analog service provided by an ATA The fault lies with the analog to digital conversion ofVoIP codecs not with the modem itself downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 4 Use a Softphone with a VoIP TSP iE Get started with prevalent and freely available SI P softphones Depending upon which TSP you choose for yourbroadband VolP service your service agreement might limit you to using only analog phones connected to an ATA However if you have a lenient Bring Your Own Device BYOD service agreement your TSP will allow you to use your choice ofi P telephony access devices This might mean you can use an IP phone aPC softphone an ATA of your choosing or even your own telephony server Chapter 4 is dedicated to this proposition with the TSP s service This hack will show you how to use Counterpath s X Lite softohone with your TSP But first a little background on telephone networks both analog and VolP When you subscribe to broadband VolP service what you re really doing is buying a single pathway through the TSP s network Likew
321. me Phone Hack 40 In the default Asterisk context for the phone you re going to be calling from add something like this Exten gt 71 1 Dial Zap 1 71 Excvem gt 72 2 0i1a1 4407 1 7 72 Eaten gt 73 3 01al 4ap 1773 If you had buddies with speed dial numbers of 71 72 and 73 the Asterisk server would attempt to call them on Skype via the connected Internet Phone Wizard Of course all the Asterisk box sees is the Zaptel interface To be even slicker you can assign an entire range of numbers to be used for Skype purposes Here I ve set aside 80 through 89 The dial plan will always dial these extensions ol the Zaptel interface passing the extension number through to the Internet Phone Wizard as dialed digits downloaded from lib ommolketab ir downloaded from lib ommotketab ir exten gt 8X 1 Dial Zap 1 EXTEN To get calls from Skype routed into your Asterisk dial plan all you need to do is modify the default context of the Zaptel channel you ve used to connect the Internet Phone Wizard Refer to Connect a Legacy Phone Line Using Zaptel Hack 44 for an example that points out how to do this Now if you really want to get fancy with Asterisk and Skype check out the next hack downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 99 Forward Your Home Phone Calls to Skype Jaso For those times when you really really need to stay in touch This is just plain cool I
322. me server so the credentials are often identical downloaded from lib ommolketab ir downloaded from lib ommoltketab ir ProgrammableKeyl OneTouchDial ProgrammableKey2 TwoTouchDial ProgrammableKey3 CallForward ProgrammableKey4 DoNotDisturb ProgrammableKeyS VMA ProgrammableKey6 Mute The ProgrammableKeyl through ProgrammableKeys settings allow you to assign functions to the UIP200 s hotkey Here s what the possible values do OneTouchDial Causes the phone to dial a phone number supplied in theoneTouchKey settings later in the file TwoTouchDial Causes the phone to dial a phone number that s associated with one of the 10 digit keys on the dial pad these 10 numbers are supplied later in the file CallForward Enables call forwarding if supported by the softPBX to which the phone is connected DoNotDisturb Causes the phone not to ring even when calls are received if voicemail is available courtesy o your SIP proxy it will answer calls instead otherwise the calling party gets a busy signal VMA Voice Mail Access Causes the phone to dial a number associated with retrieving voicemail messages The exact number is specified later in the file Mute This is a standard telephone mute setting that disables the microphone in the phone so that it won t pick up input on your end OneTouchKeyl 18005551212 OneTouchKey2 411 TwoTouchDigitO 3000 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Tw
323. mention of Voice over IP dates back to early 1995 A good deal of early IP telephony research is probably quoted in the Google Groups archives so if you re interested in the history of VoIP this is a fantastic source After all as Gavin DeGraw sings Part of Knowing where I m going is Knowing where came from downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 22 Telephonize a Sound File 422 This trick is useful for taking on hold music for a test drive or just for making recordings sound like they re coming through a telephone Whenever you work with telephony be itdesktop telephony apps or full fledged IP phone systems you re bound to encounter prerecorded sounds things like on hold messages voicemail greetings and even elevator music are often generated by computerized telephony applications You might even need to create sounds that can be used with these apps Generating your owntelephony ready sounds is a snap using desktop recording software You can even resample your recordings so that you ll know exactly how they ll sound in a VoIP environment This way you can preview them Sound effects producers who need to make somebody s voice sound as though it s being heard through a telephone employ a technique calleddownsampling This gives recordings that tinny telephone flavor For a perfect phone sound simulation you d also need to chop the high and low frequencies of the sound using an eq
324. mns that contain CDR data Then return to the wizard window and click Next The final step asks you where you want to put the report choose the option to place it in a new worksheet Then click Finish As shown in Figure 4 7 you ll now Nave a pivot table toolbar with the names of your CDR columns on it Figure 4 6 The CDRs imported into Excel downloaded from lib ommolketab ir downloaded from lib ommotketab ir aePe _ Master CDR csv H 1 i j l K i L SPEI Dial IAX2 1990m icrotech 19 L2fO3 19 48 5 12 05 19 46 a l2f0S 19 49 LFE Dial LAMZ 199 Q microtech 19 5 12 05 19 57 5712 05 19 57 5 12 05 19 57 Cree Dial IAX2 1990m icrotech 19 5 12 05 19 58 y 12 05 19 58 127 05 19 58 76 Dil IAX 2 1990 m crotech 19 S il2 05 20 00 S i2 05 20 00 57 12 05 20 01 Cree Dial TAX2Z 199 m icratech 19 S 12 05 20 02 S 12 05 20 02 5 12 05 20 03 Ere Dial 1AK2 159 microtech 19 12 05 20 05 S 12 05 20 05 6 12 05 20 05 es Dial LAX2 199 miicrotech 19 5712 05 20 06 57 12 05 20 06 5712 05 20 06 ptt Dial 1AX2 199 miicrotech 19 S 12 05 20 13 5 12 05 20 13 S 12 05 20 13 ST S 8 Dial JAM2 199 mierotech 19 5 12 05 20 23 5 12 05 20 23 S 12 05 20 23 Ei Fe Dial AR 199 microtech 19 5 12 05 20 25 5 12 05 20 25 5 12 05 20 25 TEJ Hangup 5 13 05 14 58 5 13 05 14 58 5 13 05 15 00 Rives voice MailMan 201 5 13 05 20 56 5 13 05 20 56 5 13 05 20 56 Bee OI Ma Me in 200 S 13 05 20 56 S 13 05 20 56 S 13 05 20 57 466 elt IAXZ 199 Gm lcrate
325. motketab ir Hack 44 Connect a Legacy Phone Line Using Zaptel aa You don t need to buy a VoIP phone to make use of Asteriskuse your home phone instead About a dozen interface cards support the Zaptel standard allowing you to connect something as simple as a two wire analog telephone or something as sophisticated as a digital T1 breakout box called a channel bank for connecting digital business phones In this hack you ll see how to connec a phone line using such an interface cardeither an X100P or a TDM400P both manufactured by Digium To build a Linux PBX that can communicate with the PSTN the network that 90 percent of the world still uses for telephony you ll need at least one trunkchannel to communicate with the FXS interface in the phone company s central office switch This channel will provide you with a dial tone from your local phone company so calls to and from the PSTN can be handled by theAsterisk server There s not much to setting up an FXO channel with Asterisk One way covered in this hack is to install an X100P TDM4OOP or similar FXO line card in the Asterisk server The other way is to use an FXO media gateway Hack 43 The connection from your premises to the phone company s switch is called the local loop 4 5 1 Install an Interface Card To get started you ll need to obtain an Intel i537 based FXO interface card such as the Digium X100P If you d like to save a few bucks and build your own X1OO
326. motketab ir Once you have modified the zapata conf file connect your home line to the FXO port To test go off hook and try dialing an extension off the Asterisk system from the rotary phone Remember that using a rotary phone takes a little longer to dial so be patient Once you have confirmed that you can dial internally access the outside line configured for pulse dialing and call someone 5 14 3 Pass Through Pulse Dialing Signals With many of the other VolP product manufacturers the rotary end to end solution is not as easy to configure or operate Most VolP gateways support pulse dialing on the FXO connections to the phone company But they typically do not support pulse dialing with the FXS connections i e the phones A simple hack to overcome this limitation at least in theory is available on some of these gateways One could assign a dedicated FXO port that is enabled for pulse dialing directly to an FXS port The FXS port in turn would automatically seize the outside line when the rotary phone goes off hook In this way the pulse signals sent by the phone are passed through to the phone line The drawback of this configuration is that you will not be able to dial internal extensions directly Oh and you ll be scratching your head if your local central office doesn t support pulse dialing Make sure your local central office supports pulse dialing or this hack definitely won t work 5 14 4 Do Pulse Without Any Special Har
327. mware Note theAllow Allow Events and Supported headers showing all of the different SIP functionality that this host supports This is the intended purpose of the opTIONs request in SIP to determine the functionality of a remote SIP service OPTIONS IS Just One of several methods by the way methods is SIP s word for requests that SIP implements The rnvITE method is used to establish calls and the suBScRIBE method is used with presence capabilities like user availability and location These methods are akin to GET and POST in HTTP And like HTTP the host receiving the methods has a variety of numericresponses In this case 200 represents that the method was successful and the response message contains the desired information Like HTTP the response for resource not found is 404 So if a SIPINVITE was sent for a user that didn t exist on a particular SIP device the response would be a 404 If you point this script at other SIP hosts you ll see a large variety of responses due to the variation in behavior of different SIP implementations on the Internet and due to the variations in available functionality from one SIP host to the next A SIP video conferencing device might not support the Same methods as listed on the Allow line as a SIP phone for example 6 3 1 See Also e The official SIP specification athttp www fags org rfcs rfc3261 html e Switching to VolP O Reilly e Practical VolP Using VOCAL O Reilly Brian Degenhard
328. n electromechanical device called a stepper switch The stepper switch allowed thecalling party to control whom he would connect to without the need for an operator Today that control mechanism might be analogized to phone number But the important point is that Strowger s invention made it possible to connect phone calls without a telephone operator s intervention SIP promotes the ability to call a party without the need for a SIP gateway or gatekeeper as long as you know the recipient s sip address called a SIP Uniform Resource Indicator URI To pay homage to the pioneering Mr Strowger we have included this hack to connect a rotary phone such as a 1920 Western Electric candlestick phone Figure 5 5 to a VolP network First let me give you a quick overview of how a rotary phone works The rotary phone provides signaling to the central office by establishing a flow of current and then interrupting the flow momentarily to signal a pulse The number of pulses produced in a given time period represents the number being dialed Five pulses represent the number 5 Adding a rotary phone to a VolP network is a relatively painless process as long as you can build an intermediary gateway that understands pulse signaling and SIP and can signal both legs of the call What can be painful is finding a VolP device thatsupports rotary or pulse dialing That s why you might be better off building this gateway yourself 5 14 1 Do Pulse with an IA
329. n the dork calls for a date he or she will instead get a professional rejection courtesy of theRejection Hotline Aside from being cruelly entertaining the Rejection Hotline provides a great demonstration of a large scale soft based voice system By the time you re done with this book you ll probably have enough VoIP chops to start your own version of the Rejection Hotline 2 2 4 Broadvox Direct Users Can Use Find Me Follow Me so that They Can Be Reached Wherever There s a Phone You bet When you subscribe to the Broadvox Direct VoIP service you get aweb based toolset that lets you configure a find me follow me call list That way when folks call your home phone the service can attempt to track you down on your cell phone at Mom s housewherever you might be downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 9 Track Vonage Account Info on Your Desktop JE This tiny desktop tool helps keep track of your minutes and voicemails too If you ve never used Konfabulator now known as Yahoo Widgets or Apple s Dashboard widget system you should try it out Widgets are very simple specialized desktop apps that provide short useful information in real time They can be floating windows or they can be embedded into your desktop Remember Active Desktop from 1997 which let you dock an informational web page into your Windows desktop Well widgets are about nine times better The widget experience i
330. nager recording calls 2nd reporting telephone activity with Excel routing calls with distinctive ring 2nd 3rd software components for Linux teaching Asterisk box to speak 2nd 3rd 4th tuning up logs voicemail greeting updates 2nd 3rd 4th voicemail server web based administration interface 2nd asterisk modular software daemon Asterisk Gateway Interface Asterisk Manager API commands asterisk perl module Asterisk Manager example Asterisk AGI Perl module Asterisk Manager class AstLinux 2nd 3rd 4th 5th 6th 7th addmailbox script current features hardware requirements installing from CD ROM installing from Linux installing from Windows keydisk PBX only mode AstShape Provider script 2nd downloaded from lib ommotketab ir downloaded from lib eommolketab ir AstShape script 2nd 3rd 4th ATAs analog telephone adapters automatic registration with VoIP service provider choosing your own connecting analog phones to Mac mini PBX connecting to your network connection to modular phone jack 2nd keeping firmware up to date 2nd listing of ATAs that provide media gateway number of phones connected placement of providing dial tone to your analog phones Sipura SPA 3000 audio chat 2nd audio fidelity Audio Hijack Audio Voice Cloak auditing for VoIP network QoS Quality of Service using traceroute authentication Authentication ID Authorization User setting X Lite auto attendant for calls Call Soft
331. nail s pace it took my batch of only 48 contacts about 10 minutes The next toolbar item changes depending on the contact of the message or the contact selected Say you re navigating your Inbox and you open a message from somebody whose phone number is in your address book This button changes to show the phone number of the person who sent the mail and enables you to Skype call that contact via SkypeOut with one click If the contact is in your Skype buddy list this button will call the contact via his Skype name If you have neither a phone number in your address book nor a buddy list entry for this contact the button will attempt to look up the contact s Skype name via the Skype directory So you re covered in any event unless your contact doesn t have a telephone or a copy of Skype installed The last button on the toolbar anA lets you launch a text chat with the current contact too Figure 3 2 Exporting your contacts to Skype Skype Import Contacts Import Contacts to Skype The wizard is searching for contacts in your addressbooks Please wait You can minimize this window while Skype is searching For your contacts Skype will inform vou when the search is complete Found so far c 48 contacts from Microsoft Office Outlook Mow we will search the Skype network For these users This might take several minutes Statys Searching Skype network 31 completed Previous 3 3 2 Journal Your Skype Calls
332. nap on Mac OS X as long as you have Ambrosia Software s free trial version of WireTap Pro or Rogue Amoeba s Audio Hijacktwo applications that permit you to siphon audio produced by desktop applications into sound files in real time In this case we ll use the X Lite softohone Hack 4 and Wiretap 1 0 to make an AIFF sound file that contains an incoming caller s side of the conversation With this hack you can record anaudio chat on your Mac To make a recording that mixes both ends of the conversationyour voiceand the caller s voiceyou ll need more elaborate sound recording software like the pay for version of Ambrosia s WireTap WireTap Pro or Rogue Amoeba s Audio Hijack Pro Both applications allow you to record from the Mac s audio output for grabbing the caller s voice and its audio input for grabbing your voice First we ll need to get WireTap http www ambrosiasw com utilities wiretap set up to make the recording Since the audio fidelity of a phone call on the X Lite softphone isn t likely to be higher thar 8 bit 8 kHz we ll configure WireTap to save using the same fidelity Once you ve launched WireTap go to Preferences from its application menu and note the soundcompression settings By default the fidelity will be 44 1 kHz stereo and 16 bit sampling depth as shown inFigure 2 15 Click the Settings button to change that As shown in Figure 2 16 you can drop the sample rate and depth to match an appropriate level fo
333. nce You can answer such calls on any of the three to boot But sometimes you might not want to be bothered Say that you re out on your million dollar pleasure yacht in some exotic port being fed plump red grapes while lounging behind your Ray Bans Now would not be a good time to receive a Skype alert unless it is from the ship s master chef informing you that your filet mignon is ready Times like this call for the Skype Answering Machine SAM at least if you re a Windows user though the Skype API exists for Mac SAM is only for Windows This software add on for Skype is a fully featured answering machine that can record your callers messages and even greet unknown callers differently from callers that are already in your buddy list Grab it from http www freewebs com skypeansweringmachine Any time you add a new Skype add on like SAM Skype will prompt you to grant permission for the new add on to access Skype This is a security precaution that s built into Skype so don t be alarmed If you ve used Windows for any length of time you re probably quite accustomed to these security warnings It s a tiny download Close Skype before you run the installer Once it s installed you ll see a small green SAM icon in your system tray Double click it to launch the user interface and the first thing you ll see is SAM s call log dialog This is where each of your calls will be logged and you can listen tc the messages that folks leave
334. nd Call411 will even audibly announce incoming callers phone numbers if you wish You can obtain Call411 from http www soft411 com company Soft411 Call411 htm downloaded from lib ommolketab ir downloaded from lib ommotketab ir Figure 2 8 Call411 is a basic caller I D display tool Mr John Doe 1 234 555 2438 iS Settings allt 4 l oe x xj SER 4 0 Enable Ring sound A ELETT _ Start ringing when a Caller is IDentified Last dy 01 35 21 Enable Textto Speech wew otk 1 com NAME NUMBER Example NAME ANUMBERS e Onhy say number if no name is received Tools like PhoneTray and Call411 are ideal for traditional phone service but Should work OK with VoIP services like Vonage and Packet8 too since their ATA behaves just like a traditional phone line with caller ID and all 2 8 3 Call Soft and Call Soft Pro If and when you outgrow Call411 and PhoneTray s features and you find yourself wanting a complex voicemail and auto attendant to handle your calls you can graduate to a more capable commercial application such as Call Soft Pro available fromTOSC http www toscintl com and shown in Figure 2 9 This full featured telephony system lets you tap into many of the standard calling features on your phone line Even things likedistinctive ring are supported Call Soft Pro is a message recorder and interactive voice response tool so your ca
335. ndards based messaging apps recommend J AJ AH in addition to Gizmo It s a Windows based SIP softphone application that lets you call traditional phone numbers like Gizmo but alsosupports the AX protocol and Skype giving it the ability to communicate with several VolP networks simultaneously I ve always had to maintain several instant messaging accounts to keep in touch with all of my online buddies on Yahoo AIM and ICQ To avoid running several instant messaging clients have adopter Trillian a Windows based instant messaging client that can talk to all of these networks letting me manage all my IM activity from a single interface A similar Mac multinetwork IM tool is Adium As Skype Gizmo and other VoIP networks grow you ll probably need to attach to them simultaneously as you would in the realm of IM J AJ AH lets you connect to VoIP networks using several major standardsSIP AX and Skypesimultaneously saving you from having to run several VoIP clients at the same time downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 13 Record VoIP Calls on Your Windows PC HAGK 13 Unless you re using Gizmo you probably can t record your VolP calls without a little outside software assistance If you constantly forget things which do or you re a private investigator which I m not you might have wondered how to record calls so that you can listen to them later Recording calls on tradit
336. ndy later To figure out the length just launch the file in your favorite sound player It should show you the length If you use copyrighted music for commercial or nonprivate use you ll need permission from the artist who created the music or you ll have to pay royalties for using the music Next record your announcement using Windows Sound Recorder Cacophony or your favorite sound recorder Drop the resulting file in the same folder with your music file Ohand you might want to note its length too Use these two commands to perform some conversions on your raw audio files sox bg_music mp3 xr 14400 c 1 bg_music wav sox announce wav r 14400 c 1 announce wav These two commands grab the files resample them both to 14 400 kHz and make them mono c 1 not stereo sounds ready to be mixed together The following command mixes the two resultant files into a single file downloaded from lib ommolketab ir downloaded from lib ommotketab ir soxmix bg_music wav announce wav mixed wav If you find that the background music is too loud or soft you can adjust it and remix the files sox bg_music wav v 0 25 bg_music_quiet wav soxmix bg_music_quiet wav announce wav mixed wav Then all that s left is to get the file into the format you need for yourtelephony application If it s for an Asterisk announcement you ll probably want it in GSM format at 8 000 kHz sox mixed wav r 8000 mixed_for_asterisk gsm Convertin
337. ne and VolPfan com Ted is a strong advocate of open standards andStar Wars movies He updates the web site http www macvoip com at least a couple of times a year Ted lives with his wife Kelly and two amazing kids Jacob and Madelyn in suburban Cleveland downloaded from lib ommolketab ir downloaded from lib ommotketab ir Contributors The following people contributed their writing code and inspiration toVolP Hacks Brian Degenharadt Brian s experience in the high tech industry includes work in such diverse areas as network engineering online media delivery and console game development Currently he serves as CTO of Four Loop Technologies maker of the Switchvox PBX Brian has contributed to numerous open source projects including the GIMP and the Squid web proxy cache He currently resides in sunny San Diego with his wife Tristan Kristian Kielhofner Kristian is president of KrisCompanies http www kriscompanies com a consulting firm based in Lake Geneva Wisconsin and creator of AstLinux a Linux distro configured specifically for Asterisk that features a very small footprint Kristian has been working with Linux professionally for more than five years since he began doing Linux system administration at the age of 16 Kristian started KrisCompanies in 2004 to help local businesses with their technology needs In addition to creating AstLinux he has also been involved with AstShape and Polycom configuration files
338. ng VoIP is packet jitter Jitter is the difference in time that packets take to arrive at the final destination The greater the difference the worse your calls will sound and the more you ll want to hang up that phone and return to a traditional telephone system But it doesn t have to be that way Instead of throwing your hands up and giving up on the converged network dream you can just get serious about the jitter problem and the first step is identifying instances of jitter on your network You ve seen how to do this in a big picture fashion usingRRDtool and sip ping pl Hack 75 but that technique has a few shortcomings Though it gives you a long term assessment of jitter conditions on the network it doesn t do so using RTP the protocol that carries real voice payloads and it takes samples only every five minutes meaning that you can t assess the jitter conditions for any given call This is where Ethereal can really help youout 6 13 1 Identify Jitter If you ve looked at Graph Latency and Jitter Hack 75 you ve already seen the face of the enemy With some help from Ethereal you can zero in on jitter and prepare to squash it like a bug When you re examining jitter you re mainly concerned with RTP packets To use Ethereal here you must first locate an RTP packet in the trace file screen You ll need to have grabbed a packet sniff like the one you grabbed in Peek Inside of SIP Packets THack 81 Once you find an RTP p
339. ng VolP latency and jitter In addition to network latency it will also measure any latency injected into the system by the SIP application listening on the other end However since SIP based VolP relies heavily upon another protocol altogether to carry the actual audio streamsReal time Transport Protocol RTP measuring latency and jitter merely by sending SIP messages isn t foolproof Yet since SIP and RTP are almost always UDP and ping packets are not this kind of measurement is better than using ping and traceroute Plus many VolP aware routers give the same preference to SIP as they do to RTP so the results you get using this technique might not be too far off downloaded from lib ommolketab ir downloaded from lib ommotketab ir 6 5 1 The Hack Timing the round trip of SIP with Perl is only half of this hack The second half is using a Unix program called RRDtool http people ee ethz ch oetiker webtools rrdtool to graph the data RRDtool is a generic utility for graphing data over time You ll need to build it by following the instructions at the author s web site Once you ve installed RRDtool on your Linux PC you ll use it to graph the SIP latency of three different SIP providers The first thing to do is create therrd database file which RRDtool will use to accumulate the data you ll graph later Srrdtool create voiphacks rrd s 300 DS providerl1 GAUGE 300 U U DS provider2 GAUGE 300 U U DS provider3 GAUGE 3
340. nside of SIP Packets Hack 82 Dig into SDP Hack 83 Sniff Out Jittery Calls with Ethereal Hack 84 Log VolP Traffic Hack 85 Secretly Record VolP Calls Hack 86 Log and Record VolP Streams Hack 87 Intercept and Record a VolP Call downloaded from lib eommolketab ir downloaded from lib ommotketab ir 6 1 Hacks 7287 Introduction Switching to Voice over Pespecially in an enterprise environmentis wrought with perils that you won experience on a non VolP network Real time applications like voice require a high quality real time network And at least by itself traditional Internet Protocol IP networking gear doesn t fully deliver on that promise Fortunately you can apply some old toolssuch as Perl and Etherealto VoIP networking to troubleshoot and improve your IP network When problems occur your trusty old network troubleshooting apps will come to the rescue In this chapter you ll use Ethereal to sniff Session Initiation Protocol SIP signaling messages as they traverse the network inside of User Datagram Protocol UDP packets If you re a seasoned hacker or a timid script kiddie you can start with this chapter s Perl scripts which graph and monitor VolP activity on the network You ll also be able to monitor latency and jitterthe two things VolP admins want to avoid like the plagueusing standard IP networking commands If you play your cards right you might even learn how to beat a SIP mangling firewall OK you
341. nt 4794432 IRER Packets Rec 30047 RTP Bytes Recw 4770680 SIP Messages Sent 17628 SIP Bytes Sent 6296556 SIP Messages Reew 17761 SIP Bytes Recy 6485410 External IP Line 1 Status Hook Stata On Registration State Not Registered Last Registration At Next Registration In Message Waiting No Call Back Active No Last Called Number atai 192 168 0 132 5060 Last Caller Number atal Mapped SIP Port Call 1 State dle Call 2 State Idle Call 1 Tone None Call 2 Tone None Call 1 Encoder Call 2 Encoder iCall i Decoder Cali 2 Decoder Call 1 FAX Call 2 FAX iCall 1 Tyne Call 2 Type Call 1 Remote Hotd Call 2 Remote Hold Call 1 Callback Call 2 Callback Call i Peer Name Call 2 Peer Name Call i Peer Phone Call 2 Peer Prone Call 1 Duration Call 2 Duration iCall Packets Sent Call 2 Packets Sent Table 5 2 Top 10 Sipura options Recommended Tab title Option name Explanation value System cma pool ntp org Sets the SPA s clock automatically Server Admin System Make it up Sets the admin password Password downloaded from lib ommolketab ir downloaded from lib ommoltketab ir 7 Tab title Option name pecommende Explanation value Sets the SIP timeout value Crank this up for SIP T1 a a O latency network connections Regional Time Zone Zone Your time zone time zone Sets Sets the SPA s time zone SPA s time zone IP address of syslog System Syslog Server
342. o Capture Anmahyze Slatictics Help No Time Source Destination Protocol Info i 6 000000 10 1 1 202 10 1 1 100 UDP Source port 5060 Destination port 5060 2 6 862158 10 1 1 202 10 1 1 10 SIP SOP Request INVITE sip 201610 1 1 10 with session desc 3 6 863717 10 1 1 10 10 1 1 202 SIP Status 407 Proxy authentication Required 4 6 870328 10 1 1 202 10 1 1 10 SIP Request ACK s1p 201810 1 1 10 6 870407 10 1 1 202 10 1 1 10 SIP SOP Request INVITE sip 200610 1 1 10 with session dese 5 6 671769 10 1 1 10 10 1 1 202 SIP Status 100 Trying Ae a es es os a a gar SsLPsSoP Status AUU OF With session deseripricr 8 8 660598 10 1 1 202 10 1 1 10 SIP Request ACK s1p 201810 1 1 10 9 10 239491 10 1 1 202 10 1 1 100 UDP Source port 5060 Destination port 5060 10 15 696749 10 1 1 202 10 1 1 10 SIP Request BYE 510 201010 1 1 10 11 15 0697314 10 1 1 10 10 1 1 202 SIP Status 200 OK EB session Description Protocol S session Description Protocol version Cv 0 5 wner cCreator Session Id Co root 1702 1702 IN IP4 10 1 1 190 Owner Username root Session ID 1702 Ses i0n Version 1702 owner NetWork Type IN Owner Address Type IFA owner Adress 10 1 1 10 session Name si session Connection Information Cc IN IP4 10 1 1 10 Connection Network Type TN connection Address Type IP4 Connection Address 10 1 1 106 B Time Cescriotion active time tj 6 4 Session Start Time 0 Session Stop Time 5 Media Description mame and address mj aud
343. o s only phone model to support 802 3afthe 7970 This is an unfortunate form of vendor lock in but all is not lost You can do a couple of things to get Cisco IP phones to draw power from non Cisco switches If your budget permits the obvious though proprietary solution to this problem is to use Cisco PoE switches to power the phones Some other switch makers like Foundry Networks also support Cisco s proprietary PoE standard If you can t afford to forklift your switches you might instead want to power your Cisco phones by way of apower injector which is a patch panel that adds inline power to a CAT5 CAT6 cable connection Consider Cisco PoE compatible injectors like those made by PowerDsine http www powerdsine com But if you can t do that either do the next best thing hack Hacking inline power will almost certainly void your IP phone s warranty and probably your switch s or power injector s too A short circuit could fry your Switch and phone if you re not careful Proceed with caution By changing some wires on a standard UTP Ethernet patch cable you can make a compatibility cable that lets you plug Cisco IP phones into any 802 3af source as shown inFigure 5 3 Essentially you are flipping wires 4 and 7 and 5 and 8 Be advised this technique could void the warranty of your phone and your switch Figure 5 3 The wiring diagram for a hacked PoE cable downloaded from lib ommolketab ir downloaded from lib omm
344. o the other ASIP 200 ox response indicates agreement with the SDP parameters and a 4xx response indicates disagreement or incapability For a much deeper discussion on SIP have a peek inside my book Switching to VoIP O Reilly 6 12 1 Inspect Successful Capabilities Negotiation Using Ethereal configured with the same filter string from Peek Inside of SIP Packets THack 81 you can capture a successful capabilities negotiation In its default configuration Asterisk Supports G 11 so that just about any IP phone including X Lite can place calls to it In this case X Lite will be used to call Asterisk extension 201 and theSDP exchange for this call will be captured If you don t have such an extension on your dial plan you can call Asterisk s default auto attendant demo at extension 500 instead If you ve removed this extension in your hacking of Asterisk just run a make samples from your Asterisk source directory Hack 41 to get the default config back again When you place the call on X Lite use Ethereal to capture the SIP packets and zero in on the SDP content carried in the INVITE methods and 200 ox responses In Figure 6 9 you can see that the call setup was successful Figure 6 9 Ethereal can parse SDP content so that it s easier for you to troubleshoot call setup problems downloaded from lib ommolketab ir downloaded from lib ommotketab ir lntitled Fihereal ella Fle Ect View G
345. oesn t support saving the whole thing as an MP3 you can run the finished WAV version through SoxX see Chapter 2 for a refresher on SoX to make it ready for publishing to the Web S sox my_skypecast wav my_skypecast mp3 Figure 3 9 Total Recorder s parameters dialog downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Recording source and parameters Preset Untitled ave as Recording source f Software i Convert using Recording parameters specified below M Record alzo Input stream Internet telephony only Accelerate recording silent mode Speed Remove silence prevent Internet transmission gaps M Prisene pauses Recording level o of the orginal 100 Balance Le R Dz e g Sound board Default device f Usetheline Stereo Mix f Use multiple lines E i Soe ma a a e e el A ta ant fie ey ee oe eS one fe f Wee the linet elected an the mre Pao vanced Ft Recording parameters Format PLM Change Attributes aA 050 kHz 6 Bik Mono OF Cancel Help Chances are that your Linux box already has SoxX installed If not or if you don t have a Linux box you can grab Sox for Windows at http prdownloads sourceforge net Sox Sox12177 zip download 3 10 4 Three s a Crowd One of Skype s knock dead features is its multiparty conference call support Up to fifty participants can talk together just like a traditional conference call B
346. of useful information for a short simple Perl program Besides viewing status output using theeventloop method of the Asterisk Manager class yOu can also use the asterisk perl Perl extensions to issue commands to Asterisk Consider this simple Perl script usr bin perl Wee lib arip 7 172 use Asterisk Manager my Sastman new Asterisk Manager Sastman gt user hansolo Sastman gt secret falcon Sastman gt nost localhost Sastman gt connect die Sastman gt error n print STDERR Sastman gt command iax show peers The output of this script at least on my Asterisk server which has a single permanent IAX peer set up looks like this Name Username Host Mask Port Status mtech 199 66 46 190 45 S 2o5 200 e2 012 2569 Unmoni cored Of course with the command method of the Asterisk Manager class you can send any Asterisk console command and get its output So you can grab the whole dial plan usingshow dialplan or zaptel show channels to show all the current Zaptel activity on the system Once you get that output you ve got to parse it Otherwise your Perl program won t be able to do much more than print it out So recommend that you brush up on those great Perl text parsing functions If you re familiar with the Asterisk Manager API command structure you can also send API commands allowing you to originate phone calls hang up and transfer calls in progress and do other fun things
347. of your voice than a laptop s built in mic or a cheap 9 USB mic Of course a good mic onyour desk won t help your interviewee sound any better They ll need a good mic too Using different voice chat tools iChat Skype Yahoo Chat etc will certainly deliver different levels of compression and equalization and you might discover that you prefer one over the other Using different post processing tools GarageBand AcidPro etc willafford you greater flexibility in making your podcast sound the way you want it to But don t expect any of these applications to have the magical podcast preset You ll need to experiment until it sounds like what you think a broadcast Should sound like If you re talking a lot listen to talk radio for inspiration not only with respect to production but also the subject matter 3 10 6 See Also e Podcasting Hacks O Reilly downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 37 Answer Your Skype Calls Even When You re Not Around HACK 37 Thanks to the Skype API Windows and Mac developers can create useful tools to extend the functionality of the Skype platform One such tool is the Skype Answering Machine for Windows Skype makes you ultimately accessible In fact if you loginto your Skype account using Skype clients on three different computers on three different continents simultaneously any Skype calls ti you will alert you on all three computers at o
348. oits Asterisk provides modules for all kinds of signaling protocols and telephony applications anc you might not need them all Use thenoload directive in etc modules conf to specify those that you d like to disable noload gt pbx_kdeconsole so noload gt chan_modem so In this case the two modules being disabled are the KDE log console module which provides a graphical console for the KDE desktop environment and the modem module which is used for ISDN connectivity with Asterisk Keeping unnecessary modules from loading also conserves memory on the server 7 2 3 2 Clean up xinetd xinetd is Fedora s catchall daemon for Telnet finger and a number of other Unix network applications It s the successor to inetd Its configuration files in etc xinetd d are used to enable or disable support for a long list of network access services Use this configuration directory to disable all but those that you absolutely need Here s the contents of a file in this directory etc xinetd d imap controlling the imap daemon service imap disable yes socket_type stream wait no user root server usr sbin imapd log_on_success HOST DURATION log_on_failure HOST This particular service is disabled per thedisable yes line Check all the files in this folder for the disable yes line or if you prefer you can altogether remove the config files for the services you don t need downloaded from lib ommolketab ir
349. oleGuchbPigitcl 3200 TwoTouchDigit2 3002 You use the oOneTouchKey1 through OneTouchKey4 settings to supply the phone numbers that are used with up to four ProgrammableKey settings So you can set up to four of the UIP200 s eight hotkeys to be one touch dialing keys The TwoTouchDigit0O through TwoTouchDigit9 keys on the other hand are used to set up two touch dialing first the hotkey and then a number key on the dial pad Values supplied here become the phone numbers that are called whenever a two touch dial occurs VmaDirectCallNo 8080 VmwiLampIndicator Enable VmaDirectCallNo tells the phone what number to call when the VMA hotkey is pressed VmwiLampIndicator When Enabled permits the phone to light its message waiting indicator light It s probably not a good idea to disable this one Once you ve got this file set up the way you like save it in the formatuniden lt mac gt txt where lt mac gt is the Ethernet hardware address of the phone it applies to Then reboot the phone downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 70 Control the Lights Using Your IP Phone 70 Using an X10 phone controller you can turn your lights on and off from the comfort of your IP phone X10 home control interface equipment has been a favorite pastime of geeks for decades Since the early days of 8 bit hobby computers you ve been able to automate your home using your keyboard and later your mouse
350. oltketab ir Make sure your switch lets you program port by port which ports get power and which ones don t because in a native Cisco PoE solution Cisco IP phone power requirements are auto detected so power can turn itself on and off as necessary on each port There s no such provision when using a hacked cable to supply 802 3af power to a Cisco PoE using phone If this is a problem and 802 3af won t work with the hacked cable try using a device that does the two pair flip but also works with auto detection such as 3Com s 48 volt IntelliJ ack switch converter part number 3CNJ VOIPCPOD l I 3 i _s b gt is ie popular and less expensive 7960 and 940 phones Ase The Cisco 7970 IP phone does support 802 3af power sources unlike the more downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 67 Customize Your Cisco IP Phone s Boot Logo 1x67 Change the logo on your Cisco IP phone and reflect your inner geek s refined sense of monochrome style Have you ever wished you could change the boot up logo on your cell phone Have you ever wanted to use custom graphics on your appliances LCD screens Most Linux geeks love to plaster Tux the Penguin the official mascot of Linux all over the placeand what better place than a hackable display If you re like me and you have a thing for the penguin allow our underdressed friend to show himsel on your Cisco VolP phones First th
351. om lib ommotketab ir In the preceding example the receiving PBX doesn t know the extension number of the party who is calling because the calling PBX supplies the caller ID signals for the Zaptel channel and phone line being used not the caller ID signals for the extension that originated the call So the receiving user will see that she is getting a call from the other office but she won t know which user is calling her Using some Asterisk dial plan wizardry you can preserve the original caller s caller ID information throughout the interswitch calling process West office extensions conf default exten gt _30XX 1 SetCIDNum S EXTEN exten gt _30XX 2 Dial Zap 1 5551300 35 mD EXTEN In this case Asterisk will supply the originating extension number as the caller ID number setcrDNum establishes the caller ID number for outgoing channels on the current extension This config would o course have to be mirrored for the East office too Mast office extensons cont default exten gt _34XX 1 SetCIDNum EXTEN exten gt _34XX 2 Dial Zap 1 5551340 35 mD EXTEN To override the phone company s caller ID you ll need to beusing PRI Signaling and the phone company will have to permit you to supply your own caller ID information There you have it a two office Asterisk PBX network that uses existing telephone lines to simulate a direct link between the two sites downloaded from lib ommolketab i
352. om 172 16 15 15 icmp_seq 2 ttl 53 time 79 0 ms 64 bytes from 172 16 15 15 icmp_seg 3 ttl 53 time 75 4 ms 64 bytes from 172 16 15 15 icmp_seqgq 4 ttl 53 time 77 8 ms This shows that there is a 74 to 78 ms delay between my computer and the remote VoIP server This delay is called latency In the context of VolP traffic it s not always a bad thing to have consistent latency Imagine if consistently have 100 ms of latency on my call That is it takes one tenth of a second for my speech to reach the ears of the person am talking to This isn t terribly noticeable However if this 100 ms delay suddenly evaporated to 50 ms and then jumped back to 100 ms this would definitely create audible abnormalities in my speech Every time it sped up the audio would Skip over the slower packets that hadn t arrived yet Every time it slowed down there would be a Slight pause waiting for more audio to arrive This is calledjitter and this is the true source of quality problems on the VoIP frontier However this is not to say that latency is not an important measurement Latency measurements can be a basis formeasuring the potential for jitter For example say have two hosts one with 200 ms oflatency and one with 5 ms of latency A 20 variation in latency will result in 40 ms of jitter from the first host and only 1 ms of jitter from the second One way to deal with jitter is to use ajitter buffer a device that basically delays playing or sending on t
353. oming call from SIP peer 7711 hits the server SIP peer 7712 the second line will be directed to call your cell phone and attempt to bridge the call know what you re thinking Why wouldn t just use my phone line s built in call forwarding service to do this My answer is Selectively Forward Calls Hack 46 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 46 Selectively Forward Calls 46 You can pass caller I D signals into Asterisk and have them acted on appropriatelyincluding auto ignoring the people you don t want to speak to By making some clever use of Asterisk s built in caller ID channel variable and a little workflow logic it s easy to turn your call forwarding project from the previous hack into something even more useful In this hack we ll make Asterisk forward calls to your cell phone only if they re from a certain caller ID That way you need only be bothered with answering your cell phone if dear old Mom is calling or your boss Asterisk refers to the one or more voice communication links of a phone call as channels So when a call forwarding setup that uses two SIP peers is active it s said to use two channels Each channel has with it a number of channel specific variables that contain information about the ongoing call When the call ends the channels and these channel specific variables disappear One of these variables iS CALLERIDNUM which contains the phone n
354. omplishes that in extensions conf TedsCalls exten gt s 1 Dial SIP 201 30 exten gt s 2 Voicemail 201 JakesCalls exten gt s 1 Dial SIP 202 30 exten gt s 2 Voicemail 202 downloaded from lib ommolketab ir downloaded from lib ommotketab ir There Ted s distinctive ring will send Ted s calls to SIP 201 and J ake s distinctive ring will send then to SIP 202 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 58 Tune Up Your Asterisk Logs 58 How much log detail is too much That depends on whom you ask Asterisk s log output can be pretty granular which is bad for disk utilization and good for troubleshooting Log analysis should be the core of your daily system monitoring and security activities Like other softPBX servers Asterisk supports flexible logging providing several levels of logging detail in severe different files It also Supports using syslog By default Asterisk stores its logs in var log asterisk You configure Asterisk logging in the etc asterisk logger conf file which Asterisk reads at boot time or whenever it is started The first section of the file is general where you can assign a value to the dateformat option to specify what date format to use in Asterisk s logs To figure out the syntax of the data formats read the manpage for strftime by running man strftime The next section logfiles describes which files should be
355. on the public Internet without having to have public IP addresses The Internet is designed such that most private networks that have access to the Internet do so using a device called a NAT firewall By using NAT computers on the private networks are afforded a certain amount of security when accessing the Internet because the NAT firewall can block certain network services and log all access attempts The most common form of NAT is calledmasquerading which uses a public IP address on the firewall s external interface to conceal all of the computers connected to the private interface In this way those computers masquerade as the firewall to the outside world The problem with this practice is that it breaks applications that rely on that public IP address When a private computer makes a request theserver handling the request might attempt to respond to the requester using the firewall s public IP address rather than the requestor s private IP address which is unknown to the responding server This wreaks havoc on many protocols including the file transfer protocol FTP and SIP Indeed SIP s poor NAT traversal capabilities are famous That is if you have a router that does NAT as most broadband routers do it can potentially cause problems with using SIP This is because the SIP protocol requires SIP hosts to instruct each other on how they can be reached by using theVia header which looks like this Via SIP 2 070De 127 0 0 1766
356. ook at it because now is the time to tell you that in a matter of moments we will be erasing everything on that device If this is not OK remove the USB drive and chose another one If it is OK to lose all of the data on this Flash drive move on to the next paragraph Now that we have those warnings out of the way let s finally create your keydisk by typing the following genkd The genkd script will take care of finding the device partitioning it formatting it and copying some base configuration files to it You should see some status information and messages go by but it Should be finished in no time returning you to the command prompt If you would like to verify it was successful type the following downloaded from lib ommolketab ir downloaded from lib ommoltketab ir ls mnt kd You should see a file there calledrc conf If it s there you should now type reboot to restart the system and begin using the keydisk If it s not there make sure that your device really is dev sda and that it is connected etc Once the system has booted back up you can start making configuration changes The etc rc conf file is where you are going to want to begin to look In an attempt to keepAstLinux small I ve included only the vi text editor as it was part of the BusyBox collection of utilities was already usinc http www busybox net If you re not comfortable editing text files withvi you can use the web interface To
357. ory you ll need to be logged into your Linux computer at a shell prompt as root Type these commands to run the CVS check out routine and download the source code cd usr srec export CVSROOT pserver anoncvs cvs digium com usr cvsroot cvs login cvs checkout zaptel libpri asterisk Ur Ur U UU Alternatively you can specify a particular version of Asterisk cvs checkout r v1 2 zaptel libpri asterisk When prompted use anonevs as a password If you don t use usr src as the local location for compiling programs substitute the appropriate path The CVS client you re running here will create the usr src asterisk directory that contains all the Asterisk source code Once the download completes you are ready to begin compiling Asterisk consists of several software components for Linux Not all of these packages are required as some of them are drivers for Digium s interface cards If you aren t planning to use Digium s cards you ll need to build only the last of the three asterisk libpri downloaded from lib ommolketab ir downloaded from lib ommotketab ir A driver module that supports Zaptel compliant interface cards described in this chapter s introduction so that ISDN and PRI trunks can be interfaced with Asterisk zaptel A driver module that allows legacy telephone line interfaces cards that provide FXO FXS and T1 E1 signaling to be used with Asterisk asterisk A modular software daemon that provides tel
358. ou acknowledge and understand that you cannot dial 911 from this line unless and until you have received a confirming email 2 5 Failure to Designate the Correct Physical Address When Activating 911 Dialing Failure to provide the current and correct physical address and location of your Vonage equipment by following the instructions from the Dial 911 link on your dashboard will result ii any 911 communication you may make being routed to the incorrect local emergency service provider This is a heavy handed contract item but what it means is that you have to use Vonage s prescribed email based activation routine to use its 911 call routing Of course I m not a lawyer and I can t provide an attorney s interpretation of this agreement so contact Vonage if you re unsure about it Other providers might handle 911 call routing similarly so make sure you ask before you sign up if 911 is a highly important feature The best way to deal with this intimidating contract is to know firsthand whether your TSP has you set up for 911 calling or be ready for an emergency in case it doesn t That s what you re about to do 1 8 1 The Problems with VolP Emergency Dialing With a traditional phone line the power for the line and phone comes from a central power source at the phone company s exchange switch This means that even during isolated power outages you car still make and receive callsincluding 911 calls With VolP your electric company and in hous
359. ou can simulate a direct link You can build a two office unified dial plan using two Asterisk servers This way a user need only dia the extension of the user at the other office to reach him instead of calling that office s main number waiting for prompts and then dialing the user s extension Asterisk can handle all of these steps automatically routing the call to the other office s PSTN trunk waiting until it s answered and dialing the recipient s extension to complete the connection Figure 4 11 illustrates just such a configuration Figure 4 11 Two offices with PBXs connected to the PSTN Ordinarily if a West user wanted to reach an East user he d have to pick up his phone dial the phone number of the East office wait for an answer and then request that user either by speaking with a receptionist or by dialing that user s extension This awkward process is shown inFigure 4 12 Direct Inward Dial could shorten the process but the dialing user still wouldn t be able to reach his co worker using a convenient four digit extension Figure 4 12 Figure 4 12 A caller has to dial a lot of digits to reach his downloaded from lib ommolketab ir downloaded from lib ommoltketab ir intended recipient at the other office 1 Dial 555 1300 cal Wait for IVR prompt we E POTS A Dial 3001 pors West fd PSTN a East PBK P Pa 3401 3001 4 16 1 The Configuration We ll use the same dial plan extension numbering convent
360. ou know SoX can deepen your voice like a pro announcer To do this you ll need to use SoX s shift pitch effect which takes a positive or negative integer as an argument If positive the integer will increase the pitch of the sound by the number of steps specified The higher the number the higher the pitch will be The lower the number the lower the pitch will be In this case we re going for pro announcer not chipmunk so start with a value of 2 and work your way down until it s deep but not artificial sounding sox announce gsm announce_lower gsm pitch 2 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 27 Record a Videoconference 427 Snapz Pro X lets you record video and audio togetherthe perfect way to record a video conference The term VolP usually refers to Voice over IP but it could easily mean Video over IP too since videc conferencing is such a popular use for the Internet today Tools such as Yahoo Messenger and Apple iChat allow you to do face to face video conferences across the Internet but one thing neither of these tools allows is recording your conferences Fortunately Ambrosia Software s Snapz Pro X lets you create a QuickTime file of any onscreen activities including a video conference Unfortunately it runs only on Mac OS X so Windows and Linux users are out of luck You can download a copy of Snapz Pro X from http www ambrosiasw com It will install as a backgr
361. ou re calling and one downloaded from lib ommolketab ir downloaded from lib ommotketab ir from that person back to you NAT breaks this too To make sure your SIP phone can receive voice packets behind the NAT box you ll need to tell the NAT box to forward incoming RTP traffic to your SIP phone s IP address For the sake of this example I m going to use the X Lite softphone a free STUN aware SIP softphone for Mac and Windows Download it fromhttp www counterpath com and install it Review Use a Softphone with a VolP TSP Hack 4 for a quick refresher If you want to attempt this hack with another softphone be sure it supports both SIP and STUN or no cigar Get the IP address of the computer where X Lite is installed Make a note of it I ll use10 1 1 50 for this example You ll need to tell your NAT firewall to forward all VolP traffic to this address If your NAT box runs Linux you can use the following commands to forward all the inbound VolP traffic to 10 1 1 50your IP phone This assumes the Internet facing interface is ethO and its address is 201 101 1 1 SO replace that address to suit your firewall iptables t nat A PREROUTING p udp i eth0 d 201 101 1 1 dport 5060 j DNAT to 10 1 1 50 5060 iptables A FORWARD p udp i eth0 d 10 1 1 50 dport 5060 j ACCEPT iptables t nat A PREROUTING p udp i eth0 d 201 101 1 1 dport 5061 j DNAT to 10 1 1 50 5061 iptables A FORWARD p udp i
362. ound application that you can summon with a special key combination that you ll assign right after it installs the first time it runs When Snapz s main window appears click the Movie button Here you can configure the size and aspect of the capture as well as the frame rate that will be use in the saved video file and whether to include sound from the audio output the conference participants voices or microphone input your voice You can crop resize and drag Snapz s viewfinder so that it wraps tightly around the area of the screen you want to record Just don t drag the window you re recording from out of the area of this viewfinder as it remains in a fixed size shape and location throughout the recording unless you ve configured it to follow your mouse movements which is probably not a good idea when recording from a video chat window Once you ve got everything positioned just right press the Return Enter key and Snapz will prompt you for a filename Enter this and click OK and recording will begin and will continue until you summon Snapz again using the key combination you established during installation Unless you change Snapz s default settings the saved QuickTime file will appear in the Preview immediately after the recording is complete If you want to view it later just call it up from the Finder To burn a video DVD of a Video over IP conference record it with Snapz and then import the QuickTime file into iDVD d
363. our phone will ring for 20 seconds and then the AGI script will run Depending on the time of the day and the presence of office presence file you should hear the appropriate greeting You should replace your regular voicemail greeting with downloaded from lib ommolketab ir downloaded from lib ommotketab ir the default greeting so that it flows properly once it reaches thevoicemail directive It doesn t take long to imagine some cool functionality with AGI scripts For example you can take this a step further and have an AGI script that reads your calendar that you ve published with iCal to see whether you re scheduled to be out of the office that day You can even have Asterisk announce to a caller when you have free time during that particular day and suggest that the caller try back then Dave Mabe downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 98 Connect Asterisk to the Skype Network Jaos They said it couldn t be done without the Skype API They were wrong Wouldn t it be great if your Asterisk server could place calls to and receive calls from the Skype network Imagine the possibilities putting your Skype buddy list within reach of the Asterisk dial plan so that all your calls can be routed to the appropriate Skype buddy depending on what you dial on your Asterisk connected phone Well that dream is now a realitywith a few gotchas The first gotcha is that you ll need to use
364. output format J ust about every sound codec you d want is supported from Windows Media to MP3 For even more sound conversion goodness be sure to check out Create Telephony Sounds with Sox Hack 24 Figure 2 6 Total Recorder can save audio recordings from MSN Messenger Yahoo Messenger AIM Skypeyou name it downloaded from lib ommolketab ir downloaded from lib ommoltketab ir F Untitled Total Recorder File Edit wiew PlayiRecord Process Tools Options Help O bed Pay kJ Cu r t ap Properties tatus Recording 48 000 kHz 16 Bit Mono Of il L De Data size bytes 1 313 260 A GREER ERA F dB Duration Taf In 0 Out 1002 Both O Fal 0 Recording source and parameters 2 Playing volume and recording lexe App Vol Level A 0dB 100 Advanced hoot Sa Mixer ae n Fostion auj for d n 137 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 14 Handle Calls with Windows Software Pig Have your PC screen your calls and take your messages with handy Windows tools If you ve got a Windows PC with a standard modem it doesn t have to be a voice modem you can use some really cool software applications that can identify incoming calls with their caller ID information shown on your Windows desktop Mac users can use Phlink for this purpose Hack 15 Some of these apps can even respond to incoming calls so
365. ow that we have told Asterisk what to do with incoming calls we need to tell Asterisk what voice mailboxes we want The voicemail application is configured with the fileyou guessed twoicemail conf As we did before open it with vi or the web editor In the general section uncomment forcename no and Set it to forcename yes ThiS option enables Asterisk to force a new user to record his real name when he first accesses his voicemail Asterisk determines whether a user is new by his PIN If his PIN and voice mailbox are identical Asterisk will guide him through setting up his voice mailbox Scroll down to the bottom ofvoicemail conf and create a new section that looks like this vmserv 8000 gt 8000 Lisa Hayes lisa rt com 8001 gt 8001 Rick Hunter rick rt com 8002 gt 8002 Lynn Minmei lynn rt com 8003 gt 8003 Max Sterling max rt com 8004 gt 8004 Miriya Sterling miriya rt com Here you are creating five voice mailboxes for a fictional group of five folks The fields in voicemail conf map out like so mailbox number gt PIN Real Name E mail address There are many more options but you will have to dig deeper into Asterisk on your own time to discover them You have only 30 minutes to get this done for me to be true to the title of this hack Now all that remains is actually creating the directory structure for the mailboxes AstLinux includes the addmailbox script from the contrib scripts directory of the Asterisk so
366. ower If your LAN uses DHCP the Sipura will acquire its IP address using DHCP If you pick up your telephone you should here a dial tone Enter You should hear a not so friendly voice say the words Sipura configuration menu At this point you should enter110 The same friendly voice should come back and read you your IP address Make a note of it downloaded from lib ommolketab ir downloaded from lib ommotketab ir While DHCP does make it easier to attach new devices it makes it harder to keep track of them Once you get into the web interface you should assign a Static address or use the static mapping features of your DHCP server to assign the ATAs the same IP addresses at all times After you have made note of the IP address for ATA1 repeat the process for your other Sipura ATA2 For the rest of this hack we ll assume ATA1 and ATA2 have the respective IP addresses 192 168 1 101 and 192 168 1 102 After you have the IP addresses of your Sipura devices fire up a web browser on a machine connected to the same LAN Using your web browser enter the Paddress of ATA1 You should see a gray screen filled with status information Open another window or tab and enter the IP address of ATA2 You should see a similar if not identical screen with the exception of the different IP addresses Now we re ready to have some real fun 5 6 2 Configure the Sipuras The dial plan on the Sipura ATAs is one of the more attr
367. ower cord back in again after a few seconds That should take care of the problem But do that every time Ma am bought this VolP Hacks book that taught me how to wire my ATA into my home phone wiring so that could replace my local phone service with Voice over IP and now I m very frustrated because every few weeks pick up the phone and the dial tone is gone have to run downstairs and reboot my ATA before can place any calls and I m a little frustrated the exasperated customer said Why is this happening Well it s actually quite simple The ATA receives an IP address from your DHCP server which runs on your broadband router she explained And your broadband router receives an address from your Internet provider s DHCP server That IP address can change sometimes when your DHCP lease expires breaking the UDP socket that connects your ATA with our network here at Ownage In English please the customer said Well the problem occurs because your ISP assigns you a dynamic address that periodically changes the support tech explained When it changes the ATA loses communication with our VoIP server So it s my ISP s problem No not exactly Most ISPs use dynamic addresses for residential broadband customers to prevent them from say hosting their own servers So they have their reasons for using dynamic addresses and there s little we can do about it she told him Then what do I do to stop it from
368. ownloaded from lib ommolketab ir nioaded from lib ommotlketab ir Chapter 3 Skype and Skyping nioaded from lib ommolketab ir downloaded from lib ommotketab ir 3 1 Hacks 2840 Introduction If you don t already use Skype you really don t know what you re missing Skype is the predominant desktop VoIP application a softphone and a peer to peer P2P network that operate over the Internet to link people of all stripes around the globe In fact Skype has become a verb as well as a noun You can use Skype to call people or you can Skype peoplehence this chapter s title Ohand yoli don t have to worry about finding somebody to call that s been my problem with iChat AV since Skype has been downloaded 150 million times and averages anywhere from 1 million to 2 million people logged in at a time Skype lets you make free Internet Protocol I1P based phone style calls to any other Skype user anc allows you through optional paid services calledSkypeOut and Skypeln to place and receive calls to and from regular phones via the Public Switched Telephone Network PSTN Skype s sound quality is often reported to be superior to that of a traditional telephone to boot Skype has several things going for it that other VoIP softphone solutions don t It s the only P2P softphone application that runs on Windows Mac Pocket PC and several flavors of Linux Fedora SUSE Debian and Mandrake anyway It s also the only softphone app
369. ows We ll start with the dial plan config for the East office Bast office extensions conf default exten gt _34XX 1 Dial Zap 1 5551340 35 mD EXTEN And we ll institute a mirror of that config so that West office users can dial 30XX extensions West office extensions conf default exten gt _30XX 1 Dial Zap 1 5551300 35 mD S EXTEN Let s dissect this exten directive First _30xx is a wildcard expression that matches any number dialec that begins with 30 1 is the extension priority The Dial command tells Asterisk to dial the number of the other office on the Zap 1 channel and to wait for up to 35 seconds for the call to be answered Then the D S EXTEN option tells the Dial command to send DTMF digits representing the extension number that was dialed by the user EXTEN IS an Asterisk variable that always contains the extension number used for the current call Finally as with allbial commands the call will be connected after the DTMF digits are sent The net result of this config is that the users at West can dial 30013099 to reach the users at East and the users at East can dial 34013499 to reach the users at West all without any PSTN dialing or auto attendant interaction Here the PSTN trunks are used like private trunks to connect two switches while the dial plan makes it easy for the users 4 16 2 Control Caller ID When Using PSTN Trunks downloaded from lib ommolketab ir downloaded fr
370. p of phones Or the distinctive ring can just be passed through to all of the phones on the private network which will ring distinctively and the intended recipient can answer her call on any available phone You can configure each Zaptel channelto detect up to four different distinctive signals The first thing you ll need to do is open zapata conf and add this configuration to the section for the trunk in question usedistinctiveringdetection yes Enabling distinctive ring on a Zaptel channel will cause a slight delay before Asterisk can answer incoming calls because the distinctive ring signals can take up to five seconds for the Zaptel channel to detect The signals used by distinctive ring consist of analog electrical cadencesvariations in voltage that cause analog phones to produce certain ring patterns Asterisk uses thedring attribute in zapata conf to describe the signals Unfortunately these signals vary from one regulatory jurisdiction to the next and you ll have to figure out what value to givedring attributes yourself Here s how When an incoming call is received on a POTS interface Asterisk records the ring pattern downloaded from lib ommolketab ir downloaded from lib ommotketab ir in Asterisk s verbose logging output assuming you launched Asterisk with vvvv on the command line Use the tail command with its f option to watch your logfile for changes as they occur tail f var log asterisk full While ta
371. p to Skype network 2nd placing and receiving calls via the Internet animated Answering Machine 2nd 3rd 4th rotating greetings with Windows AOL Instant Messenger AIM Apache use by AMP web based GUI Apple AirPort Express Mac mini 2nd 3rd 4th Apple iChat AppleScript caller ID for Phlink calling Address Book contacts with SKype 2nd 3rd application based QoS ARP Address Resolution Protocol ARP poisoning 2nd Asterisk 2nd 3rd 4th 5th 6th 7th 8th 9th 10th 11th 12th 13th 14th 15th 16th 17th 18th 19th 20th 21st 22nd 23rd 24th 25th 26th 27th 28th downloaded from lib ommolketab ir downloaded from lib eommolketab ir 29th 30th 31st 32nd 33rd 34th 35th 36th 37th 38th 39th 40th 41st 42nd 43rd 44th 45th 46th 47th 48th 49th 50th 51st 52nd attaching SIP phone 2nd connecting legacy phone line using Zaptel 2nd connecting PSTN phone line using Sipura SPA 3000 ATA 2nd connecting standard phone line using FXO gateway 2nd connecting telephony devices to connecting to Skype network dial plan commands for PBXs linked over the Internet Monitor command routing Skype calls to faxes 2nd 3rd forwarding home calls to your cell phone 2nd four line phone server 2nd 3rd 4th 5th getting daily weather forecast installing and testing server on Linux PC 2nd 3rd 4th integrating X10 controls with phone system 2nd LDAP client linking servers with PSTN 2nd configuration Mac mini PBX monitoring from Perl scripts 2nd music on hold 2nd 3rd PrivacyMa
372. pXpbx 2 8 1 fc3 sh When prompted by the script answer y and be sure to enter a password for the sipxchange user that the script creates That s all there is to installingsipXpbx Next you ve got to generate an SSL certificate for sipX to use mkdir SHOME sslkeys cd SHOME sslkeys usr bin ssl cert gen ssl keys sh usr bin ssl cert install cert sh server 01 usr bin ssl cert install ssl keystore sh server 01 Use the default password of changeit and answer yes when the script asks whether you trust the certificate 7 14 3 Launch sipXpbx Starting and stopping sipXpbx is a snap Use theservice command like this to start sipX and replace start WIth stop to stop it service sipxpbx start If sipXpbx complains about HTTPD syntax errors the first time you try to launch it just give your Fedora machine a reboot downloaded from lib ommolketab ir downloaded from lib ommotketab ir 7 14 4 Finish sipXpbx Setup by Web Interface sipX uses the JBOSS application server as the foundation of its excellentweb based GUI To access the web based site installation wizard in your web browser visit the following URL replacing sipx your domain with the address of your sipX server http 7 Sipe yvour comain 60s0 pds ui install instal Ip On the web page that appears enter installer as the username and password as the password Then click continue and you ll be greeted by the sipX Configuration Server as shown inFi
373. pe the following fdisk 1 dev sda downloaded from lib ommolketab ir downloaded from lib ommotketab ir You should see the partition table and drive layout information for your CF card Now it s time to burt the image At the command prompt as root type the following gunzip c path to imagefile img gz gt dev sda where path to imagefile img gz iS where you downloaded the image file to and dev sda IS where your CF card is located After the command completes and you are returned to the shell prompt you can remove your USB CF writer As with my Windows installs highly recommend that you read the AstLinux User Guide Because you didn t download a package you should go back to http www astlinux org and download the User Guide to familiarize yourself with AstLinux 7 9 6 Install from CD ROM The newest way to install AstLinux is via a more traditional means an install from CD ROM As with the other versions of AstLinux you can download the install CD ROM image from the Downloads section of http www astlinux org Once you have downloaded the ISO image you can write it to a CD ROM under Windows using such tools as Nero Under Linux cdrecord or a graphical frontend such as K3b works quite well Once you have written the ISO image insert the CD in the drive of your soon to be AstLinux machine Make sure that the machine is set to boot from CD ROM and power on Once the machine boots you should see a very sim
374. ple instruction screen Typinginstall and pressing Enter will start the AstLinux installer It will attempt to detect any hard drives in the system and prompt you as to which one you would like to install to You should choose your selection carefully as AstLinux will overwrite any data on that disk Actually the install portion will also detect USB CF writers as hard disks da sdb sdc etc This way you can boot the machine from the CD and write to an AstLinux CompactFlash card without ever touching the machine s hard drive 7 9 6 1 Don t install at all While someone who Is serious about setting up an AstLinux Asterisk server would not use this method in production the same AstLinux CD ROM image used in the preceding section can also be used as a Live CD This is actually the default Once you have created the CD you simply boot the machine and accept the defaults Hopefully in a matter of moments you will be running AstLinux without having to overwrite your hard drive And hopefully you will like what you see and decide to run the installer as mentioned earlier in this hack downloaded from lib ommolketab ir downloaded from lib ommotketab ir 7 9 6 2 More about the AstLinux CD ROM The AstLinux CD ROM also includes a nifty Windows Autorun portion that will give you access to the AstLinux User Guide a link to the web site and the tools and utilities provided by the Windows instal package Try it out 7 9 7 Boot Time
375. pops up a dialog confirming the addition of you as a new contact in her list aS6ript ere hlto ol00in 1 yve com 167 blugin 138 gt lt 7 scrip lt a href ADDCONTACT voiphacks gt lt img src http jyvetools jyve com addcontact gif border 0 onclick setENDown gt lt a gt Finally to make a link that allows surfers to send you a Skype voicemail sorry the Skype Answerinc Machine a free voicemail alternative to Skype s official Voicemail service won t work with the J yve SVMTO prefix try a link like this one SGripe rc http 7 plugin ve com 167 Plug ny Se gt Serie lt a href SVMTO jyvetestl gt lt img src http jyvetools jyve com sendvoicemail gif border 0 onclick setENDown gt lt a gt If you don t like the premade graphics that J yve provides in the lt img gt tags you can use your own or substitute text instead downloaded from lib ommolketab ir downloaded from lib ommotketab ir 3 7 4 Trigger Conference Calls from the Web Skype also includes a function that lets you create a link that starts a conference call with theSkype users of your choosing This might be useful for somebody who hosts a regular conference call with the same people every weeklike an editor and her authors By keeping a central link on a web page that all the attendees can access the weekly conference can still go on even if the regular host isn t available to set it up A fill in member need on
376. port spanning With port spanning you can program an Ethernet switch to let you snoop traffic on ports other than the one where your PC is connected Check out Cisco OS in a Nutshell O Reilly for an introduction to port spanning To demonstrate SIP packet observation with Ethereal we ll set up a filter that allows us to capture SIP registration signals in two scenarios one for a successful SIP registration and another for a failec SIP registration As in the other projects the SIP server s IP address is 10 1 1 10 In this instance Asterisk is used as the SIP server downloaded from lib ommolketab ir downloaded from lib ommotketab ir X Lite offers excellent diagnostic logging too Some of the packets you observe with Ethereal in this project will correlate with entries in the X Lite diagnostic log which you can view by selecting Diagnostics from the right click menu in X Lite s Ul 6 11 1 Configure the SIP Softphone If you re setting up X Lite for the first time you ll need to click the Configuration button right of the center next to the Clear button see Figure 6 2 Once you click this button you ll see the Configuration menu In the Configuration menu double click Menu and then select System Settings SIP Proxy default This will take you to the SIP client configuration as shown in Figure 6 3 Here you can configure the softphone to register using a number and or password to match what you ve established
377. r downloaded from lib ommotketab ir Hack 56 Link Several PBXs over the Internet 56 Since wide area networking is the cornerstone of I P networking VolP can be extended outside the local area network Six PBXs on six continents all managed by one person No problem One of the beauties of VoIP is the last two lettersIP IP stands for Internet Protocol and IP is the protocol that makes the Internet and private wide area networks WANs possible It provides the fundamental addressing and routing scheme that keeps data traffic flowing around the globe Just as people have been setting up VPNs to link remote offices LANs over the Internet you can now use the Internet to link several remote PBXs to create one large interconnected voice network This will enable extension to extension calling from one office to another over the Internet all at no cost per call Asterisk is the perfect solution for this purpose Keeping VoIP secure as it travels the globe is no simple matter You can use Internet based VPNs but they can degrade quality A more expensive and more reliable alternative to secure global VolP trunks might be a managed VPN service or a frame relay service will assume that you already have Asterisk in use as the PBX at all of your offices If this is not the case you might want to look into setting up an Asterisk gateway machinea server that provides VoIF enablement for a legacy PBX see Build a Four Line Phone Ser
378. r a telephone call and consequently make your AIFF recording smaller There s no point in saving 16 bit recordings if the audio coming from the recorded call is only 8 bits deep and there s no point in creating a stereo recording of a phone call Once you ve made the changes click OK to dismiss the Sound Settings window and close the Preferences window too if you wish Next fire up your X Lite softphone this will also work with Skype and iChat and make Sure it is registered to send and receive calls with your VoIP provider as described in Use a Softphone with a VoIP TSP Hack 4 Then call somebody or have somebody call you As the call begins click the red circle record button in WireTap s floating window shown inFigure 2 17 This will start an AIFF recording of the current audio output including the audio coming from the softphone When the call concludes click the square stop button Figure 2 15 WireTap s Preferences window downloaded from lib ommolketab ir downloaded from lib ommoltketab ir o x a U a oO Preferences File saving settings Save files to Desktop F Change 1 W Application name Prefix Coo sample iTunes_recording000 aiff Recording file settings ww Open files after recording them Open files in QuickTime Player hea sound compression settings Compressor IMA 4 1 Settings Settings 44 100 kHz stereo 16 bit wv WireTap window always visible
379. r accurately for them With CDRs in Excel Crystal Reports or even a homegrown Perl program a savvy telephony administrator can do the following e Determine which channels are used the most and the least e Determine which endpoints are called most often e Calculate the percentage of outgoing calls that are out of your area code e Create a list of calls broken down by endpoint e Create an invoice for a paying subscriber to the softPBX Asterisk s CDRs can also be stored in PostgreSQL MySQL and even syslog depending upon the modules you compile and install 4 8 1 Creating a Call Report downloaded from lib ommolketab ir downloaded from lib ommotketab ir One of Excel s coolest features is the Pivot Table Report actually used Excel for years without touching this menu option passing over it dozens of times until one day had to build a sales report for an application I d been developing had a choice between coding the report myself building it in a tool like Access or performing the analysis in Excel The only problem with that last option was tha didn t know how to do the analysis in Excell knew only that it could be done So turned to the Pivot Table Report or should say lahempivoted to it and built a sales summar in five minutes which to this day is still in use at the office where built it Needless to say I ve sworn by the Pivot Table Report function ever since never knew what was missing ou
380. r it At less than 500 the mini is great for the cost conscious and for those who don t trust the likes of Windows in a real time application like IP telephony a Mac provide a secure friendly alternative Of course there are comparably equipped small PCs but none with the tiny two inches tall and six by six inches square form factor of the Mac mini So if you need a Space conscious cheap VoIP server you need look no further than the mini Would you rather use an Xserve with this hack Great idea The Xserve has faster processors and a RAID hard drive array This means high performance PBX action It also has an extra Ethernet interface and a swappable power Supply making it better in mission critical situations than the mini But since the Mac mini has no card slots and since multichannel PCI telephony drivers are not available for OS X attaching analog phones and phone lines to a Mac mini isn t the same as in a traditional PCl equipped server chassis where you can snap foreign exchange office foreign exchange station FXO FXS cards into place to connect phone lines This is where VoIP comes in handy Just because the Mac mini can t connect directly to analog lines and phones doesn t mean yol can t use them with it You can connect analog lines and phones to a Mac mini by way of an analog telephone adapter ATA or an FXO gateway device An ATA will connect an analog phone to an Ethernet network and an FXO gateway will atta
381. r meanings Set this to the downstream speed of your connection in kilobits Use a speed test like the one available at http toast net to get an accurate idea of your connection s actual speed for instance the overhead of PPPoE accounts for an approximate 10 13 drop in speed from what is advertised by many consumer DSL packages Test and test often Also you will want to set this number to about 85 of your actual test speed See the sidebar VolP QoS for an explanation DOWNLINK 5500 downloaded from lib ommolketab ir downloaded from lib ommoltketab ir VoIP QoS Most broadband service providers configure their networks for bulk traffic soeed They know that to most customers speed is measured by how many KB s their web browser displays when downloading a large file However this is not the whole story With VoIP a measurement called latency is far more important The best possible way that have ever heard to describe the concept of latency is the Concorde R I P versus Boeing 777 analogy The British Airways Concorde can get 92 people from New York to London in about 3 5 hours The Boeing 777 can get 440 people from New York to London in about 6 5 hours Which is faster If you had to transfer a large number of people using only one plane the 777 would be faster even though it travels at half the air soeed of the Concorde If you had to transfer a small number of people very quickly the Concorde would be fast
382. ractice even if you don t use Skype There are three ways to do this Either restrict the MAC hardware addresses allowed to connect to your base station or use a Wireless Encryption Protocol WEP shared key to keep your neighbor s hands off your wireless LAN The third way is to use both techniques This is the most secure methoco of all To find out the MAC address on your Pocket PC power it up and tap Start Then tap Settings and then the System tab on the bottom of the screen Next tap the WLAN icon it might say iPAQ WLAN depending on your brand of Pocket PC Then tap the Status button The MAC address will be listed near the bottom of the screen Add it to your wireless base station s access list downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 35 Degunk International SkypeOut Calls P35 With SkypeOut you can easily make international long distance calls but keeping track of international country codes isn t so easy Neither is keeping track of calling rates which vary from one country to the next The SkypeOut service is one of several paid services Skypeln Voicemail etc that Skype provides for its users The service lets you place calls to regular phone numbers on the PSTNthat s the telephony network still used by everybody who isn t as cool as you are SkypeOut minutes are purchased in advance and at least forinternational calls their rate is quite competitive You can SkypeOu
383. ral Asterisk PBXs over the Internet 2nd PC expansion PCI cards PC softphones PCI interface cards PCMA codec PCMU codec PDF file peer to peer P2P network peers configuring IAXy for use as SIP Perl Grandstream compatible ringtone script script investigating how SIP works script monitoring SIP hosts on VolP system 2nd 3rd 4th script recording latency and jitter data scripts to monitor Asterisk PBX 2nd Phlink 2nd 3rd 4th 5th 6th 7th 8th 9th caller IDs in AppleScripts controlling iTunes custom greetings greeting callers differently each day running when you are logged off phone numbers searching for your own phone number phone service phone service providers phone company lines phones PhoneTray Dialup application downloaded from lib ommolketab ir downloaded from lib eommoltketab ir PhoneTray Free application PHP modifying configuration files for AMP music on hold ping utility latency measure measuring latency pitch effect Pivot Table Report Excel 2nd 3rd plain old telephone service POTS line for 911 calls Plug and Play PnP Pocket PC Pocket PC version of Skype PocketSkype podcasting integrating Skype into 2nd 3rd 4th Mac OS X tools for Polycom IP500 phone port default for SIP port numbers used by VoIP applications VoIP Port SPAN port spanning ports SipPort setting used for VoIP protocols post processing tools for podcasts PostgreSQL database POTS Plain Old Telephone Service line conn
384. rded in Cain amp Abel using ARP poisoning to intercept it downloaded from lib ommolketab ir downloaded from lib ommotketab ir File View C onfiqure T Tools Help BY Protected Storage J Network i sniffer g LSA Secrets 13 08 2005 18 21 07 L3f08 2005 18 24 56 13 08 2005 18 26 54 13fOS 2005 18 29 16 13 08 2005 18 29 35 2005 20 41 39 13 08 2005 13 08 2005 13 08 2005 13 08 2005 13 08 2005 13 08 2005 2 10 1 1 10 15770 PCM 10 1 1 204 6000 PCM 10 1 1 204 8000 PCM 62 176 59 24 4024 67 16 1395 226 4311 10 1 1 204 1165 4 f Darr 4 Routing F Passwords vor Lost packets 0 s 10 1 1 104 20204 PEMU 8hhz Mono 10 1 1 10 10104 POMU 8khz Mono 10 1 1 10 16314 POMU 8Khz Mono 10 1 1 204 1165 10 1 1 204 1165 10 1 1 203 35548 RTP 2008081 3222211296 RIP 2005031 3222543421 RTP 20080681 32227 134271 Codec not supported Codec not supported Pretty amazing eh One more thing can t emphasize enough how unethical this hack would be if performed in an environment where you aren t authorized to take these actions strongly urge you to be responsible with this type of hack and keep yourself out of trouble while you learn about VolP security Unless you have administrative authorization and the express right to monitor use
385. re just plain curious read on This line installs the root hierarchical token bucket HTB queue and points default traffic to the 30 class tc qdisc add dev SDEV root handle 1 htb default 30 This line defines the queue used for VoIP As say in thescript this is the Crown Prince of Bandwidth Nothing has higher priority than VoIP inAstShape tc class add dev SDEV parent 1 1 classid 1 10 htb rate S UPLINK kbit burst 6k prio 1 The same for the interactive class tc class add dev SDEV parent 1 1 classid 1 20 htb rate P UPLINK kare burst 6k prio 72 The default class tc class add dev SDEV parent 1 1 classid 1 30 htb rate S 9 SUPLINK 10 kbit burst 6k prio 3 The bulk class tc class add dev SDEV parent 1 1 classid 1 40 htb rate S 8 SUPLINK 10 kbit burst 6k prio 4 Now that we have our queues defined we need to assign traffic to them Any IP packets with Tos 0x18 belong in the VolP class te filter add dev SDEV parent 1 0 protocol ip prio LO W324 match 15 tos Uxle Oxi lovia Leo downloaded from lib ommolketab ir downloaded from lib ommotketab ir Any IP packets with Tos 0x10 minimum delay belong in the interactive class te filter add dev SDEV parent 1 0 protocol ip prio 20 32 Matoh ip bos UZIO Oxit tlowid Lt20 By default most SSH client server programs will set the IPtos field to 0x10 How convenient Add DNS to interactive too to filter ada day DEV parent 170 protocol ip prio i
386. rom these devices you ll need something a little more James Bondish like atransducer pickup This is a microphone that you stick to the outside of your telephone handset on the back of the receiver that is sensitive enough to record the audio inside the handset Since even digital and IP telephones use purely analog handsets a transducer pickup can record them all Some pickups such as Radio Shack s model 44 533 includ a built in suction cup that adheres easily to the handset Like a recorder switch these pickups provide a 1 8 inch mini plug that you can mate with your sound card s audio line in jack to make digital recordings Recording phone calls can get you in trouble unless all parties on the call are aware that the call is being recorded Check your local laws before recording any phone calls 5 2 1 See Also e Record VolP Calls on Your Windows PC Hack 13 e Record an Audio Chat on Your Mac Hack 23 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 60 Make IP to IP Phone Calls with a Grandstream BudgeTone P60 With minimal effort Grandstream s BudgeTone series of I P phones can make and receive calls on your networkeven without a PBX server In most enterprise VolP setups you have a PBX that connects all of the phones on the network The PBX acts as a centralized signaling authority and access control server for all of the telephone users But some IP phones don t need a PBX at
387. rom which graphs like the one in Figure 6 8can be produced The illustration technique used in Figure 6 8is a standard way of representing data conversations In Chapter 7 of O Reilly s Switching to VoIP the details of SIP signaling are covered with a half dozen examples that are graphed in this fashion A standard convention of a SIP call flow graph is the direction arrow In this example there are many of them one for each SIP message sent When you click on one of the arrows it automatically drills down to the specific packet regarding that part of th call flow in the trace window This is fantastic for literally stepping through the signaling steps of call setup and tear down downloaded from lib ommolketab ir Figure 6 8 A SIP call graphed by Ethereal downloaded from lib eommolketab ir Graph Analysis Se ae z l eee F EF T rae kjk papp T o a a A h d o n em rk a S F rN it f s 4 i Ck eee aT I rI ae ir ra a bon S T 7 z ta att i i a ee i eg it i ar e e T TER mafi Hli f uas Bt al i i ii Y i PH 4 i ae 1 on pl 54 041 55 367 3 Ja i r B d f f m a Bi y J 1 T Fo wE 1 To tet e a ee fo N N i a a T E E Ti mo 8 E n i ts b mik dele oh E Tat 7 L is 55 404 62 480 69 592 69 901 69 901 69 935 82 350 82 391 Joel Sisko downloaded from lib ommotketab ir 192 168 1 100 147 135 8 128 147 135 8 247 Comment SIP From sip 646 sip broadyoi
388. rovides the following script to pause iTunes music playback when the phone rings This script is really cool As Forrest Gump would say that s about all have to say about that Save it as ring scpt or modify your existing ring and put it in the Phlink Items directory on incoming_call given call the_call set my_paused to false tell application Finder if exists of every application process whose creator type is hook is true then tell application iTunes if player state is playing then set my_paused to true pause else set my_paused to false end if end tell end if end tell tell application Ovolab Phlink tell the call make new bag with properties name pauseitunes waspaused my_paused end tell end tell return false end incoming_call Now to resume iTunes automatically when the phone call is hung up add this to yourhangup script on do action given call the_call tell application Ovolab Phlink tell the call try if waspaused of bag pauseitunes is true then tell application Finder if exists of every application process whose downloaded from lib ommolketab ir downloaded from lib ommotketab ir creator type is hook is true then tell application iTunes to play end if end tell end if and Ley end tell end tell end do_action For more fantastic Phlink scripting magic be sure to visit the message board at http www ovolab com and for some more cool ideas for iTunes Phl
389. rred vernacular This is especially problematic with consumer access technologies such ascable modem or DSL In a typical residential or small office setup you will have one relatively high speed link to the Internet and that link is responsible for carrying email web surfing and even the occasional big download of CD ROM image or something Now try to put voice traffic on this link Humans are very sensitive to delay when listening to speech If one web site on your computer loads 250 ms slower than another web site you re not really going to notice However if there is a 250ms delay in a conversation you ll perceive that as a very annoying delay and it will make ordinary conversation difficult With all of this traffic on one link how can make sure that someone downloading a song from iTunes does not cause the audio on my VolP call to suffer It s easy with a technology referred to as Quality of Service QoS QoS is a general term applied to a family of technologies that essentially manipulate the first in first out FIFO queues on devices Remember that PC from before or that router or ATA Normally all of the IP traffic from that device will be placed into a FIFO queue for delivery to the remote endpoint With QoS we can manipulate that queue and pass judgment on packets matching certain attributes that move them to the front of the line regardless of what time they got in because they are more important to us This is what yo
390. rrov until you reach IP Address then press Menu again Now you can key in the IP address substituting stars for periods Press Menu to accept the address and then arrow down to the Subnet Mask and enter the appropriate value for your LAN Repeat this process for the Default GW option enterin the right value for your LAN s default gateway probably the address of the nearest router Next use the arrow keys to select DNS Server 1 press Menu and enter the IP address of the nearest DNS server Then reboot the phone by powering it off and on Try pinging the phone s IP address from a nearby PC to see if it s communicating with the network 5 11 2 Connect the Uniden to TFTP To alter the phone s automatic configuration mode i e TFTP based configuration you need to unlock the configuration menu To do so press the phone s Menu key and press the down arrow unti you reach Unlock Config Press Menu again You ll be prompted for a password which you can enter using the number pad on most UIP200 firmware versions the password is 2002 Press the Menu key to confirm the entered password downloaded from lib ommolketab ir downloaded from lib ommotketab ir Next press the up arrow until you reach Network Settings Press Menu then press the up arrow unti you reach TFTP IP Address Press Menu enter the address of your TFTP server and then press Menu again to confirm it Now press the Cancel key to go back to the main menu Use t
391. rs calls on the network where you re working you should keep ARP poisoning confined to the test lab downloaded from lib ommolketab ir downloaded from lib eommolketab ir Chapter 7 Hard Core Voice Section 7 1 Hacks 88100 Introduction Hack 88 Build a Killer Telephony Server Hack 89 Build an H 323 Gatekeeper Using OpenH323 Hack 90 Turn Your Linux Box into a Fax Machine Hack 91 Build an Inbound Fax to Email Gateway Hack 92 Teach Your Asterisk Box to Speak Hack 93 Build a Mac PBX Hack 94 Monitor Asterisk from Your Perl Scripts Hack 95 Build a SoftPBX with No Hard Drive Hack 96 Build a Standalone Voicemail Server in Less Than a Half Hour Hack 97 Automate Your Voicemail Greeting Hack 98 Connect Asterisk to the Skype Network Hack 99 Forward Your Home Phone Calls to Skype Hack 100 Get Started with sipx downloaded from lib eommolketab ir downloaded from lib ommotketab ir 7 1 Hacks 88100 Introduction If you ve been involved in Linux VoIP hacking for very long hey you made it toChapter 7 so you ve been around a while you re probably already quite familiar withAsterisk the open source PBX and the predominant open source VoIP platform Aside from Asterisk tons of open source voice networking projects are out there including OpenH323 GnuGK sipX SIP Express Router OhPhone SaRP and GnoPhone Try Googling some of these You ll find there s enough open source VolP stuff to ke
392. rs to know the extension at any time Break up the menu into chunks that you can handle downloaded from lib ommolketab ir downloaded from lib ommoltketab ir lt CiscolPPhoneMenu gt lt Title gt EXT 12 4 lt Title gt lt s lt Menultemn lt Name gt PIM lt Name gt lt URL gt http domain cisco services pim xml lt URL gt lt Menultem gt lt Menultem gt lt Name gt Work lt Name gt lt URL gt http domain cisco services work xml lt URL gt lt Menultem gt sMenultem gt lt Name gt Play lt Name gt lt URL gt http domain cisco services play xml lt URL gt lt Menultem gt lt CiscolPPhoneMenu gt So you now have three sections to add things into Work on PIM first and add a list of favorite restaurants hey you ll want to reward yourself with a pizza when this is through Go to the pim xml file on the web server Add some local and not so local places and then allow the menu to dial the number for you To keep my favorite places safe l Il suggest fictional restaurants and phone numbers Actually if you re in the Detroit area nothing beats National Coney Island lt CiscolPPhoneMenu gt lt Title gt EXT 1234 PIM lt Title gt lt Menultem gt lt Name gt Restaurants lt Name gt lt URL gt http domain cisco services pim xml lt URL gt lt Menultem gt lt CiscoIlPPhoneMenu gt lt CiscolPPhonebirectory lt Title gt EXT 1234 PIM Restaurants lt Title gt lt DirectoryEntry
393. rsion of X PRO but for the purposes of this hack the feature disparity between versions makes no difference Installation is straightforward On Windows run the installer package and on the Mac drag the X Lite program icon into your Applications folder Once installed launch X Lite step through its Audio Tuning Wizard and look at its user interface By some strange coincidence it resembles a nice looking business phone Imagine that Vonage Users Beware If you re a Vonage subscriber you can download the Vonage branded version ofX Lite s commercial counterpart called X PRO from your account page on Vonage s web site If you re using Vonage you re limited to using Vonage s version of X PRO and you won t have nearly the flexibility that the non Vonage version of the software provides Indeed once you have the Vonage version running the only administrative customization you can do is to change your username and password You can t really get at the softphone s SIP guts because Vonage s version keeps all that stuff off limits to the end user Those seeking a deep hacking experience should probably consider BroadVoice instead Unlike Vonage BroadVoice openly supports noncrippled softophones such as X Lite 1 6 2 1 Setting up the basics After you ve gotten through X Lite s Audio Tuning Wizard you re ready to dive into theSIP configuration settings These define how the softphone will authenticate and interact with your TSP s SIP serv
394. s and the green colored FXS modules connect phones So if you want downloaded from lib ommolketab ir downloaded from lib ommotketab ir three analog phones to share a single analog phone line you would use three FXS modules and one FXO module The Linux driver framework that allows Asterisk to use the TDM4OOP card is called Zaptel and if you worked your way through Turn Your Linux Box into a PBX Hack 41 you ve already got the Zaptel drivers installed Since these drivers don t yet work with Mac OS X you ll be able to use the TDM4OOP card only on i386 Linux l Il cover enabling the drivers in a moment If you re using certain servers the TDM400P might not be compatible with certain Dell PowerEdge servers Also some Intel equipment has known issues with the TDM400P card So check Digium s web site http www digium com to be sure you could save yourself some time and aggravation VoiceTronix makes alternative cards that you might want to consider too But first you need to get the card installed This is pretty straightforward If you ve got a spare PCI Slot you re ready to snap the card into place There are four numbered modules on the card which correspond to the four numbered eight wire jacks on the case plate of the card Before Inserting the card screwing down the back brace and replacing the PC s cover you might want to note which jacks are for connecting phones and which are for connecting phone lines It
395. s being used to send a text message This is ideal for enabling people to get in touch with you even if they don t have Skype because they can enter their call back number and it will be sent to you in the text message This could be a great customer service tool for your company web site or just a nice novelty item for your personal blog Figure 3 6shows what an incoming J yve text message looks like Figure 3 5 AJ yve Q Card downloaded from lib ommolketab ir downloaded from lib ommolketab ir Card tedwallingford B y z To talk with me please enter your phone number a Phone 440 324 1778 Ext tedwallingford 5 OR Tel Skype Name Local Time 11 01 PM Call Me Right Now HH gt Name kelly Wallingford Ted need to know what showtime you want before order the movie tickets Call me ASAP Subject Kelly From Jyve com Me Figure 3 6 AJ yve text message sent from the Q Card downloaded from lib ommolketab ir downloaded from lib eommolketab ir 4 sye com sretteerervers JYVE COM 278 00 Add Topic Leave Profile Call Send File JYVE COM eh a ONE WAY INSTANT MESSAGE fhe hate Somebody wants to talk with you Skype Name CALLTO Name Phone CALLTO 440 324 1778 Ext Please return call Right Now Subject Ted need to know what showtime you want before order the movie tickets Call me ASAP Kelly i downlo
396. s best withYahoo Widgets a widget framework that seamlessly integrates with Mac OS X and Windowsspecifically Mac OS X 10 2 and higher or Windows 2000 and XP Literally thousands of different widgets are available that run on both Mac and Windowseverything from weather reports and stock tickers to cute little iPod remote controls and telephony related goodies One such goody is the must have vonageGauge widget by Martin Koistinen which gives you a one glance update of your remaining Vonage minutes as well as a count of voicemails waiting to be listened to Figure 2 1 2 3 1 Installing Yahoo Widgets It s quite worth your while to install Yahoo Widgets even if you can t benefit fromvonageGauge Throughout this book reference a number of other cool Yahoo Widgets that will aide you in your telephony travails The place to start ishttp widgets yahoo com Here you can download a version of the Y Widgets system for either platform To install on Windows just run the installer that you downloaded To install on Mac OS X drag the Konfabulator icon which might eventually become the Y Widgets icons from the downloaded DMG volume folder into your system s Applications folder Then launch it by double clicking it Figure 2 1 The vonageGauge widget in action downloaded from lib ommolketab ir downloaded from lib ommoltketab ir You ll be stepped through a wizard that helps you decide where you want to store downloaded widgets
397. s from the Mayor of Gotham Of course he does He s that cool his car is OK too If you ve worked your way through Tweak Your Sipura ATA Hack 62 you know Sipura Technology makes some very powerful and flexible ATAs So powerful infact that you can use them to set up a point to point hot line with no SIP proxies or registrars A bat phone or automatic ring through in the telco world is best known from the popularBatman television series Batman would have such a burning desire to speak with the commissioner that he didn t even have time to dial The simple act of picking up the phone automatically connected him to the designated remote station Here is what you will need to get this going with your two Sipuras Two Sipura ATAS As of this writing the 841 1000 1001 2100 2000 and 3000 were widely available but Sipura has just been acquired by Cisco so these model numbers could change Static IP addresses or dynamic DNS Each Sipura will need to Know where the other is On a simple LAN this is incredibly easy J ust assign static P addresses and move on Over the Internet behind NAT and or firewalls this task can get complicated While it s too much to cover here you will want to look into port forwarding and dynamic DNS 5 6 1 First Things First Take out one of your shiny new Sipuras This will be called ATA1 Connect the phone to line 1 if you have more than one line and Ethernet cables Then connect the p
398. s own line in the section labeled Application based QoS and click on the High Priority radio button for each After restarting the router all RTP traffic sent from the router will be handled before any other traffic This technique is especially good if your LAN has multiple VolP devices that send media streams through the router 1 7 1 2 Prioritize all the traffic from your VoIP ATA If you have only a single VolP device tosupport like a TSP provided ATA it might be best if you tell the router to prioritize traffic by Ethernet port instead of by application as in the preceding paragraph Specifically you want your router to prioritize traffic that comes from the Ethernet port where your ATA is connected To do so use the High Priority and Low Priority radio buttons for the numbered Ethernet ports Set them up however you want and reset the router 1 7 1 3 Prioritize all the traffic from an attached Ethernet switch By setting the priority of a particular Ethernet port you are telling the router to prioritize anything from the device connected on this port even if this device is another switch So an easy way to give priority to all your dedicated VoIP devices like IP phones and ATAs is to connect them all to the Same switch and then connect that switch to a high priority Ethernet port on the router 1 7 2 Prioritize Traffic on a Standalone Switch Many workgroup Ethernet switches offer QoS features that used to be found only on ad
399. s retained It s a snap to import this into Excel or your favorite spreadsheet for analysis You can download the file from your server using FTP or you can run the following command to email it to you keep in mind that large logfiles might not work well with this trick cat var log asterisk cdr csv Master csv mail me mydomain com Of course replace me mydomain com with your email address If your Linux server has sendmail or a similar Simple Mail Transfer Protocol SMTP agent running most do the contents of the file will be emailed to you You can then copy and paste them into Excel as shown in Figure 4 4 Place the cursor on column A row 1 before pasting Figure 4 4 A portion of the Asterisk ASCII call detail record CDR logfile copied and pasted into the Macintosh version of Excel downloaded from lib ommolketab ir downloaded from lib ommotketab ir ail e 0O Ir Workbook1 A E C f 104 912162711111 default 104 SIP 104 19ec Zapfi 1 Dial Zap 1 12162762106 2 104 912162762111 dafault 104 SIP 104 a053 Zap 1 1 Dial Zap 1 1216276210g 3 7348055200 s house 17348055000 lt 7348055200 gt Zap 1 1 SIP 104 4169 Dig 4 4403282441 s house T Wallingford lt 440 325 4141 gt Zap 1 1 STP 104 Ob 60 103 104 FE 103 SIP 103 3603 S1P i04 7c64 Dial SIP 104 40 r 2004 08 J 103 104 detauk 103 SIP 103 3123 SI
400. s up to the firewall to forward them to the appropriate client on the private network So there are two steps to hacking out a functional STUN server First set up and configure the server Second configure your NAT firewall so that it forwards inbound VolP traffic to your SIP phone To build a STUN server you ll need a Unix box Linux BSD Mac OS X etc that has a non NATted connection to the Interneti e it has a public IP address There is a Win32 version of STUN too but compiling on Unix is a lot less trouble You can obtain the STUN server software from http sourceforge net projects stun Download and unpack it into your Unix machine s usr src directory then compile and install it like this wget http kent dl sourceforge net sourceforge stun stund_0 94 Oct29 tgz cd usr sre tar xvzf stund_0 94 Oct29 tgz cd stund make cp server usr sbin stund usr sbin stund amp H HH OH That last command launches the STUN server Make a note of your STUN server s internal IP address on a small network this is probably your default gateway as you ll need to provide it to your SIP phone as the STUN server address 6 6 2 But What About RTP So STUN will allow your SIP client to know what to put in the SIP Via header but remember that the VoIP call also relies upon RTP an entirely different protocol for transporting the voice packets Thes packets must flow on two separate UDP sockets one from you to the person y
401. s you will need to configure Display Name Your name or as much of it as you can fit User Name The SIP username provided by your TSP This ts likely to be your phone number including the area code Authorization User This is normally the same as the SIP username provided by your TSP though some TSPs might issue a distinct authorization username In circumstances where multiple phones with their ow phone numbers are authorized for the same subscriber the two usernames might vary Password The password you and your TSP established when you set up your VoIP account downloaded from lib ommotketab ir downloaded from lib ommotketab ir Domain Realm This tends to be the domain name associated with yourSIP user URI which is similar to an email address For 4403281414 sip broadvoice com your domain realm would be Sip broadvoice com Your TSP will issue you an appropriate realm name if it Supports the use ot softphones SIP Proxy This is the address of the proxy server that will handle all your VolP registration activitythings like authentication and notifying the server that you re available to receive calls Unlike a SIP URI which always contains sip in the domain the SIP proxy address can be any valid host name Again your TSP will provide you the address to use when you sign up Outbound Proxy This address is used to handle SIP requests that are bound for other SIP domains Since most VolP TSPs don t support ca
402. se Mpg123 you can just isSuenake mpg123 from your Asterisk source directory To get MAD start at http mad sourceforge net for a list of mirror sites to download the MAD distribution You ll need three pieces thel D3 library the MAD library and the madplay application Each is in a separate archive that you ll need to download unpack and install as follows ed root mkdir mad cd mad wget http kent dl sourceforge net sourceforge mad madplay 0 15 2b tar gz wget http internap dl sourceforge net sourceforge mad libmad 0 15 1b tar gz wget http peterhost dl sourceforge net sourceforge mad libid3tag 0 15 ib tar gz tar xvzf madplay 0 15 2b tar gz tar xvzf libmad 0 15 1b tar gz tar xvzf 1libid3tag 0 15 1b tar gz cd libid3 configure make make install H i cd libmad i JE configure make make install cd madplay configure downloaded from lib ommolketab ir downloaded from lib ommotketab ir make make install Now type madplay and press Enter If you get a failed to load message about one of the library files as shown here the installation routine might have put the libraries in the wrong location madplay error while loading shared libraries libid3tag so 0 cannot open shared object file No such file or directory If this is the case try moving them On my system had to move them to usr lib mv usr local 1lib libmad so usr 1
403. sed party line system that really works only with Windows Skype is great for two party direct voice calls multiparty conference calls with up to fifty participants and text chatting Even cooler is the fact that Skype s user directory has advanced search functions so you can find somebody of a certain age gender name or country But Skype doesn t have the social networking depth of the Yahoo system so if you re looking for a voice chat room where people can freely come and go from the conversation Skype is inferior to Yahoo Chat And unlike the Yahoo and iChat instant messaging tools there s no built in Support for videoconferencing For that you ll need todownload and install one of the video add ons Hack 39 3 1 3 What About Security As an ethically concerned VoIP hacker you re probably wondering if this P2P network is secure The answer is that Skype is and isn t secure The fact that Skype encrypts all call signaling and media transmission does point to security but the fact that your calls are routed using anonymous PCs that are members of the Skype P2P network points to lack of security As computers become faster and faster it might someday be trivial to crack Skype s encryption and when that day comes the P2P network itself will be a security flaw For the time being Skype is fortunately quite secure so dig in One hundred fifty million downloaders can t be wrong If you need more information than what is provided in
404. sion are like RTP So if you are using SIP you need to permit inbound SIP signaling on UDP ports 5060and 5061 Consider the following iptables policy commands iptables P INPUT J DROP iptables A INPUT UDP dport 5060 5061 7 ACCEPT iptables A INPUT p UDP dport 5036 4 ACCEPT iptables A INPUT p UDP dport 5004 jJ ACCEPT iptables P OUTPUT J ACCEPT downloaded from lib ommolketab ir downloaded from lib ommotketab ir This set of iptables commands manipulates the kernel s firewall such that theserver can accept only RTP Inter Asterisk Exchange IAX and SIP traffic all outbound traffic OUTPUT chain is permitted This policy is based only on UDP port numbers If incoming traffic isn t on ports 5060 5061 5036 or 5004 it is dropped A truly hardened server would restrict outbound traffic too For more information about securing VolP refer to Switching to VolP O Reilly downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 89 Build an H 323 Gatekeeper Using OpenH323 HACK 89 H 323 is a VolP signaling protocol that predates SIP by five or six years but its use in commercial telephony and desktop conferencing apps NetMeeting for instance is widespread OpenH323 is an open source implementation of the H 323 signaling protocol suite managed by Quicknet Technologies the same company that makes the Internet Phone Jack line of analog interface cards OpenH323 is distributed
405. sk Using the TR16A demonstrated how you could connect to an Asterisk phone system using an FXS port and an IP phone to operate the X10 controller X10 has another controller known as theCM15A The CM15A connects to your local PC using a USB cable and it boasts a free SDK that you can use to write custom scripts Using the Asterisk system s IVR function you can program Asterisk to run a Script Ir turn that script can control the X10 modules allowing for a pure software based solutionthat is no FXS interfacing necessary 5 13 1 Hacking the Hack To have Asterisk dial the PIN code for you automatically add it to the extension definition that dials the controller assuming your PIN is 1212 so that it dials your pin automatically after connecting exten gt 100 1 Dial Zap 1 30 D 1212 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Joel Sisko downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 71 Use a Rotary Dial Phone with VoIP Meri Or How Grandma Mabel Learned to Love Voice over IP What do a 19th century Missouri undertaker and SIP have in common The common thread is their ability to connect two parties without a third party intervening That is they both serve an intermediary role between caller and callee signaling the call and providing a pathway for its sound signals In 1891 a Missouri undertaker Almon B Strowger was granted a patent for a
406. so that nobody breaks the rules Kristian Kielhofner downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 79 Build a 100 PSTN Gateway in 10 Minutes or Less s7 The Sipura SPA 3000 is a marvel of engineering For less than a hundred bucks you can interface your phone line with your VolP network What s cooler than that When your phone line is connected to your plain old analog phone it works and works well But whe it s connected to your VoIP network it takes on a whole new personality Suddenly you can do all kinds of cool stuff This hack will show you how to connect aPSTN phone line to an Asterisk based VolP network using the Sipura SPA 3000 ATA This device is like other ATAs in that it has one FXS port However the SPA 3000 has a trick up its sleeve a single FXO port as well Not only does it have the hardware bu Sipura s firmware is actually quite flexible allowing you to do all kinds of things to impress your friends and make life easier hopefully For this hack you ll need an Asterisk machine nearby Hack 4 am going to demonstrate this hack using Asterisk and the SPA 3000 But because the SPA 3000 Speaks SIP you can just as easily use it in conjunction with most other SIP compatible devices out there In fact the Clipcomm CG 200 gateway Hack 43 would make a fine substitute The Asterisk server has asip conf file that allows calls to be placed into the default context from
407. softPBX is configured this way not likely outboundProxy1 allows you to specify that you want to use a different SIP server for nonlocal calls In most cases OutboundProxy1 will be the same as Proxy Server Registrarl MT edu cde cae cle RegistrarlPort 10 1 1 10 Like a SIP Proxy which routes calls a SIP Registrar also handles connections from SIP phones but for a different reason A SIP Registrar keeps tabs on SIP users and informs requesting callers as to where a particular SIP user can be found i e what IP address that user is registered from You can specify a different registrar But in most cases this will also be the same address as your SIP Proxy as proxies and registrars tend to be on the same server more often than not DnsServer_1 TOs te ee LO downloaded from lib ommolketab ir downloaded from lib ommotketab ir This setting lets you override whatever DNS server address you provided in the Quick Setup menuunless the phone is using DHCP in which case it will use whatever DNS server is recommended by the DHCP server when it acquires its address RegisterExpiresec 3600 RegisterRetrySec 90 These settings tell the phone how often to register with the SIP registrar and how often to retry failed registrations SiIpProrL 3060 SipPort tells the phone which UDP to use when listening for SIP messages such as incoming calls 5060 is the default and in most scenarios you want to leave it at 5060 G711MuTxPacketL
408. ss is kept private and is only needed For voicemail and account management Homepage http feww macvoincom Pry Desertion i el List my profile in the White Pages for public searches Don t forget to check the List my profile in the White Pages for public searches checkbox if you d like to hear from other Gizmo users Otherwise they won t be able to find you when searchingGizmo Project s central user database If you d like to search for some buddies to add to your contact list start by clicking the Search button in Gizmo s main window Its search function which is similar to but less elaborate than that of Skype shows you the city state and country of each user if they ve entered that information in their profile Gizmo also has a big selection of rather cool built inavatars buddy icons or you can select your own image file to use Placing a voice call with Gizmo is as easy as entering the Gizmo name of the person you want to talk to and clicking the round phone icon in the upper right of the main interface window If you don t yet have any buddies in your contact list a great place to start is theGizmo Project Party Line which you can call by typing partyline In place of a normal Gizmo name Calling the Party Line connects you to other folks in a voice chat room who might be able to help you start your social network with Gizmo If there s nobody in the Party Line chat room at the outset of your ca
409. stant message use thermtTo prefix and the following code Serivt sre htip plugin eco 1S pli eS lt script gt lt a href IMTO voiphacks Hello World gt lt img src http jyvetools jyve com IMTO gif border 0 onclick setENDown gt lt a gt 3 2 Track Visits to Your Site by Jyve Users You can create a trigger on a web page that sends text messages to you when the web page is visited without allowing the surfer to compose or add to them The visitor won t see an instant downloaded from lib ommolketab ir downloaded from lib ommotketab ir message window either The idea here is to send you a message when somebody hits a certain page on your site Using the following avaScript in one of your page body s onLoad events you can use a pop under window to send you a Skype instant message automatically whenever a visitor with the J yve plug in visits lt script language JavaScript gt E function popunder pu window open SENDIMTO voiphacks I visited your site default DuDLUr window focus pu close peen sJ sCrIpt gt lt body onLoad javascript popunder gt 3 7 3 Simplify Communication for Visitors to Your Site The best thing you can do if you want to keep in touch with somebody is get into their buddy list and vice versa If you d like to have a link that lets the surfer add you as a contact to hisSkype contacts list try a link like the following which
410. started run this command in theAsterisk source directory make samples This creates a basic sample set of Asterisk configuration files and places them in etc asterisk You downloaded from lib ommolketab ir downloaded from lib ommotketab ir might want to peruse these filesespecially extensions conf and sip conf where you ll likely be Spending a lot of time If you ve used an RPM package or some other precompiled Asterisk distribution or if you ve obtainec a Linux distribution with Asterisk already installed you can still obtain the source distribution files from Digium s CVS repository and issue only themake samples command This will give you the Sample configuration files without actually rebuilding Asterisk on your PC 4 2 1 Start and Stop the Asterisk Server The Asterisk program has two modes of operation server mode and client mode The server is the instance of Asterisk that stays running all the time handling calls recording voicemails greeting callers while users are away and so on The client is the instance of Asterisk that allows you to monitor and manipulate the server while it runs The mode the program uses depends on how Asterisk is invoked at the command prompt or within a shell script To launch Asterisk in server mode execute this command asterisk vvv amp The more v s the more verbose Asterisk s console output will be To connect Asterisk in client mode on the local machine already running
411. stomer to discourage customers from hosting high traffic services on their residential broadband connections So there s less available bandwidth to you for sending than for receiving The VolP media stream most likely to suffer as a result is the outbound stream the one carrying your voice to the person on the other end of the call As such it s appropriate to prioritize outbound traffic to overcome the limits many ISPs force on outbound bandwidth downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 6 Got 911 HACK 6 For a multitude of technical and political reasons Internet TSPs have been slow to make reliable Emergency 911 dispatch dialing available for their customers Here s how to know if you ve got it If you recently signed up for VolP telephone service the likelihood of you having911 service is low but some TSPs do offer it The fastest way to find out if your TSP offers it is to contact them and ask Vonage for instance supports 911 call routing to most public safety jurisdictions but you ve got to activate this feature first Here s a snippet from Vonage s end user agreement You acknowledge and understand that 911 dialing does not function unless you have successfully activated the 911dialing sic feature by following the instructions from the Dial 911 link on your dashboard and until such later date that such activation has been confirmec to you through a confirming email Y
412. stration attempts The password field is optional its use is policy dependent asgnugk accepts blank passwords Finally the E 164 number is the phone number to which the endpoint is registering and ultimately the phone number that will be used to route calls to this softohone Be sure to check the Use gatekeeper check box too When you close the Preferences window click the Start Phone button and then click the Console button You ll see whether the softphone s registration attempt with theH 323 gatekeeper was successful The console log of OhPhonexX shown inFigure 7 2 contains the details of the registration attempt Now if you register a second softphone from a second PC you can call back and forth between them using the gatekeeper as the E 164 alias translator This works the same way with H 323 hardphones Callers dial the E 164 digits and the gatekeeper provides the E 164 resolution that allows the software in the phone to do its H 225 H 245 and RTP signaling to facilitate the call Figure 7 1 OhPhonexX s Preferences window has all the options an H 323 downloaded from lib ommolketab ir downloaded from lib ommoltketab ir endpoint could possibly need to register with a gatekeeper OO Preferences Mi Use gatekeeper Dialing Prefix User Information User Alias ID Password E164 number 2112 Save Password to KeyChain Load Password from KeyChain Once a call
413. sue these commands configure prefix usr make make install cp app rxfax c usr src asterisk apps cp app txfax c usr src asterisk apps cp Makefile patch usr src asterisk apps cd usr src asterisk apps patch lt Makefile patch H HS HOH If you re worried about the security concerns associated with compiling as root you can use a nonroot account to compilespandsp These commands compile the spandsp package which provides a source code patch for Asterisk As such you ll need to recompile Asterisk now cd pathtoasterisksource asterisk make clean make install The next step to faxing with Asterisk is to enable fax detection on the Zaptel channel you want to use for faxing This doesn t stop the channel from being used for normal voice calls it just enables the channel to discern fax calls from normal calls To enable this function be sure that the channel s section in etc asterisk zapata conf has this entry downloaded from lib ommolketab ir downloaded from lib ommotketab ir faxdetect both The valid parameters for the faxdetect Option are incoming outgoing both and no By default fax detection is disabled 7 4 1 Receiving Faxes Now consider the following snippet from a dial plan ancoming Loca exten gt s l Answer exten gt s 2 Dial SIP 202 45 rm exten gt s 3 Voicemail 202 exten gt fax 1 SetVar TIFFILE var spool faxes thisfax tif exten gt fax 2 rxfax S TIF
414. sure that the ports defined in thescLASS1PoRTS variable match Class 1 for a in CLASS IFORTS do tc filter add dev SDEV parent 1 0 protocol ip prio 11 u32 match ip dport Sa Oxffff flowid 1 10 tc filter add dev SDEV parent 1 0 protocol ip prio 11 G52 match ip sport Sa Oxtrrt tlowid 1 10 done downloaded from lib ommolketab ir downloaded from lib ommotketab ir Likewise the simple loop in the following snippet makes sure that the ports defined in the SCLASS2PORTS variable match Class 2 for a in S CLASS2PORTS do tc filter add dev SDEV parent 1 0 protocol ip prio 24 u32 match ip dport Sa Oxffff flowid 1 20 tc filter add dev SDEV parent 1 0 protocol ip prio 24 u32 matoh ip sport Ga URTEEF Flowid 1 20 done The following code says that any traffic not matching the other rules is bulk and ends up in the bull class neither economy nor premium te filter add dey DEV parent 1l protocol ip 1 Pric 3U u Meben ip dat U 0 0 0 70 Ilovid 1 30 For any type of commercial service you will certainly want to work on this script a bit Smart users could cheat you by hacking their IP ToS headers or by using a different port number to stow away in the premium class So regardless of how you implement classes of service you ll also need to ensure that users of your service are authenticated with usernames and passwords and you ll need to emphasize good logging so that you always know who s using which levels of service and
415. t downloaded from lib ommolketab ir downloaded from lib ommotketab ir downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 74 Audit a Network s QoS Capabilities HAGK 74 Networks without Quality of Service QoS measures aren t always suitable to carry voice traffic So how do you know whether a network path supports QoS Using pathping and traceroute during peak traffic periods you ll be able to establish whether a particular IP route is a good place for time sensitive traffic like VolP media streams You ll know the jitter and latency qualities of the network you ll have identified problem routers and potential traffic bottlenecks you ll Know whether each router supports Resource Reservation Protocol RSVP a QoS Standard that allows network bandwidth to be reserved for each call and you ll Know how well the network supports 802 1p precedence tagging I explain what 802 1p is in this hack keep going Though Linux is better equipped to provide VoIP services and to serve as a base for troubleshooting Windows does have a nifty command line tool that you can use to determine if IP routing supports basic class of service measures pathping which ships with Windows 2000 and Windows XP lets you see how well your Internet provideror your corporate networksupports 802 1p and RSVP This makes pathping a particularly useful Windows only VolP networking tool On non Windows boxes though you
416. t too uses a sampling resolution of 8 bits anc 8 000 Hz as shown in Figure 2 13 Once these settings are made clicking OK dismisses the Resample dialog and performs thedownsampling on the sound file In Figure 2 14 the resulting sound waveform is telephonized Listen to it now and hear the difference It sounds much flatter less crisp and possibly more robotic It sounds like you re hearing it through a telephone which is the idea here If you are considering putting on hold music or background music on your VoIP system this technique will let you hear roughly how it s going to sound Some musical recordings might sound very poor once they ve been downsampled but now you ll know how they sound before your callers do Figure 2 13 Cacophony s Resample function lets you downsample sounds from hi fi to tele fi downloaded from lib ommotlketab ir downloaded from lib ommoltketab ir Resampling Method Faster Better Quality Channels fs Mono Stereo Custom 2 _Output Bit Rate 8 bit 16 bit Resample Sampling Rate 44100 Hz CD Quality 22050 Hz 11025 Hz Custom Figure 2 14 The resulting waveform is a mono low fi sound downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 23 Record an Audio Chat on Your Mac 1423 Have you ever wished you had a recording of a past conversation Recording incoming audio from any application is a s
417. t lab the IP phone should refer to the IP address of the Asterisk server 10 1 1 10 say being used as its SIP server Configure the SIP User ID setting as 103 too For the DTMF Mode option select SIP Info Then apply the config changes and reboot the IP phone The same configuration options are supported by other makes of SIP phone too The configuration page for a BudgeTone phone that has been configured to use a local SIP server your Asterisk box is shown inFigure 4 1 Figure 4 1 A Grandstream BudgeTone that has been configured to use a SIP server at 10 1 1 10 downloaded from lib ommolketab ir downloaded from lib ommolketab ir el IP Phone Configuration We hup 1 10 1 1 103 dologin htm PPPoE password statically configured as IP Address 10 1 1 Subnet Mask 255 255 255 Default Router 10 1 1 1 DNS Server 1 64 a 140 17 80 DNS Server 2 20 0 hko 2 ol l Server 10 11 10 e g Sip mycompany com or IP address Outbound Proxy if a e g proxy myprovider com or IP address SIP UserID m the user part of an SIP address utha 7 can be identical to or different from SIP Authen ID iiih User ID Authenticate 3 Password Name 0 optional e g John Doe 4 3 3 Allow the IP Phone to Place Calls Via Asterisk Until you authorize a SIP phone to communicate with Asterisk using Asterisk s SIP configuration file you will always rec
418. t numbers involved in the RTP media path downloaded from lib ommolketab ir Figure 6 15 The Cain amp Abel VolP call log downloaded from lib ommoltketab ir File View Configure Toots Help lawe o m olz S LOmMPr Er eaOneo OF A A E sire dr i0 1 1 204 6000 PCM 10 1 1110 10104 POMU 3khe Mano RIP 2005081 322221 96 0 POL 2046000 PCM 10 1 1 1016314 POMU Skhe Mono IRTP 2005061 s22204942 1 wey 10 2 1 20451165 RIP 200S061 32227 13421 ery ER 13 03 2005 18 21 07 13 03 2005 13 08 2005 18 24 56 13 08 2005 Lost packets 09 After you stop the sniffer by clicking the Start Stop Sniffer button on the toolbar again it toggles Sniffing on and off you ll see the filenames where Cain amp Abel has saved the recorded calls The WAV files produced by Cain amp Abel end up tn Program Files Cain VolIP and you can play them by opening them in your favorite sound player or by right clicking them here in the Cain amp Abel GUI 6 16 2 The Tricky Way to Intercept Calls If you want to record a call between two devices that can t run Cain amp Abel like a call between two I phones or a call from a Mac softphone to a Linux softphone the method described in the previous section won t work Instead you need to enable your Ethernet switch to share packets destined for the devices involved in the VoIP call with your PC running Cain amp Abel With your PC connect
419. t on by passing over that peculiarly intimidating Excel menu option And when it comes to those telephony logs the Pivot Table Report function ever since never knew what was missing out on by passing over that peculiarly intimidating Excel menu option And when it comes to those telephony logs the Pivot Table Reports you can generate some very cool call activity analysis List your top callers List your top system users Or just figure out your total long distance and local utilization down to the minute to verify your phone bills We ll do one report that sums activity in minutes by caller and a second report that adds a breakdown of the total minutes for every phone number called by each caller To get started we ll first need to get our hands on the Asterisk CDR logfile as described at the beginning of this hack Next we ll insert a blank row at the top of the worksheet and key in the names of the CDR fields at the top of each column as shown in Figure 4 6 This will be needed to make the Pivot Table Report The names of the CDR fields are laid out in Table 4 1 Once the CDR columns are labeled select Data Pivot Table Report Now you ll get a wizard Click Next on the first step where you ll find yourself being prompted to provide the name of a data range where the source data exists In this case the source data is the first sheet in the workbookthe one that contains the CDR data Select this sheet and drag select all of the colu
420. t that ability this hack will be less than useful and not much fun either But how do you know if you can control the port numbers used by VoIP applications J ust about all VolP software lets you control the port numbers employed by the signaling protocol SIP and the voice stream RTP Asterisk lets you adjust these settings in etc asterisk sip conf and IP phones and softphones like X Lite have user configurable port number preferences To view or adjust the range of RTP ports used by Asterisk take a peek inside etc asterisk rtp conf downloaded from lib ommolketab ir downloaded from lib ommotketab ir As mention in other hacks SIP is just the call signaling protocol RTP is the protocol that carries voice and other data video images data etc The port numbers used by RTP are pseudo randomly selected from a predefined port range In Asterisk this can be configured in etc asterisk rtp conf Actually the default range of 1000020000 is considered by many to be too wide for most installs Please adjust according to the size of your installation Altering the ToS bits can be a little trickier Most IP phones and VolP services tag media packets with the highest possible priority so forcing them to downgrade some of the packets into a lower class of service is hard Ideally you ll build yourpremium and economy service classes using the port number rather than the ToS bits Once you have met these conditions you are rea
421. t to any existing phone numberyour buddy s cell phone in Uruguay or dear old mom in Kalamazoo The trick isn t merely calling folks it s remembering how to dial their numbers If you already do a lot of international calling you re probably familiar with the system of country codes that exists to route calls around the globe Country codes are dialed before area codes as a part of the receiving party s phone number But there are almost aS many country codes as there are countries so you might appreciate this shortcut to help you remember how to dial international numbers on Skype At http www skype com products skypeout rates Skype maintains a nifty script that allows you to enter the local phone number including area code and select the country Then the script shows you exactly how to dial it using Skype as in Figure 3 8 J ust follow the steps on the page enter the phone number and pick a country Besides learning how to dial the number you ll also get a handy SkypeOut rate quote for calls to this number in euros Figure 3 8 The SkypeOut Dialing Wizard downloaded from lib ommolketab ir downloaded from lib eommolketab ir TO i SkypeOut Dialing Wizard ge cS ES IO E ono Google ici SkypeOut Dialing Skype for Windows Need some help figuring out how to dial a number on SkypeOut This page is for you then Whether you re calling your nextdoor neighbour or a friend living in another country you always n
422. tab ir downloaded from lib ommoltketab ir When you use this command it will create a dump file in the current directory that contains every IP packet sent or received by the default interface This file is going to get big pretty quick so run this command only for as long as IS necessary to capture the call you re placing to the server Then at the conclusion of the call hit Ctrl C to stoptcodump 6 15 1 4 Wave goodbye to privacy Now here s the truly fun part The point of vomit is to pick the G 711 RTP packets out of the dump file created by tcodump test file as shown earlier and string them together into a WAV file Try it vomit r test file gt test wav Run that WAV file through SoxX if you need it in another formatL Hack 24 and off you go Just don t record any calls without full knowledge of the participants or you could find yourself in legal trouble downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 86 Log and Record VoIP Streams 86 Biblically speaking there s not a whole lot that Cain and Abel had to do with Voice over IP But the program that bears their names is a really cool VoIP tool If you re not a Unix fan or you just don t have the time to compilevomit and its dependencies to record a call I ve got the solution for you There s a program for Windows calledCain amp Abel It uses the WinPCap library just like Ethereal for packet sniffing network devic
423. ted on newer models like the 7970G You can always specify a differen URL in your color phone s config files that points to a color version of the same image if you need to Support color and monochrome displays The older and current phones will alter an image to fit the screen and color if it does not match This auto correction might not be perfect so you might want t run your image through Photoshop or the GNU Image Manipulation Program GIMP to meet the size and color requirements When did this hack used a PNG format picture of Tux the Penguin Tons of greatimages like this are available at http images google com opened my image in the GIMP to have a look Then resized the happy fellow so that his height matched the height of the LCD 65 pixels Finally downloaded from lib ommolketab ir downloaded from lib ommoltketab ir converted to grayscale and saved Figure 5 4 shows the finished product Cute isn t he Figure 5 4 The Tux logo as he appears on a Cisco 7960 s display then simply uploaded the file to my web server at the URL specified in my 7960 s TFTP configuration file The next time booted up my 7960 there was Tux happy as usual Andrew Latham downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 68 Configure Multiple IP Phones at One Time P68 Uniden s IP phones like the UI P200 are excellent business grade telephones that can be mass configured by TFT
424. tem logs When packets are logged several bits of information about each packet are stored e The protocol of the packet UDP TCP ICMP etc e The date and time the packet traversed the NetFilter chain where it was logged e The size of the packet e The source and destination addresses and ports sockets e The originating MAC hardware address when the packet comes from an Ethernet interface e Of course the prefix that you specify in the log prefix option if any dmesg an application for reading kernellogging messages like the ones you captured earlier provides flat text output that you can redirect to a file or pipe into another application for further filteringlike grep Suppose you want to isolate the traffic prefixed with Chicago into a file by itself You can use this command dmesg grep Chicago gt chicagoVoIP txt Or better yet email that log to somebody perhaps so that they can import it into a spreadsheet for further analysis In the following example hitting Ctrl C will stop thedmesg application and an email will be sent containing the Chicago entries dmesg grep Chicago mail chicagoVolIPadmin example com By combining good logging with good high level network traffic assessmentL Hack 75 you ll have downloaded from lib ommolketab ir downloaded from lib ommotketab ir an excellent grip on what s happening beneath the application layer on yourVolP network You ll know how it all works
425. terisk Managerthe remote API that allows users who ve provided the right password to control certain aspects of Asterisk s operations via a TCP connection If your Asterisk server is open to the Internet pay Special attention to Asterisk Manager log entries To change Asterisk s default log location edit etc asterisk asterisk conf and change the astlogdir directive to a path of your choosing Then make sure that path has appropriate permissions to allow Asterisk to write files in whichever path you choose A sampleasterisk conf follows directories astetcdir gt etc asterisk astmoddir gt usr lib asterisk modules astvarlibdir gt 77a Lies ek astlogdir gt var log asterisk astagidir gt var lib asterisk agi bin astspooldir gt var spool asterisk astrundir gt var run asterisk Syslog can be a target for Asterisk logging output too To enable it use asyslog keyword in the logfiles section similar to the console keyword syslog local0 gt Warning error downloaded from lib ommolketab ir downloaded from lib eommolketab ir Chapter 5 Telephony Hardware Hacks Section 5 1 Hacks 5971 Introduction Hack 59 Record Calls the Old Fashioned Way Hack 60 Make IP to IP Phone Calls with a Grandstream BudgeTone Hack 61 Build a Custom Ringtone for Your Grandstream Phone Hack 62 Tweak Your Sipura ATA Hack 63 Build a Bat Phone Hack 64 Brew Your Own Zaptel Interface Card Hack 65 Build a Speed D
426. terminals in H 323 speak by IP address and port number exclusively whereas H 323 still relies on their Ethernet MAC hardware addresses This makes SIP more flexible There are two ways to enable the configuration change you ve just made One is to restart Asterisk asterisk rx restart Bear in mind that restarting your softPBX might be acceptable at home or even in a small office environment but you d better make sure no calls are in progress if you restart it in any production environment lest you draw the ire of angry phone users Perhaps a better way to handle the additior of a new endpoint to the softPBX is the reload method To do this issue the AsteriSkreload command using the rx shell option asterisk rx reload Or log into the Asterisk CLI as in Turn Your Linux Box into a PBX Hack 41 and issue the Asterisk reload command pbx CLI gt reload No calls are interrupted when the reload occurs This should keep everyone who is using the system at that moment happy Now you can place calls to the Asterisk server and to the other peers and channels that will be connected to it The default configuration installed with Asterisk when you compiled it allows for several interesting demonstrations of its capabilities using aSIP phone You also can try them using an analog phone if you have a Zaptel card installed and a phone hooked upbut that s another projec Hack 44 4 3 4 Listening to Asterisk downloaded
427. th installing X Lite 2nd 3rd 4th sound codecs sound emotes sound files converting between formats with Sox converting with Sox creating for voicemail greetings uLaw Sound Recorder sound recorders Soundflower 2nd SourceForge SoX SOund eXchange 2nd 3rd 4th 5th 6th adding sound effects converting wav file to gsm deepening your voice on greeting messages mixing recorded Asterisk calls resampling and re leveling sounds sound file format conversion spandsp package 2nd sending faxes speech guidelines for professional sounding announcements speech synthesis 2nd 3rd 4th speed dial service on Cisco IP phones 2nd 3rd Speex codec SSL support stability telephony server standalone Ethernet switches standards standards for signaling call events station to station calls stepper switch stop commands Asterisk streaming media devices streaming MP3 radio station for music on hold Strowger STUN Simple Traversal of UDP NATs 2nd building STUN server supernodes Skype SVMTO prefix switches syslog database file System _ command Asterisk downloaded from lib ommoltketab ir downloaded from lib eommolketab ir Index SYMBOL A B C D E F G H CY J K L M N O P Q R S M U M wW x D Z T1 tail f command TCP IP listeners TCP UDP port numbers tcodump 2nd 3rd TDM time division multiplexing 64Kbps PCM bitstream codecs that mimic dedicated time slots
428. th total unrestricted power This is generally considered a bad idea as an exploit to Asterisk can lead to someone taking over your entire machine To avoid this the Apache Web Server doesn t usually run as root This hack shows you how to run Asterisk as a less godly user To do so create a user called asterisk In the following command use the Red Hatadduser command adduser c Asterisk PBX d var lib asterisk asterisk Next you ll need to alter Asterisk s Makefile located at usr src asterisk Makefile Using your favorite text editor find the ASTVARRUNDIR constant in the file and alter its definition to match what follows ASTVARRUNDIR S INSTALL PREFIX var run asterisk The directory referenced here needs to be writeable by the user runningAsterisk just as the directory normally used should be writeable only by root By changing the setting you re allowing Asterisk to use a directory that can be written by its own nonroot user account Now recompile Asterisk using this sequence of commands cd usr src asterisk make clean make install Once the recompile and install are done you ll need to make sure the new user account has appropriate permission to several Asterisk related directories including the one you referenced in the altered Makefile chown R asterisk asterisk var lib asterisk chown R asterisk asterisk var log asterisk downloaded from lib ommolketab ir downloaded from lib ommolk
429. the interprocess communication goodness of the vintage Arexx scripting language on my old Amiga 4000 computer got to thinkingwouldn t it be cool to do some voice hacks on that 25 MHz classic Then realized that the Amiga s pokey 680x0 vintage processors don t even have enough processing power to encode and decode modern audio codecs My hopes of splashing the cover of O Reilly s Make magazine with a really tasty Amiga VolP hack were dashed and returned to the 21st century realm of VoIP Hacks A great place to start building Phlink AppleScript hacks is with caller ID When Phlink receives an incoming call the first script Phlink calls isring which you ll create in the Library Application Support Phlink Items directory The call doesn t have to be answered to execute this script the line just needs to ring Now while I m not going to give you a full blown explanation of AppleScript O Reilly AppleScript The Definitive Guide does a far better job than could hope to anyway these examples should suffice to let you hack Phlink Since we re starting out with thering script take a look at this example which announces the caller ID of the call while it s ringing on incoming_call given callername theName if the Name is COCU True else say You are receiving a call from amp theName as string return false downloaded from lib ommolketab ir downloaded from lib ommoltketab ir end if return callAgain end incoming_cal
430. there s an equally unique hack In this case we re going to launch Phlink under a different user account This user account will be automatically logged in at boot time allowing the Phlink application to launch in that user account Then we ll create an Apple Script login item to switch to the user selection screen automatically giving you the option of logging in as any user you want To get started open up System Preferences and click the Accounts preference pane Click the icon to create a new account and call it Phlink Make sure it is set to log in automatically upon startup Now log in as this user Be sure to enable Phlink as one of its login items To enable login items for the Phlink user return to System Preferences and select the Login Items tab Now you can add Phlin to and remove it from this list causing it to launch whenever the userphlink logs in Now launch the AppleScript editor and create a script with this single line do shell script System Library CoreServices Menu Extras User menu Contents Resources CGSession suspend The purpose of this one line AppleScript is to present the user switching dialog on the screen We ll use this AppleScript to get back to the traditional login screen once thePhlink user has logged in and the Phlink app has launched Save this AppleScript and then make it a login item for the userphlink Be sure it s listed after the Phlink login item Then save and exit Preferences Now
431. this chapter take a look atSkype Hacks O Reilly downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 28 Get Skype and Make Some New Friends 28 Looking for like minded buddies on the Net Look no further than Skype s built in contact search function You can obtain the Skype software at http www skype com Though it s available for the Pocket PC Hack 34 it is best to have your first Skype experience using a desktop operating system Windows Mac OS X or Linux The Pocket PC version is really nifty but not entirely practical Nor is it very customizable So download one of the desktop OS versions from Skype s web site and install it Windows users will need Windows 2000 XP or newer Linux users will need SuSE Fedora Core Debian or Mandriva consult Skype s web site to find out precisely which kernel versions are Supported Mac OS X users will need version 10 3 Panther or newer On all platforms 256 MB of RAM is a reasonable minimum though you might be able to get by with less 3 2 1 Set Up Skype Once you ve downloaded Skype setting it up is as simple as running its installer Windows or dragging its application icon to the Applications folder Mac On all platforms the Skype installer is practically foolproof You ll find that ease of use isSkype s middle name for instance you can call a person simply by double clicking his name The first time you run Skype you can set up
432. ting txt file it looks in the same directory for greeting aif or greeting mp3 Whichever of these files is present is played back to the caller To create your own audio greeting use a recording program such as Cacophony Hack 22 to record and mix this greeting as you see fit To get files into the right format for a greetingi e AIF or MP3use Sox Refer to Create Telephony Sounds with Sox Hack 24 for tips 2 9 3 Answer Fax Calls Phlink can answer fax calls In the Preferences window on the Fax tab is a checkbox to enable automatic answering of incoming fax calls You can have theMac OS X fax viewer handle the faxes or use the script option to handle them yourself And speaking of scripting now that you ve covered the basics of Phlink let s begin customizing downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 16 Run Phlink Even When Logged Off HAGK 16 Phlink is a great application but it s a desktop program not a server app that s made to run in the background So when you log off it shuts down and can t answer callsunless you customize it to do so To get Phlink to launch upon login is really easyjust make it a login item for your user account in OS X Preferences But getting Phlink to stay running even after you ve logged out is a challenge Of course Phlink is most useful when it s running at all times so you need to be able to do this Thankfully for every unique need
433. tions gsm Of course if the file you re converting doesn t have a file extension in its name you can express your intentions more explicitly sox t gsm another_brick t aif another_brick_in_aiff_format By specifying the encoding type with the t option you can tell SoX specifically how to convert the file regardless of filenames and extensions But that s not all you ll find SoX useful for 2 18 2 Adding Sound Effects Aside from converting files between formats SoX can add some cool effects too Equalization reversal chorus reverb time shifting and vibrato are some of the most commonly used effects options Some of these effects are probably more useful in a pro audio environment than in VoIP but there are uses for audio effects even intelephonylike an on hold message that hypes a particular product or event Such an announcement might benefit from a little reverb or delay J ust think about some of the sound effects used in monster truck advertisements beckoning your attendance on SundaySunday Sunday Consider the following syntax which adds reverb to a sound sox bigFootSunday aif bigFootSundayVerb aif reverb 1 1000 15 This example takes bigFootSunday aif and adds 1 000 milliseconds of reverb with a 15 millisecond delay before saving the file as bigFootSundayVerb aif You can combine sound effects too So for instance you can place a reverb and an EQ effect together sox gilmour aif gilmour aif reverb 1 1200 30 highpass 1
434. tis tagged separately from Internet traffic which is indicated by the catch all network addresso 0 0 0 In a VoIP network that s connected to the Internet but doesn t use the Internet as a call path it would be a good idea to log all VolP traffic originating from the Internet Such traffic could be an indicator of system abuse just as email systems are abused by spammers Logging this type of traffic would tip you off to somebody trying to originate long distance phone calls from your systemsomething I ve encountered in the field myself To dig into your VolP traffic consult Table 6 2 which is a list of commonly used VolP port numbers by protocol Table 6 2 Commonly used VolP ports Service Standard port numbers SIP 5060 5061 usually UDP H 323 2099 2517 and 2979 UDP and TCP MEGACO H 248 2944 and 2945 TCP and UDP downloaded from lib ommolketab ir downloaded from lib ommotketab ir Service Standard port numbers TFTP 69 TCP and sometimes UDP Asterisk Manager API 5038 TCP RTP Varies often among 5000 5001 10000 and 10001 always UDP I AX 5036 and 4569 UDP 6 14 2 Read and Analyze VoIP Traffic Logs Once you ve used iptables to tell NetFilter to catch some VolP traffic and log it the log data is stored in the kernel facility of a syslog database file where it can be retrieved using the dmesg command on Red Hat Linux Other operating systems might provide different tools for viewing sys
435. to name a few and there are probably twice that many ways to implement them So how do you choose Good question There s no simple answer because each specification is aimed at a different solution RSVP is a bandwidth reservation technique for WANs and Virtual LAN VLAN is a traffic segmentation technique for LANs Both have implications for QoS that are deserving of their own book I like the hardcover classic Quality of Service Cisco Jitter might in fact be a losing battle depending on how you use VoIP andwhere your VolP calls travel If they go across a network jurisdiction that s out of your control like the Internet it might be impossible to provision QoS and you might never get acceptable voice quality The moral of this story is very basic you can perfect VoIP quality on private controlled networks but on the Internet it s a crapshoot Joel Sisko downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 84 Log VoIP Traffic HACK 84 A Linux PC s built in I P router and firewall NetFilter can be a useful tool for logging Vol P traffic In a scenario where several satellite offices on a WAN or the Internet are linked together as an IP telephony network origin and destination based logging is crucial because it will tell you which office is using the most VolP capacity which is using the least and when it s all being used When a Linux NetFilter firewall is used to protect a
436. trol the time and date configuration on the phone TimeZone sets the local time zone of the phone expressed as an offset of Greenwich Mean Time In this case the phone is 6 hour behind because it s in the Eastern time zone EnableDsT allows the phone to switch from Eastern Daylight Time to Eastern Standard Time automatically and vice versa andSntpServerIP and SntpRetrySec enable the phone to use Simple Network Time Protocol to synchronize its clock with the other devices on the network PreferredCodec gililu g 1la o 29 Language English downloaded from lib ommolketab ir downloaded from lib ommotketab ir The PreferredCodec setting tells the phone which codecs you prefer to use when connecting calls If the device on the other end of the call is deemed not to support your preferred codec the phone goes to the next one in the list G 711 uLaw is most common in the United States and G 711 aLaw is common elsewhere in the world G 729 is a bandwidth conserving codec If this phone is going to be used by a road warrior or telecommuter with unpredictable bandwidth capacity or used over a small 128 kbps at the least wide area link you might consider putting G 729 at the front of this list Uniden s firmware doesn t like spaces in this list by the way Language tells the phone which language to usefor the menu prompts Your choices are English Spanish and French sorry ubergeeks no Klingon StunServerAddr 0 0 0 0 StunServerPort 347
437. u ll do with your VoIP packets downloaded from lib ommolketab ir downloaded from lib ommotketab ir A router with Linux i e a PC with two Ethernet interfaces and Linux installed will help us When yo place this router between two networkssuch as the Internet and your LANyou can use it to enforce QoS measures Thanks to the wonderful folks at http www lartc org was able to create a traffic shaping script that works very well for prioritizing VoIP traffic It is called AstShape and it is included in the AstLinux distribution However you can use it in any Linux distribution that includesiproute2 This Should be just about any major modern distribution out there today The first thing you need to do is visit my web site http www kriscompanies com and click on Downloads Asterisk AstShape The AstShape script will begin downloading and should finish very quickly After you have saved it to your hard disk copy it onto your router in a place like usr local sbin Whatever directory you pick make sure that it is in yoursPpaTH You can see your SPATH by executing echo PATH from the command prompt Once you have AstShape installed make it executable by running the following Chown root root astshape chmod 750 astshape This will ensure that no one other than root can run this script Open AstShape in your favorite text editor and take a look around will show the first few lines of AstShape and explain thei
438. ualizer tool but downsampling alone produces a pretty convincing phone sound Here s how it s done The easiest way to downsample a sound is by using a simple sound editing tool such as Richard Bannister s Cacophony for Mac OS X or Windows Sound Recorder which comes with Windows In essence you open the sound file change itssampling resolution to 8 bits per sample and 8 000 samples per second and then save the file On a standard telephone call there are 8 bits per sample and 8 000 samples per second in the media stream This matches your prerecorded sound ti the sampling resolution of a typical phone call In Figure 2 12 an MP3 sound file has been opened in Cacophony Its left and rightwaveforms are displayed since it s a stereo sound Figure 2 12 An MP3 file opened in Cacophonyready to telephonize downloaded from lib ommolketab ir downloaded from lib ommoltketab ir 99 P 03 Karaoke Superstar mp3 D Current 00 00 00 0 a F Length ooz me i a Play Pause stop Selection 000000 FF 6 Record 1 a B m H n y ee fa sch a i j L i i fi I Ih 116384 16 bit 44100 Hz Clicking Cacophony s Resample function displays the Resample dialog which lets you specify the exact factors you ll use to downsample the sound Alltelephony applications are mono not stereo All legacy telephone equipment and most VoIP equipmen
439. udo passwd root You ll be prompted for a password This password will establish the root user s password so that you can use the root account from now on Yes that simple command turns on the root account Now you should be able to launch Asterisk on the Mac using the preceding Asterisk command If you get few screens full of log output that eventually ends up at aPBX prompt you ve launched Asterisk successfully Yet even after all that Asterisk won t be set to start automatically every time you boot your Mac so you ll need to make sure it s launching every time This is easy In System Library Startupltems create a folder called Asterisk In that folder create a text file called Asterisk and place the following code inside bin sh it Ted s Asterisk Telephony Server Startupitem it feces Tre common StartService echo Starting Asterisk telephony usr sbin asterisk vvvg amp StopService echo Stopping Asterisk telephony usr sbin asterisk rx stop when convenient RestartService echo Restarting Asterisk Stopservice StartService Pincervice 9S1 This script handles the proper startup and shutdown of Asterisk when the system boots and shuts down Likewise rebooting your Mac PBX will also automatically shut down and restart Asterisk Before you reboot though you ll also need to create a file calledStartupParameters plist in the same downloaded from lib
440. uired for each option using the keypad Use this menu only to set up the IP address subnet mask and router default gateway address To get the phone enabled for the next configuration step turn off DHCP and assign an IP address subnet mask and router address downloaded from lib ommolketab ir downloaded from lib ommotketab ir More advanced configuration is performed using the BudgeTone s built in web configuration tool When you access the IP address you assigned to the phone using your web browser you ll be prompted to log in to the phone The default password isadmin Then you ll be confronted with a big page of configuration options Many of these options are available only through this interface not from the phone s keypad menu After you apply your configuration changes you need to power cycle the BudgeTone Some IP phones offer a Telnet interface rather than or in addition to a webb ased one To use these tools you must connect to the phone with a Telnet client rather than with a web browser In any event once you ve set the network configuration on theBudgeTone ping its address from another host on the same network subnet to make Sure it s speaking Transmission Control Protocol nternet Protocol TCP IP 4 3 2 Set the IP Phone to Use a SIP Server The IP phone whose address I ll assume is 10 1 1 103 must be set to use yourAsterisk box as a SIP server if you re to interact with the Asterisk demo In your tes
441. uld be recorded to paper tape or deciphered directly by skilled operators Unfortunately design constraints of the straight key limited its transmission capabilities to a mere 2C words per minute Additionally the vigorous brass pounding required of early telegraphers sometimes led to a repetitive stress injury calledglass arm Known today as carpal tunnel syndrome Sanders Kleinfeld was the production editor and Audrey Doyle was the copyeditor forVolP Hacks Sanders Kleinfeld proofread the book Philip Dangler and Claire Cloutier provided quality control Elle Troutman Zaig wrote the index Marcia Friedman designed the cover of this book based on a series design by Edie Freedman The cover image is from the Classic Business Equipment CD in the Classic Photographic Image Object Library Linda Palo produced the cover layout with Adobe InDesign CS using Adobe s Helvetica Neue and ITC Garamond fonts David Futato designed the interior layout This book was converted by Keith Fahlgren from Microsoft Word to Adobe FrameMaker 5 5 6 using open source XML technologies The text font is Linotype Birka the heading font is Adobe Helvetica Neue Condensed and the code font is LucasFont s TheSans Mono Condensed The illustrations that appear in the book were produced by Robert Romano Jessamyn Read and Lesley Borash using Macromedia FreeHand MX and Adobe Photoshop CS This colophon was written by Sanders Kleinfeld downloaded from lib ommolketab ir
442. umber of the calling party as signaled by the calling peer On the PSTN caller ID signals originate from the exchange switch of the calling party We can use this variable to figure out whether we want to forward a call Unless you re paying for caller ID service your Asterisk server won t receive caller ID signals and this hack won t work Some phone companies and just about all VolP service providers include caller ID for free Consider the following incoming Priority 1 Check to see if the call is Mom s home phone gt It 20 GO CO priority 27 at MOL Continues EO Priority Z exten gt s 1 Gotolf CALLERIDNUM 3138853352 5 2 Priority 2 See if the call is Mom s cell phone It S0 GO to priority 3 1f not continue Co Priority exten gt s 2 GotoIf f S CALLERIDNUM 3132981848 25 3 gt Priority 3 and 4 This call s ot Mom so Just drop it exten gt s 3 Playback carried away by monkeys exten gt s 4 Hangup Priority 5 Dial my cell phone for 30 seconds to connect Mom exten gt s 5 Dial S MYCELLPHONE 30 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Priority 6 and 7 If not answered in time drop the call exten gt s 6 Playback carried away by monkeys exten gt s Hangup Note the syntax of the Gotorlf command If you re familiar with logic control structures in programming the should look like a then in an tf then workflo
443. unes hack Your iTunes music library makes the perfect source for on hold music or just for a cool telephone gimmick like a remote phone jukebox The following AppleScript will actually search through your iTunes music library and find non copy protected songs i e songs imported from CDs or MP3 files and not purchased online to play for the caller on do_action given call my_call tell application iTunes set track_found to false set num_retries to 0 repeat until track_found set my_track to some file track of library playlist 1 if kind of my_track contains Protected is false then set track_found to true else set num_retries to num_retries 1 if num_retries gt 100 then exit repeat end if end if end repeat set my_song_file to the location of my_track end tell tell application Ovolab Phlink tell the_call to play my_song_file as alias end tell end do_action downloaded from lib ommolketab ir downloaded from lib ommotketab ir my_song_file Is a variable that stores the location of a song to play for the caller which is triggered to play in the fourth from last line in the script You can trigger this bit of AppleScript from any of Phlink s event handling scripts ring greeting hangup etc The Ovolab Phlink user manual written by fellow O Reilly author Matt Neuberg provides a scholarly introduction to all of Phlink s event handling scripts 2 13 1 Automatically Pause iTunes Resume iTunes Ovolab p
444. up a conversation or just to raise the silliness level a bit Yes even I the ever stodgy VoIP aficionado have been known to be silly once in a great while To get started with sound emotes you ll need to pick up a copy of Porto Ranelli sHotRecorder 2 0 for Windows http www hotrecorder com This ad supported shareware application lets you select from a small batch of prerecorded sound emotes including applause a room full of people laughing a baby cryingin other words a wisecrack for every occasion Also included in HotRecorder is a voicemail utility for Skype I prefer Skype Answering Machine though and a sound recorder so you can add your own sound emotes To play them back during a Skype conversation just click the one you want from the selection on the Emotisounds tab in the HotRecorder application For some real fun try importing sound clips from your favorite movies That s just sound though To bring the visual aspect of emoticons into the 21st century download a copy of 3D Avatar Messenger http share skype com directory Skype 3d avatar messenger view and install it on your Windows PC It s kind of hard to describe what this application does though downloaded from lib ommolketab ir downloaded from lib ommotketab ir Yahoo s Mvironments are probably the closest analogy Figure 3 15 shows the application s interface 3D Avatar Messenger is a Java application that uses the Skype API to sendanimated thre
445. up and running you re ready to start hardening Though I ve geared this discussion toward an Asterisk server it is principally accurate for any softPBX Why not use Windows for a phone server Well as it turns out Windows doesn t support nearly the selection of free high qualitytelephony software such as Asterisk SipX OpenH323 and other open source server side stuff It also turns out that Windows itself comes towing a rather pricey license fee So figured I d stick to what you can download from the Net without breaking the bank The less money you spend on licensing fees the more you can spend on other parts of your telephony solution When hardening a server you need to examine two basic aspects of the soft PBX the software that installed and the software that s running In terms of hardening the software that s installed but not running or not needed the course of action is quite simple remove it downloaded from lib ommolketab ir downloaded from lib ommotketab ir 7 2 3 1 Remove unnecessary software That means getting rid of Bind if you re not using it removing Apache if it isn t needed and unloadinc MySQL if you ve no need for it J ust because the software isn t running doesn t mean you can t use it to facilitate a sophisticated security exploit so remove it altogether if you don t absolutely need it Since we re dealing with Asterisk you can disable a number of modules to reduce the risk of security expl
446. ur old fashioned residential style phone The device is sort of an analog telephone adapter ATA that allows your PC to act as a gateway between the analog phone and the Skype network Too cool 3 14 1 Make the Connection For this hack you ll need a computer running Windows 2000 or newer and the Actiontecl nternet Phone Wizard pictured in Figure 3 16 which needs to connect to your PC s USB port The two RJ 11 ports on the device s back allow you to connect it to an analog phone and a plain old telephone line This pass through connection allows you to place certain calls on your ordinary phone line if you prefer That connection can also handle emergency calls 911 which is important since Skype has no provision for 911 call routing Perhaps the wizard s most valuable characteristic is the way it translates Skype s features so that a traditional telephone can use them For instance when you receive a Skype call the phone will ring and you can answer it later while you re still on that call if you receive a second Skype call your phone will use a call waiting signal to let you Know another call is ringing in This way you can switcl between two Skype callers as you would with call waiting on a legacy telephone line Figure 3 16 Actiontec s I nternet Phone Wizard downloaded from lib ommolketab ir downloaded from lib ommoltketab ir Also supported are conference calling and speed dial integration This way you still can
447. urce code It is extremely easy to use and will do all the work for you based on your input At a shell prompt simply type addmailbox to get started It will ask you for the voicemail context Enter vmserv It will then ask you for the mailbox number Enter 8000 Congratulations You just gave Lisa Hayes a mailbox To create mailboxes for the rest of the users simply rerunaddmailbox replacing 8000 with 8001 8002 and so on 7 10 2 Final Setup downloaded from lib ommolketab ir downloaded from lib ommotketab ir If you would like to use the voicemail to email functionality provided by Asterisk and AstLinux you will need to edit the etc rc conf file and fill in the variable smrp_SERVER with the IP address or hostname of an SMTP server that will relay mail for your Asterisk server If you are delivering email to only one domain name you can use the SMTP server for that domain name as it will accept mail for it from any system After making any other configuration changes you will want to reboot your server to verify its configuration Now how are you going to get your callers into this system If you have a SIP platform you simply need to send the callers into the PBX with a simple SIP URL such as8OO0 lt I PAddress of Asterisk server gt This SIP URL will put the caller into Lisa Hayes s mailbox If you are using PSTN hardware POTS T1 E1 etc you are going to need to set up Zaptel and Zapata Hack 44 The same principles
448. ure 3 4is blue believe it or not If blue isn t your color you can visit http www skype com community for a host of other colors sizes and styles of ready made Skype Me graphics Once you stick one of these buttons on your home page all Skype users need do is clicl it and seconds later you ll receive a Skype call If the user clicking your Skype link doesn t have Skype installed and is ona Windows machine the callto handler will try using NetMeeting instead For this reason you should add a Skype download link for them and suggest visitors install it before they try to Skype you using your link downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 32 Speak Jyve 32 Skype is not just a communications tool it s a social networking platform too Skype inherits many of the cool features of instant messaging includingpresence that useful list of status names that your buddies can use to see whether you re available to chat This includes status indicators that tell if you are Away Busy or Availableand of course Skype Me which allows you to connect to people around the world interested in chatting with strangers But wouldn t it be cool if you could put a status indicator on your web site so that folks would know whether it s a good time ti Skype you Thanks to Jyve you can do precisely that J yve is a social networking service for Skype that adds some pretty useful features to an
449. used for logging output and how detailed each should be The syntax for this section is filename gt level level level Consider the following logging configuration general logfiles messages log gt notice warning error debug log gt notice warning error debug verbose In this example messages log will contain a digest version of Asterisk s logging output and debug log will get everything in minute detail Be careful with logs thoughAsterisk won t start once the logfiles reach 2 GB insize On a busy system a file like the precedingdebug log would hit that size pretty quickly so make sure your logfile rotation includes Asterisk If you use console aS a logfile name Asterisk will assume you mean the console device not an actua downloaded from lib ommolketab ir downloaded from lib ommotketab ir logfile So if you add this to the logfiles section the desired level of logging will be output to the console session where Asterisk is launched logfiles console gt warning error Some attackers cover their tracks by removing commonly used logfiles that could contain evidence o their tampering with the system So it s generally a good idea to keep logfiles in anondefault place That way if an attacker uses an automated program to remove logfiles the program will be less likely to find and destroy Asterisk If you were an intruder and wanted to control Asterisk you might start by attacking the As
450. useful yet often overlooked web interface parameters To reach the web interface simply enter the Sipura s IP address in your web browser Once you see the gray status screen click the Admin link in the top right hand corner When the page refreshes click Advanced You should see several more tabs appear Now we are ready While fully encourage you to review the user guide and browse the configuration pages have Summarized in Table 5 2 my Top 10 Sipura options for your hacking pleasure Figure 5 2 The I nfo tab on the Sipura ATA s web interface downloaded from lib ommolketab ir downloaded from lib ommoltketab ir SIPURA technotogy Ine S Sipura Phone Adapter Configuration Info System User 1 PSTN User cern Lee bac etyenced System Information DHCP Enabled Current IP 192 168 0 81 Hos Wane SipuraSPA Demain kialon com Current Netrmask 259 295 255 0 Current Gateway 192 168 0 1 Primary ONS 192 168 0 1 Secondary ONS Product Information Product Name SPA 3000 Seral Number BBO12D406633 Software Version 2 0 11 GWg Hardware Version 2 0 1 5886 MAC Address OO0EOBCAESIA Oient Cerificate Installed iSystem Status Current Time 6 3 2005 02 45 17 Elapsed Time 1 day and 08 20 07 Broadcast Pits Sent 0 Broadcast Bytes Sent 0 Broadcast Picts Recv 5169 Broadcast Bytes Recv 473913 Broadcast Pkts Dropped a Broadcast Bytes Dropped RTP Packets Senti 20218 RTP Bytes Se
451. vanced managed switches These days inexpensive switches like the NETGEAR GS605 provide support for ToS and 802 1p By placing such a switch between your broadband router and your VoIP device witt voice traffic prioritized you can ensure that outbound voice streams get sent to your broadband router before anything else 1 7 3 What Happens When VoIP Passes Your Router Unfortunately no matter how well prioritized and orderly your VolP media traffic is when it s forwarded by your broadband router it still might get slowed down ripped up and otherwise tattered as it makes its way across the Internet The same is true of mediapackets that come from downloaded from lib ommolketab ir downloaded from lib ommotketab ir the Internet to your routerthe packets carrying the voice of the person speaking to you Since you re receivingnot transmittingthose packets you can t really prioritize them That s the responsibility of the routers that carried the packet to your routerand many routers on the Net these days are ignorant of QoS In short you can control traffic sent from your network but not traffic sent from other networks to yours At first blush this sounds like a threat to broadband VoIP but over the last few years many have discovered that the outbound traffic is all you really need to prioritize to have success with a broadband TSP This is because most broadband ISPs limit the amount of outbound bandwidth available to each cu
452. ver Hack 49 and Connect a Legacy Phone Line Using Zaptel Hack 44 for ideas on how to use legacy TDM interface cards Now let s assume e You have three offices one in Chicago one in Tokyo and one in London e None of the offices has more than 99 separate extensions e They all have 24 7 Internet connectivity e They all have static IP addresses e Their Asterisk installs are directly on the Internet or the network administrator has forwarded passed UDP port 4569 to the Asterisk server in each location It is worth pointing out that the last three items do not necessarily have to be the case With all of Linux s power you can actually work around those issues While this is too much information for me to cover here you can use the register feature of AX with dynamic IP addresses even those downloaded from lib ommolketab ir downloaded from lib ommotketab ir behind NAT A simple Google search should return the necessary details to use register statements successfully to work around those problems Let s also assume that none of your locations has overlapping extensions That is each location has globally unique extension numbers For the purposes of this hack we are going to assume that your extensions are set up like so Chicago 81XX where XX is 0199 Tokyo 82XX London 83XX So your first extension in Chicago is 8101 and your last possible extension in London ts 8399 If your extensions ar
453. very consistent in nature can vary from your readings here sincetraceroute uses ICMP packets that are very bursty in nature The routers along the route that you re evaluating might already be configurec to treat VoIP traffic with greater precedence but at least you ll get an idea of the general service conditions along the route 6 4 3 See Also e http www microsoft com technet for more information on pathping downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 75 Graph Latency and Jitter 75 Use sip_ ping pl to record latency and jitter data in a nice pretty graph It s time to step beyond the basicping and traceroute and to graph packet flow using Perl In Monitor VoIP Devices Hack 72 you saw how to determine the availability of a SIP host using Perl and the SIP OPTIONS packet The next logical step is to time how long it takes to receive a response to monitor the latency on the link between you and the device you re monitoring This hack uses the script from Monitor VolP Devices Hack 72 but this time you use thet option to time the round trip sip_ ping pl t 192 168 0 123 Uad This shows me that it took almost 51 milliseconds for my Snom handset to respond to a SIP request This isn t a terribly useful test however considering I m timing this response from my desktop computer hooked up to phone s built in switch Latency and its cousin jitter starts to be an issue
454. w broadvoice com Yes Broadvox Direct http www broadvoxdirect com No Net2Phone http www net2phone com No nikotel http www nikotel com No Packet8 http www packet8 net No SOYO http phone soyo com No VoicePulse http www voicepulse com Vonage http www vonage com Once you ve subscribed to a VoIP TSP service many allow you to subscribe on the Web and you ve received your ATA in the mail you ll probably be itching to hook it up and use it Most of the time setting up an ATA is straightforward All ATAs have anEthernet interface for connecting to your network via an eight wire CAT5 patch cable with two RJ 45 connectors and one analog telephone interface for connecting to a residential style single line phone using afour wire patch cable with two RJ 11 connectors 8x8 Inc s DTA 310 standard equipment for Packet8 service is such a device So is the Sipura SPA 2000 Hack 62 pictured in Figure 1 1 Figure 1 1 The front and rear panels of the Sipura SPA 2000 ATA downloaded from lib ommolketab ir downloaded from lib ommotketab ir But other ATAs might offer additional capabilities For instance the Sipura SPA 2000 standard equipment for VoicePulse service offers an extra analog phone connector so you can easily connect two phones or perhaps a phone and an answering machine As shown inFigure 1 1 the SPA 200 s front panel has two phone connectors and a status LE
455. w developments and cool little tweaks in Apple s flagshipiLife applications iTunes iPhoto iDVD iMovie and GarageBand But for all the pomp and circumstance surrounding these ravishing rollouts Apple seems to have missed a critical component of the digital lifestyle one that was aroun long before DVDs or MP3stelephony Fortunately an Italian company called Ovolab has created a really cool application that serves as the missing telephony link for iLife Phlink is a hardware software combination that answers calls with a voicemail greeting logs them and even allows you to set up AppleScripts that you can control remotely from a touch tone phone The hardware piece of the Phlink setup is a USB device with two RJ 11 type ports one for your standard phone line and another for your analog legacy telephone The software component available at http www ovolab com phlink consists of an application that looks like iTunes see Figure 2 10 You get all of this for less than the cost of dinner at a really nice restaurant Installing Phlink is a snap Just plug the USB interface into an available spot on your Mac or its keyboard There s no power adapter to worry about thankfully Plug your phone line into the line port on the USB interface and plug your analog phone into the aux port Then drag the Phlink icor from the included CD ROM to your Mac s Applications folder Figure 2 10 Phlink s main interface downloaded from lib ommolket
456. w statement A colon separates the then target from the else target The targets correspond to the step numbers in each of the exten directives of course If you think Asterisk dial plan syntax is atrocious well you re right Don t get too hung up on it now though There are some good references out there for Asterisk dial plancommands including http www voip info org and the unforgettable classic Switching to VoIP O Reilly For now just keep hacking and you ll get comfy 4 7 1 Hacking the Hack With a little modification you should be able to forward incoming calls todifferent numbers depending on their caller ID values J ust rearrange the previous example so that eachGotolf numbered target step contains aDial command with a different phone numberone for Mom one for Dad etc You can even forward calls with no caller ID signals like those from telemarketers to a fun destination Hack 48 downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 47 Report Telephone Activity with Excel i With a little help from Microsoft Excel you can dig into your CDRs chart your top callers and create utilization records for the users of your PBX server Most commercial softPBX systems provide a detailed logging mechanism for keeping track of when and to whom calls were made and received Asterisk provides this too n var log asterisk cdr csv Master csv a flat text log of all call activity i
457. ware readiness for QoS It will tell you which routers support RSVP and which routers need to be either reprogrammed or upgraded to support it 6 4 2 Measure the Latency Time and Jitter on a Call Path downloaded from lib ommolketab ir downloaded from lib ommotketab ir The cumulative latency on a route is a good indicator of how latent it is and therefore how well it wi work as a VoIP call path An easy way to record latency between hops routers on a route is by using traceroute on Windows tracert Using traceroute you can discover the route to the host at the specified address send severall CMP packets to each hop on the route and then be shown the following The highest round trip latency to each router in milliseconds The lowest round trip latency to each router The average round trip latency to each router The IP address and or hostname of each router Whether an ICMP ping response was received from each router The syntax for traceroute is very simple traceroute www macvoip com Tracing route to www macvoip com 65 31 69 11 over a maximum of 30 hops il 1 ms 1 ms 1 mas 10 le elal 2 14 ms 13 ms 18 ms 10 248 164 1 3 18 ms 16 ms 12 ms bicOl elyehel oh attbb net 24 131 64 38 4 19 ms 21 ms 34 ms 12 244 65 61 5 lms 23ms 24 ms 12 244 72 70 0 an 7S 20 me 206 MS torb p0l2401 polpecip e lti ne IALA T oL Ws 27 ms 27 MS Tbri clo n dny lip ati net 14s 12227 0 0 ma 20 N3 34 mS ggr P00 nony Ipate
458. widgets yahoo con downloaded from lib eommolketab ir downloaded from lib ommotketab ir Hack 36 From Podcasting to Skypecasting 36 With some simple recording tools you can easily integrate Skypeand other VolP calling softwareinto your podcasts Audio blogging or podcasting as it s called is a form of amateur radio broadcasting that uses the Web as a means of distribution just as blogging is a personal form of journalism that uses the Web for distribution Podcasters record their broadcasts in MP3 audio files and their audience downloads them for listening on a PC iPod or other portable MP3 player one is available For a more detailed explanation of podcasting check out Podcasting Hacks O Reilly R The art and science of podcasting deserves its own book Fortunately a great 3 10 1 If You Build It They Will Come Creating cool content for podcasts isn t always easy Without a broadcast license it s not a simple thing to use legally anyway copyrighted music in podcasts since their distribution is heavily protected by the long legal arm of the Recording Industry Association of America RIAA the same folks who dumped millions into suing harmless teenagers who download Cold play singles that sell fo 99 cents apiece through iTunes But digress The point is that podcasts can get old really quickly unless you have a constant flow of good audio content and this doesn t come easily Fortunately the
459. will need at least the following e A Soekris net4801 PC Engines WRAP board net4801 image or any modern PC hardware with z Pentium or better processor downloaded from lib ommolketab ir downloaded from lib ommotketab ir A CompactFlash IDE adapter i586 image e A USB CF adapter or an IDE adapter A computer already running Linux or Windows e A CompactFlash card of 32 MB or larger a 256 MB SanDisk is recommended A PC with two Ethernet devices one is acceptable as discussed shortly 7 9 4 Install from Windows have tried very hard to make AstLinux as easy to install and configure as possible The simplest wa to get started is to go to the AstLinux web site athttp www astlinux org and look for the Downloads section Once there you should find the Windows install package which you will want to download and save to a local disk Once you run the install package follow the prompts until it notifies you of a successful installation Under the Programs group in the Start menu you should see a new entry called AstLinux with shortcuts to creating CFs and some documentation Attach your USB CF adapter with CF inserted and click on the shortcut for the image that you woulc like to create A screen will appear prompting you to select a target disk This is the harder part of the install because many people don t know one disk from the other What can tell you is that it is usually the last disk listed but cannot
460. wo quick config changes to fire up your POTS line The first change Is in etc zaptel conf Add the following lines to the end of the file Tokos loadzone us default zone us The first line tells the Zaptel configuration program ztcfg to set the X100P card to use FXS Kewlstart signalinga variation of conventional FXS loop start signaling The number1 is referenced because only one Digium card is installed and it has only one channel like the X100P card so it s card number one and its channel will be 1 as well If two cards were installed side by side the first line would say fxsks 1 2 Instead If there were more than one channel per card like the TDM400P a single channel number would be used for each channel on that cardi e xsks 1 4 for a card with four lines attached xsks 1 8 would work fine if you had two TDM4OOPs installed with four FXS modules apiece The next two lines in the code snippet localize the FXS signaling functionality of the X100P interface with loadzone and defaultzone Other valid zones include fr de and uk Now you might be asking yourself Why am configuring the FXO interface card to use FXS downloaded from lib ommolketab ir downloaded from lib ommotketab ir signaling The answer is simple to communicate with the FXS device interface at the central office the local interface must use FXS signaling Recall from this chapter s introductionthat only FXS devices can receive signals meaningfully fro
461. work is calledARP poisoning The result is that your PC intercepts the packets destined for the intended MAC address so you can do with them whatever you like Once intercepted the packets must be forwarded to the correct MAC address or a denial of service will occur on the device you re snooping In the case of VolP your PC can record or play in real time the media stream before passing the packets to the actual intended receiver A classic man in downloaded from lib ommolketab ir downloaded from lib ommotketab ir the middle hack this technique is simplified by the outstanding network tool Cain amp Abel To get started use Cain amp Abel s host discovery tool Click the Sniffer tab and then the Hosts tab at the bottom of the GUI Then click the in the toolbar This will pop open a dialog where you can tel Cain amp Abel to discover all of the devices on your network used it to discover the IP andMAC addresses of my Cisco 7960 phone 10 1 1 104 and my Asterisk server 10 1 1 10 Both are listed in Figure 6 16 Figure 6 16 Cain amp Abel s list of discovered hosts on the LAN Sele Fie View Configure Tools Help jogo Bo 2 t3 0mPr er eaonese i Hy Protected Storage Network MB Sniffer gif E Traceroute E ceou iA JP address MAC address OUI fingerpri _Host nam esi B16 B8 cr mo m mo 10 1 1 104 QOOZFDO6DF54 Cisco Systems Inc f 10 1 1 1 003046157305 DELTA NETWORKS INC
462. y recording a conference call you can create a podcast friendly panel discussion Even better if you use Alex Rosenbaum sSkype Answering Machine software to greet callers while you re away Hack 37 you can even use Skype to do the broadcasting via the greeting the Skype Answering Machine plays for callers as they Skype you Check out the next hack for the details 3 10 5 Experiment Your Way to a Perfect Skypecast A lot of variables are involved in recording a podcast Many of the fundamental techniques ofvoice recording carry over from the radio broadcasting industry The human voice is very dynamic It s very quiet one moment and very loud or boomy the next Yet the human ear prefers a much more controlled aural dynamicthat is we d rather listen to the human voice at a consistent volume level especially when there s background noise That s why professional radio broadcasts arecompressed downloaded from lib ommolketab ir downloaded from lib ommotketab ir or processed so that the quiet moments are louder and the loud moments are quieter This way the sound is at a much more consistent volume level traditional phone calls work in much thesame manner Microphones matter A microphone with awindscreen that afro shaped sponge you see on news reporters microphones will reduce the unprofessional sounding harshness caused when pronouncing the letters f and p A microphone with a wide frequency response range will pick up more
463. you Click the Options tab as shown inFigure 3 10 to see where the fun settings are Figure 3 10 Skype Answering Machine s Options tab downloaded from lib ommolketab ir downloaded from lib ommoltketab ir 4 SAM tedwallingford x Messages Options About Answering Settings When Online answer after 2u seconds When Away NA DND answer after 2 seconds Greeting Message Settings Voice greetingwey Hel Text Cal answered by SAM Answering Machine for Sky Recording Voice Message Settings Max recording time for new voice messages 120 seconds a o s You can tweak the answering delay time and select your own recording to use as the outgoing greeting SAM supports only WAV files but you can record those easily enough using the Windows Sound Recorder But since SAM also answers calls with atext message you can specify the text to use as your auto away message You can also set the maximum length of incoming recordings The Advanced button will allow you to establish a second configuration set for calls originating from folks that aren t in your buddy list as shown inFigure 3 11 Figure 3 11 Skype Answering Machine s Advanced Options dialog downloaded from lib ommolketab ir downloaded from lib ommoltketab ir SAM Advanced Options Callers that ar not on your Buddy List Settings a Use other eile for narn eile List callers Greeting Message Settings Voice pics
464. you can connect the ATA to any modular jack in the house and all of the analog phones connected to the other jacks will be able to use the service provided through the ATA Before you do this however it s very important to disconnect the wires from the phone company at the demarc because the electric current supplied over thephone company lines could damage the ATA It s best to find the demarc while your phone company service is activethat way you can hear the dial tone disappear when you ve disconnected the right pair of wires at the demarc Find your demarc usually a gray or brown box mounted on the exterior of the building Inside the box is a cross connect terminal with screw taps On one side of the terminal are the wires going into the building On the other side are the wires from the phone company downloaded from lib ommolketab ir downloaded from lib ommotketab ir Carefully disconnect the wires from the phone company the dial tone on the modular jacks inside th building should disappear You can take a cordless phone with you to the demarc to listen while you re working Even if your phone company lines are deadthat is you have no phone company serviceit s still a good idea to disconnect them Disconnecting the wires from the phone company side of the demarc will prevent electrical damage to your ATA in the event the phone company turns the lines back on by mistake Don t accidentally disconnect your DSL line If you have
465. you just don t want to get off the couch and walk to the caller I D display Your Linux server understands and wants to help My first exposure to synthesized speech was on a Commodore 64 the speech demo took an eternity to load off a floppy diskette and the speech sounded like an English as a Second Language student was speaking it directly into a pillow Today with DSP and decades of additional speech programmin research in the bag synthetic speech is much more passable and folks are constantly coming up with novel uses for it In Detroit have a buddy whose Linux server used to announce logfile entries and tell him when the doors around his house were opened and closed While this speaking server was mysteriously silenced right around the time he got married still love his hack Adapting speech capability arounc Asterisk is a logical use for two of my favorite pieces of open source software Asterisk and Festival the University of Edinburgh s speech synthesizer With a little bit of dial plan configuration your Linu Asterisk server will be announcing your incoming calls in no time and announcing a whole bunch of other stuff if you want Mac OS X users will find that Festival is similar in some ways to thesay command on OS X though Festival provides much more functionality to Asterisk than say does For example you cannot use Say to speak to callers as you can with Festival First you ll need a sound card installed and work
466. you re ready to try it Reboot your Mac If all goes well your Mac will log in as the Phlink user automatically launch Phlink and return you to the screen where you can choose which user to log in as or the username and password prompt if that s how your Mac is configured Now Phlink will handle your calls even while you re logged in as another user or logged off altogether because it s running as its own user in the background downloaded from lib ommolketab ir downloaded from lib ommotketab ir Hack 17 Greet Callers Differently Each Day rei7 Many voicemail systems let you use a different greeting depending on the day of the week or the time of day but not Phliinkthat is unless you know how to use cron As you know from Let Your Mac Answer and Log Your Calls Hack 15 a file in the Library Application Support Phlink Items directory contains your outbound greeting Either it s greeting txt for a synthesized voice greeting or it s an audio file in the form ofgreeting aif or greeting mp3 But suppose you want to use a different greeting depending on the day of the week Thank goodness for cronthe trusty Unix relic is a workhorse Assuming you have all of your daily greetings stored in the same place you can create a script that cron can use to update the greeting based on the day of the week Here s the directory listing on my machine Mac Mini Library Application Support Phlink Items kellyS 1s al total 326
467. your Internet connection provides If you double click the icon you ll get a pop up dialog that gives you more details about your available bandwidth and you ll find out Gizmo s opinion of your Internet connection apparently Gizmo doesn t particularly care for mine seeFigure 2 5 Figure 2 5 Gizmo tells you how well it expects to perform using your broadband connection downloaded from lib ommolketab ir downloaded from lib ommoltketab ir gt Gizmo Project Call Quality Assistant Primary Call Quality Factors Tranter Protocel UDP pretened Downboad 512 kby s Upload 128 kbs Other Contubuting Factors Cal quality t determined by condition on both sides of the call A conmechon with someone wih a poor GOnnechon of 4 mobie phone might result in reduced call quality 2 6 2 Share the Love If you ve come this far with Gizmo and still haven t placed an actual call why not use that 25 cent call out credit that SIPphone Inc provides and call a buddy to tell him about it Do this by typing your friend s phone number into the top drop down list and clicking the round phone button After a few seconds your friend s phone will ring and you ll be able to talk for as long as a quarter will allowwhich isn t long so you might want to purchase morecall out credits by clicking the Out icon on Gizmo s Home tab 2 6 3 Also Worth Checking Out If you re really into desktop VoIP and you d like to experiment with other sta
468. ype works equally well uncradiled provided you have an 802 116 or better wireless connection This is plenty of bandwidth for Skype calling but keep in mind a couple of things downloaded from lib ommolketab ir downloaded from lib ommotketab ir e Using wireless will drain your Pocket PC s battery e Wireless LAN will add some delay to your voice signal transmission often resulting in noticeable lag during the conversation To test the sound of your Skype calls read Get Skype and Make Some New Friends Hack 28 e The range of most wireless LANs is a few hundred feet less than that indoors so your PocketSkype will only be as mobile as your LAN allows Pocket PCs aren t known for their stunning wireless reception so if you plan to rely on your PocketSkype setup invest in an extended range base station Try to use a channel that performs very well try pinging over your wireless LAN and watch for jitterbig variances in the ping time this is bad and locate the base as close to the center of its desired range as possible If you can t cover the entire area connect a WLAN extender like Apple sAirPort Express in the area the base station doesn t cover Or use a couple of base stations 3 8 1 Don t Forget Wireless Security To make sure your neighbor can t inadvertently access your wireless LAN from his patio and possibly make his own Skype calls on your dime you d better secure your wireless access point This is good p

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