IP-XX User Manual - Inteligentne Produkty
Contents
1. r Send this call through trunk Use Trunk Strip digits from front and Prepend these digits before dialing LJ Use FailOver Trunk fail over Trunk v Strip digits from front and Prepend these digits before dialing 9 Cancel port4_9 9 Ports 4 Ports 1 x ojx IK jx le jx x x x x x x port4_17909 _90 Ports 4 None Selected Calling Rule Name The name of your calling rule V vY Be Pattern Describe what numbers should use this rule X Any Digit from 0 9 Z Any Digit from 1 9 N Any Digit from 2 9 12345 9 Any Digit in the brackets in this example 1 2 3 4 5 6 7 8 9 Wildcard matches anything remaining i e _9011 Matches anything starting with 9011 excluding 9011 itself Wildcard causes the matching process to complete as soon as it can unambiguously determine that no other matches are possible For example the extension NXXXXXX would match normal 7 digit numbers while _1NXXNXXXXX would represent a three digit area code plus phone number proceeded by a one gt Use Trunk Describe which trunk should be used in this rule gt Strip Define how many digits should be removed from the dialstring Two samples for calling rule setting 1 Cut the first digit for all dialstring start with 9 and make outgoing call via port1 Calling Rule Name Out_9 Pattern _9 Use Trunk Ports 1 Strip 1 digits from the front In th2. IP 192 168 1 21 IP 192 168 1 31 IP 04 A IP 04 B he IP 192 168 1 30 Extensions 5001 AT 530 B IP 1927 168 1 20 Extensions 6001 AT 530 A Step 1 Set port forwarding in the router for IP04A The IPO4A is behind the router to register to IP04A via the internet you need to forward the IAX2 port in your router so all the packets received on the router WAN port 202 8 16 98 4569 will be forwarded to the IP04A 192 168 1 21 4569 Below is the setting page in a linksys router Applications Applications amp Gaming Port Triggering UPnP Forwarding Administration status amp Gaming Setup UPnP Forwarding UPnP Forwarding Application Ext Port IF Address Enabled UPnP Forwarding can be us to set up public services on FTP 21 O 21 192 168 1 0 E your network When users a the Internet make certan Teinet 23 23 192 168 1 0 requests on your network t Router can forward those SMTP 25 O 25 192 168 1 0 E requests to computers equip a to handle the requests lif fo DNS 53 O O 53 192 168 1 D E exampe you set the port number 80 HTTP to be TFTP 65 65 Bae j forwarded to FP Address 0o 12168140 O 132 168 1 2 then al HTTP P ay requests from outside users finger 79 O 79 1521681J0 Ei ed ee ane a a is recommended that the nee sd Oo 80 192 168 1199 k computer use static IP POP3 110
3. trunk 2 use the reverse method in IPO4B to register to IPO4A In above structure 1 AT 530A registers to IPO4A as an extension 6001 2 AT 530B registers to IPO4B as an extension 5001 3 All the extensions under IPO4A are in the format 6XXX 4 All the extensions under IPO4B are in the format 5XXX 5 Extensions under IP04A can make calls to extension under IPO4B use format 5XXX 6 Extensions under IP04B can make calls to extension under IP04A use format 6XXX 7 The two IP04 links each other via IAX2 trunk Step 1 Set up a extension 6005 in IPO4A Extension 6005 Phone number of this extension Name User_IPOAB Password 6005 IAX2 Log on password Caller ID 6005 Caller ID 36 www atcom cn Step 2 Set up an JAX trunk in IP04B to link to IPO4A via this User_IP04B extension In the page Trunks gt Add Voip Trunk Ansiog MOS T FESES iaan amiility a dahi Provider Nane To_IPO4A Hostname 192 168 1 21 Usernane 6005 Password 6005 Codecs First wlaw y Second a tew Third Fourth 6 726 Fifth Calleri FronDoasin i FroaUser insecure O Cancel Add Step 3 Set Calling Rule in IP04B all calls start with 6 will be sent to IP04A In the page Outgoing Calling Rules gt Add New Calling Rule nge Calling Rules Edit Calling Rule Calling Rule Nane Pattern 3 6 _ Send to Local Destination Destination Send this call through
4. Step 2 Add voip calling rule Go to page Outgoing Calling Rules 33 www atcom cn mies Calling Rules _ New CallingRule Calling Rule Name Out_VoIP Pattern _ 00 Send to Local Destination md dial Destination m Send this call through trunk Use Trunk Strip A digits from front and Prepend these digits before dialing iiss War eee k D fail over Trunk OoOo yoy y Strip digits from front and Prepend these digits before dialing 9 Cancel WW Save All calls start with 00 will be sent out via our voip service provider Step 3 Add this new calling rule to the dial plan1 All extensions which use dialplan1 are able to use the voipbuster service now 34 www atcom cn 3 5 Combine the IP04 with exist traditional PBX Introduce Assume that we already have a 3 8 3fxo 8extensions traditional PABX in our office and we want to add more pstn lines extensions or use voip solution in the exist solution We can combine the IP04 with the exist PABX solution use below structure Internet FXO PABN PSTN PSTN FXO PABX FNS Ci p04 FXOCIPO4 39PSTN X lite extension 6002 IiP 192 168 1 102 ext 6003 ext B04 IP 1927 168 1 103 1P 197 168 1 104 At tos a ECON or Combine the IP04 with exist traditional PBX 1 Connect the FXO port of the PABX to IP04 s FXS port so the PABX will be one of the IP04 s
5. Time Interval Name By day of week v to v Date Month a Time Entire Day gt By Days of a Month Start Time End Time Cancel F Update Name Name of the time Interval By day of week Define the day range by week for time interval By days of a month Define the day range by Month for time interval Time Entire day Define if the time interval is available for the whole day or only for the specified hours VV V V 15 www atcom cn 2 13 Incoming Calling Rules Create modify prioritize and delete incoming call rules for handling Incoming calls based on service provider and or the number called based on Time Intervals ao ana Calling Rules Hew Incoming Rule Trunk Time Interval Pattern x Destination x QO Cancel F Update none no TimeIntervals matched Goto User 7789 pra Trunk Ports 3 none no TimeIntervals matched Goto User 7469 prs Trunk Which service provide should use this trunk for its incoming calls Time Interval ranges of time that will be used in this rule Pattern Incoming call pattern VV V V Destination where should the incoming call is routed 16 www atcom cn 2 14 Voicemail General settings for voicemail function General Settings Email Settings for VoiceMails SMTP Settings Extension for checking messages 6050 Direct Voicemail Dial Max greeting in seconds 30 Dial 0
6. check this if the user is listed in the directory Call waiting enable disable Call waiting CTI Computer Telephony Integration allows access to 3rd party applications over Asterisk Manager Interface Is Agent check this if the user is available in call queue Pick up Group Specify the call pick up group Call Group Specify the call group for the user 2 8 Ring Groups Define the Ringroups to dial more than one extension simultaneously or to ring more than one phone sequentially tanage Ringo OO Hew RingGroup RingGroup Name Ring Group Hembers Available Users Extension for this ring group 6400 T806 SIP edwin ae T806 IAN2 edwin T969 SIP Gilly TIGICIANZ Gilly TT89 SIP peter T7S89 IAKZ peter T469 SIP Grace T469 IANZ2 Grace TOO1 SIP testi TOO1CIAN2 testi Ring Group Options Strategy Rine in Order Seconds to ring each member 20 If not answered Goto Hangup 10 www atcom cn 2 9 Music on Hold Customize audio tracks for different queues parked calls etc Ni E EET a A MOE Erte default Me p New MOH class fE ye Delete Manage Music On Hold Classes manage MOH class 7 default Choose file to Upload LA Upload Upload an 8 KHz Kono Music file Xmhl OOOOO_ITSCi i iSiF List of Sound Files 1000 miles wav y Delete acoustic escape wav y Delete beach carnival wav Delete
7. marked user has joined 19 www atcom cn 2 16 Follow Me FollowMe Preferences for Users FollowMe Options Jaaa i g O Enable Dasebie Music On Hold Class DialPlan DialPlani y t Configured anfi snrat t confi gui ed FE Destinations k Add Follorie Huaber Save gt Status Enable Disable FollowMe for this user gt Music On Hold class that the caller would hear while tracking the user gt DialPlan DialPlan that would be used for dialing the FollowMe numbers By default this would be the same dialplan as that of the user gt Destinations List of extensions numbers that would be dialed to reach the user during FollowMe Follow Ke N IE ER FollowMe Preferences for Users T Playback the incoming status message prior to starting the follow me step s T Record the caller s name so it can be announced to the callee on each step Playback the unreachable status message if we ve run out of steps to reach the or the callee has elected not to be reachable Save 20 www atcom cn 2 17 Directory Preferences for Dialing by Name Directory Directory Settings Dialing the Directory Extension would present to the caller a directory of users listed in the sytem telephone directory from which they can search by First or Last Name To add or remove a user from the system telephone directory edit the In Directory field of the user D
8. 168 1 103 1P 192 168 1 104 Internal calls in the same office At the beginning we need to add some extensions to make internal calls Each extension acts as an internal number There are three types of extensions we can add SIP IAX2 and ZAP Before set up the extensions we need to go to the Options gt General Preferences to set the user extensions range The default user extensions range is from 6000 6299 The extension 6000 is used for auto attendant so don t register on this extension General Preferences General Preferences Language Change Password Reboot Advanced Options Global OutBound crip Global Outdound CID Naas Operator Extension ei Ring Timeout 20 Extension preferences Conference Extensions 6300 to 6399 VoiceMenu Extensions 7000 to 7100 RingGroup Extensions 6400 to 6499 Queue Extensions 6500 to 6599 VYoiceMail Group Extensions 6600 to 6699 Reset to defaults Save User Extensions 6000 to 6299 Then go to page Dial Plan gt Create New Dialplan to create a default Dial Plan 29 www atcom cn Then go to page Users gt Create New User to create the extensions 6001 General s f Ta T Extension 600 D wane Edwin D DialPlan O D CallerId 6001 D OutBound CallerId 7 Enable Voicensil for this User D VoiceNail Access PIN code O Ensil Address 6001 D _Testmaloey Se s
9. FXS extensions and all the extensions under pabx can use all the fxs functions from IP04 the functions include make calls to the ip04 s other extensions make calls use the ip04 voip trunk make PSTN via ip04 s PSTN trunk 2 Connect the FXS port for the PABX to IP04 s FXO port So the IP04 will be one of the PABX extensions and all the extensions under pabx can use all the fxs functions from pabx the functions include make calls to the pabx s other extensions make PSTN via pabx s PSTN trunk 35 www atcom cn 3 6 Intercommunication between two IPXX Introduce This application note shows how to link two IP04 in different location With this function we can link branches together with IP04 Same method can be used when connect more than 2 IP04 in different branches 3 6 1 Link two IPOAs in the same network The simplest case to link two IPxx together is in the same network We start from this and then try to expand to different network We use IP04 here same method for other IPXXs products Below is the structure of how to link two IP04s in the same LAN AX trunk IP 192 168 1 21 IP 192 168 1 31 IP 04 A IP 04 B IP 192 168 1 30 Extensions 5001 AT 530 B IP 197 168 1 20 Extensions 6001 AT 530 A The method of connecting two IP04s in different location is 1 register the IPO4A as an extension in IP04B via IAX2 trunk so the extensions in IP04A can make calls to IP04B s extensions via this special
10. Known Issue 4 1 Call Detail Record You need to use firefox to open the CDR it doesn t work in the IE 4 2 FXO FXS port doesn t work in special cases If you find the FXO FXS port doesn t work Please check if below cases 4 2 1 Case 1 modification of tonezone causes corrupt file problem In the ssh connection mode you can see the fxo and fxs are properly load by use the command dmesg And the LEDs are on for the ports But in the asterisk CLI command zap show status is invalid Then please check the file etc zaptel conf you may see the file is error for example shows 2 loadzone nl defaultzone nl fxoks 1 2 fxsks 3 4 in this case remove the 2 from the file and reboot the devices When you modify the tonezone in the GUI this error may happen 4 3 Busy detect issue In the GUI trunk page if you edit the trunk setting There is a busy pattern parameters Please make sure that the busy pattern are set to correct parameter Otherwise you may have the busy detect issue when the PSTN side hung up the call the line doesn t hung up correctly To fix this issue you can modify the etc asterisk users conf file via file editor in GUI or ssh connection Find the line begin with busypattern add at the beginning on the line to comment the busypattern parameters 43
11. TimeZone GMT 8 00 hours Beijing Perth Singapore Hong Kong Chongqing Urumqi Taipei v Save 23 www atcom cn 2 24 Advance Options Call Detail Records Shows the call details CDR viewer lt lt prev next gt gt Viewing 1 7 of T View 10 most recent first account Source Destination et Sle anel HEt el Last app Last data Start time Answer Time Ind Time Duration woe Disposition ala f An flars Unigue ID 2306 a306 fZap 3 1 raara ea E pou hte 05 57 00 05 57 02 2 02 azi PET Gro 47 2 3806 3306 zap 1 1 BackGround demo congrats m 3306 s306 Zapf ee ONET peat Jae ansman fnocomemnaTIonj1167609653 0 E 19 00 53 19 00 54 19 01 05 Manager CDR Files 2 25 Advance Options Firmware Update Update the firmware of your device Update Appliance Firmware Download image from a HTTP URL TFIP Server TFIP Server 192 168 1 234 mos File Name ulmage nd5 igs O Reset Configs HTTP and TFTP update is available for the firmware update gt TFTP server TFTP server which include the update firmware gt File Name name of the new firmware please make sure that you are using a md5 firmware for the updating gt Reset Configs enable this if you want to reset the networking and asterisk configs after upgrade 2 26 Advance Options File Editor Here you can modify the asterisk configure files directly 24 www atcom cn File Editor Config Files
12. adsi conf k agents conf applyzap conf asterisk conf cdr conf edr custom conf cdr manager conf codecs conf dnsmer conf dundi conf enum cont eztconfig conf extensions conf features conf Eui_confighw conf Buipreferences conf http conf iaz conf iazprov conf Note Please make sure you know what the meaning in the files before trying to modify these files 25 www atcom cn 3 Application notes 3 1 Install hardware The IPXX series IP PBXs use interchangeable FXO FXS modules The FXO and FXS signaling use different modules Below are the available modules for the system Avaiable TE Photo Description Suitable Products Products ia one channel FXO module IPO1 POZ P0O4 and IPOS one channel Fas module IP01 P02 IPO4 and IPOS a se dual channel FXO module IFO and IPOS dual channel Fas module IP02 and IPOS FXO use to connect to the PSTN line FXS use to connect to the analog phone Install illustration 1 IP01 wai one FXO 26 www atcom cn 3 2 Different methods to access the IPxx There are several ways to access the IPXX series products Different ways has different usage The web SSH telnet accesses are base on network connection and the console port access is via the console cable which allows you to access the devices on a lower level gt Web access It is the most common way to access the IPxx Most settings can be done through the we
13. for Operator v Nessage Options Maximum messages per folder 25 l Max message time 2 minutes x Min message time 5 seconds Playback Options Say message Caller ID M Say message duration Play envelope Allow users to review E Q Cancel W Save General Voicemail Settings gt gt Yy Extension for checking Message This option i e 2345 defines the extension that Users call in order to access their voicemail accounts Direct VoiceMail Dial Check this to enable direct voicemail dial For instance if John s extension is 6001 you would be able to directly dial into John s voicemailbox by dialing 6001 to leave him a message Max Greeting Set the maximum number of seconds for a User s voicemail greeting Dail O for Operator Enable Callers to exit the voicemail application and connect to an operator extension The operator extension must be defined from the Options panel Maximum messages per folder This select box sets the maximum number of messages that a user may have in any of their folders Maximum Message Time This select box sets the maximum duration of a voicemail message in seconds Message recording will not occur for times greater than this amount Minimum message Time This select box sets the minimum duration of a voicemail message in seconds Messages below this threshold will be automatically deleted Say Message Caller ID If this option is
14. from M_CALLERID Hessage Hew Voicemil from 4 M_CALLERID Cancel Load Defaults fl Save 2 Configure your SMTP server SETP Settings for Voicemail notification Emails N e General Settings Email Settings for VoiceMails SMTP Settings Smtp server mail atcom com cn Port 25 9 Cancel F Save If your SSMTP server needs Authentication you need to put your username and password in the file ssmtp conf via SSH access as below etc ssmtp ssmtp conf root edwin atcom com cn mailbox account mailhub mail atcom com cn smtp server rewriteDomain atcom com cn hostname edwin atcom com cn AuthUser edwin atcom com cn mailbox account AuthPass xxxxxXxx mailbox password AuthMethod LOGIN FromLineOverride YES 3 Enable the voicemail for users and put the corresponding Email Address Edit User Extension 6001 X General Extension nane Alice DialPlan DialPlani CallerID 6001 outBound CallerID V Enable Voicemail for this User VoiceMail Access PIN code 6001 Mailbox 6001 Email Address Alice atcom com 41 www atcom cn 3 8 Call Features 3 8 1 Call Pick Up The default feature code to pick up a call is 8 If there is an incoming call for a user in the call group 2 then the users in the pickup group 2 is able to pick up this call by dialing 8 The pickup group and call group can be defined in the use
15. so far since they have almost the same software and structure so we will use IP04 as the demo unit on this article Same method is available to the IP XX series products IP01 IP02 IP04 IP08 amp IP BRI The IP XX series PBXs are open source embedded IP PBX systems They run uClinux and Asterisk and support rich IP PBX features They have big advantages on its inherent open source software structure and ultra low power consumption lt 5 watt in idle state environment friendly Below are the difference between the IP01 IP04 and IP08 platform Model No RJ45 TEL Ports RS232 Port Others Size mm IP 01 1 1 x fxo fxs 1 fix RS232 port 100 x 100 x 28 IP02 2 2 x fxo fxs 1 RS232 module 100 x 100 x 28 IP 04 1 4 x fxo fxs 1 RS232 module MMC 225x 120x30 IP 08 2 8 x fxo fxs 1 RS232 module MMC USB 225x120 x30 IP BRI il 4x BRITE 1 fix RS232 port MMC 225x 120x30 4 www atcom cn 2 Configure the device via GUI 2 1 Access the GUI The IP04 GUI is immigrated from Asterisk Now 2 0 version The default IP for IP04 is 192 168 1 100 Put the default ip in your web browser and it will redirect to the setting page of IP04 the default password for the web access is Username admin Password mysecret If you can t access the IP04 please check if you have connect the RJ45 cable to the WAN port and your computer is in the same network 192 168 1 xxx as the IP04 Note the recommend web browser of IP04 is Firefox 2 2 System Statu
16. trunk Use Trunk to_1Foda v Strip D O digits from front and Prepend these digits before dialing Use FailOver Trunk fail over Trunk Strip digits from front and Prepend these digits before dialing Cancel WV Save Step 4 Add this new calling rule Out_IP04A to the exist dial plan In the page DialPlan gt Edit DialPlan1 Planz Edit DialPlan DialPlan Name DialPlanl Include Gutgoing Calling ra Include Local Contexts V default parkedcalls conferences ringgroups voicemerms queues voicemailgroups directory pagegroups page_an_extengzion CJ Cancel E Save Active the change and apply the test 37 www atcom cn 1 Register an IP phone AT 530B to IP04B with 5001 extension 2 Register an IP phone AT 530A to IP04A with 6001 extension 3 Use 5001 to dial 6001 And you can see 6001 is ringing and you can pick up the calls Above is the way to router IP04B s call to IPO4A the method to link IPO4A to IPO4B is the same as above 38 www atcom cn 3 6 2 Link two IPOA4s in different location The generally environment for two ip04 in different location is two IP04 are both behind router and using the private IP Since the IP04 doesn t have the public IP we need to do port forwarding in the router and make IP04 is reachable to others IAX trunk ey IP 202 8 16 98 Office A SS WAN IP 115 9 30 54 A Office B
17. 009feb04_103739 Feb 04 2009 Download from Unit Restore Previous Config y Delete Note Restored the backup won t take effect on the network setting You need to modify the network setting in the GUI and save reboot 2 21 Active Channels Displays current Active Channels on the PBX with the options to Hangup or Transfer Channel HNanagement Refresh Now Active Channels 1 Refreshing Active Channels in 1 Seconds Channel State Seconds Application SIP 6002 011 2580 Up 4 Voicemail fARGL u Transfer Hangup 22 www atcom cn 2 22 Advance Options In the Options Panel choose Advanced Options gt show Advanced Options then the advanced options will be showed in the left menu hdvanc ed Options Reboot General Preferences Language Change Password Advanced Options Hide Advanced Options The advances options include Call Detail Records Active Channels Bulk Add File Editor Asterisk CLI AX Settings SIP Settings Network Settings VV VV VV VV WV Firmware update 2 23 Advance Options Network Settings Network and time zone settings Wetworking setting ethO Interface DHCP no Hostname ipOd Domain IP address 192 168 1 230 Subnet mask 255 255 255 0 Gateway 192 168 1 1 DNS 192 168 1 1 NTP pool ntp org YLAN Interface for EthO VLAN Vlan number Vlan IP address Vlan Subnet mask Vlan Gateway System TimeZone
18. 110 1 241681l0 o NNTP 119 113 192 163 1 0 E SNMP 161 oO 1161 192 168 10 O j s i r a used to configure this throug sth 2020 22 192 168 1 235 UPnP communication Be sure a that you enter a valid F httpl s080 80 192 168 129 E Address You may need to estabish a static P address httpe 8090 80 192 168 1209 with your ISP in order to property run an internet se O 4563 192 168 121 For added security 45693 1921681121 More 39 www atcom cn Step 2 Set up the service provider and calling rule in IPO4B to make it register to IPO4A This method is almost the same as above EXCEPT you need to use the 202 8 16 98 as the service provider instead of 192 168 1 21 Step 3 Use the same method do port forwarding in routerB for IPO4B Your public address from network provider maybe a dynamic ip which will be changed periodically To overcome the problem of dynamic ip you may need to use the DDNS service for more info please google the internet 40 www atcom cn 3 7 Voicemail to Email Configure example The IP04 will send a notify Email to your mail box when you have set up the Voicemail to Email function 1 Set up the preference Yoicemail Email alert preferences General Settings Email Settings for VoiceMails SMTP Settings _ Send messages by e mail only Attach recordings to e mail From edwin atcom com cn Subject You ve got new Voicemail
19. aller exits Enable caller menu i Announce callers Quiet Mode ae Wait for marked user O Cancel Update gt Extension This is the number dialed to reach this Conference Bridge gt Marked Admin user Extension If the conference bridge is to have marked users or admin users then those users should enter the conference bridge using a separate extension Admin conference users can lock and unlock the conference and can kick the most recent conference participant Marked users are special users whose entrance and exit if the Wait for Marked user or Close conference when last marked user exits can either begin or end the conference altogether gt Pin Code set an optional pin code Ex 1234 that must be entered in order to access the Conference Bridge gt Administrator PIN Code Defining this option sets a PIN for Conference Administrators gt Play Hold Music for First Caller Checking this option causes Asterisk to play Hold Music to the first user in a conference until another user has joined the same conference gt Enable Caller Menu Checking this option allows a user to access the Conference Bridge menu by pressing the Asterisk key on their dialpad gt Announce Callers Checking this option announces to all Bridge participants the joining of any other participants gt Quiet Mode Do not play enter leave sounds gt Wait for Marked User Prevent conference participants from hearing each other until the
20. an3 port3_8 port3_9 port3_17909 default parkedcalls conferences ringgroups x JlalPlan3 volcemenus queues voicemailgroups directory pagegroups page_an_extension 0 DialPland port4_ amp port4_9 port4_17909 default parkedcalls conferences ringgroups y JlalPlan voicemenus queues voicemailgroups directory pagegroups page_an_extension 8 www atcom cn 2 7 Users This page allow administrator to create extensions for every user er Extensions on PBA Create New User General Extension 6000 none DialPlan DialPlani y CallerID 6000 ut Bound CallerID L Enable Voicemail for this User VoiceMail Access PIN code Mailbox 6000 Email Address brek Technology V SIP m Tax Analog Station None flash 750 rxflash OF 1250 Codec Preference First wlaw Second csm Third None Fourth Hone E Fifth None VoIP Settings MAC Address Line Number 1 y SIP IAX Password NAT M Can Reinvite C DIMF Mode RFC2833 insecure no w Other Options F 3 Vay Calling Fl In Directory Hl call Waiting A cr DM P ts ager Pickup Group x Call Group 1 y O Cancel Update General Extension The number you can dial to reach this user Name CallerID name of the user CallerID The CID string when you dial to other internal users O
21. b interface Simply put the device s IP address in your web browser better use Firefox and enter the username and password to access the device The web access username password is admin mysecret gt SSH access This is the advance way to access the device you can use the putty software to access the device in the ssh access you can access the Linux directly and do more advance linux setting and debug The SSH user password is root uClinux gt Telnet access The telnet access is similar with the SSH access but it is not suggested because it is not as convenient as SSH access gt Console access The console access is used mainly for develope purpose or in the case when network is down Below are the connections pictures for console access For IP01 amp IP02 Below is the console port setting to access the IPXX Running the Hyper Terminal or Minicom in your computer to connect the IP04 the setting of the console port should be 21 www atcom cn Bits per second to 115200 Data bits 8 Parity None Stop bits 1 Flow control None 28 www atcom cn 3 3 Make free internal calls Making internal calls are the base requirement for a telephony system Below are the settings for this usage It is base on IP04 but setting is the same in other IPXX products System Setup X lite extension 6001 IP 192 168 1 101 LAN Local Area Network IP04 192 168 1 100 extension 6003 extension 6004 IP 192
22. dancing in space waw y Delete df sweating wav y Delete guitarra in bb minor waw y Delete in waiting wav MDelete lift me up wav Delete night train gorodetskiy wav MDelete streaming from my heart wav Delete 11 www atcom cn 2 10 Call Queues Call queues allow calls to be sequenced to one or more agents Hew Queue Extension 6500 Name Strategy ringa Music On Hold default y LeaveWhenEmpty Ho p Tare er Queue Options TimeOut 15 Wrapup Time 15 Max Len 0 E Auto Fill Auto Pause Report Hold Time KeyPress Events None x Agents You do not have any users defined as agents click here to manage users Update gt Extension This option defines the numbered extension that may be dialed to reach this Queue gt Name This option defines a name for this Queue i e Sales Name is a label to help you See this queue in the queue list gt Strategy This option sets the Ringing Strategy for this Queue The options are RingAll Ring All available Agents simultaneously until one answers RoundRobin Take turns ringing each available Agent LeastRecent Ring the Agent which was least recently called FewestCalls Ring the Agent with the fewest completed calls Random Ring a Random Agent RRmemory RoundRobin with Memory Remembers where it left off in the last ring pass gt Agents This selection shows all Users defined as Agents in thei
23. dd New DialPlan Go to page Dial Plans gt Crear New Dial Plans Create New DialPlan DialPlan Name DialPlanl Include Outgoing Calling VIOUT_PSTN Rules Include Local Contexts V default Vlparkedcalls Vlconferences Vlringgroups Wlvoicemenus Wlqueues Wlvoicemailgroups directory V pagegroups page_an_extension S Cancel Save Set the DialPlan1 to default dial plan so every new extension will use this dialplan in default Then you can use your extensions to dial out via the port1 and portz2 32 www atcom cn 3 4 2 VoIP trunking Via the voip trunking we can dial call via the voip service to reduce our cost when making international calls Internet PSTN Connect to PSTN i network via PRO part A t extension 6001 IP 192 168 1 107 Z LAN Le l Ar ee tear a IPO 192 168 1100 X lite extension e007 F 192 165 1 102 extension BOOS extension PA IP 192 168 1 103 IF 192 168 1 104 Outgoing Calls Network Step 1 Add Voip trunks Go to page Trunks gt Voip Trunks gt Add New Sip trunks Hanage SIP amp IAK trunks Analog TURS LELA E aay 0 eM 8 9 wails A Type Provider Name D voipbuster Hostname sip voipbuster com Username aniceman W Add We use the Voipbuster as our voip service provider here
24. enabled the Caller ID of the party that left the message will be played back before the voicemail message begins playing Say Message Duration in minutes If this option is set the duration of the message in mintues will be played back before the voicemail message begins playing Allow callers to Review Checking this option allows the caller to review their message before it is submitted as a new voicemail message Play Envelope Turn on off playing introductions about each message when accessing them from the voicemail application 17 www atcom cn Voicemail to Email with this function configured when there is a new voicemail for users the ip04 will automatically send the voicemail to the user s email address set in the user s profile Voicemail to Email Preference PLrererences General Settings Email Settings for VoiceMails SMTP Settings C Send Messages by e mail only Attach recordings to e mail From asterisk yourcompany null Subject New voicemail from VM_CALLERID for V M_MAII Message Hello VM_WAME you received a message lasting VM_DUR at V _DATE from CS VM_CALLERID This is message VM_NSGHUM in your voicemail Inbox CJ Cancel Save Template Variables t TAB BiVN_NANE Recipient s firstname and lastname SivVM_DUR The duration of the voicemail message BiVM_MAILBOR The recipient s extension Bi M_CALLERID The caller id of the person who left the mes
25. g P C E EITE ead r Channels 1 V2 Trunk Wane PSTN Advanced Options Busy Detection J Yes y Busy Count Busy Pattern 500 500 Ring Timeout A Answer on Hangup on Polarity Switch Polarity Switch Call Progress Progress Zone Use CallerID Caller ID Start CallerIp D s Received y Pulse Dial CID Signalling Bell vsa mailbox Flash Timing z 750 Receive Flash Timing 2 F Add Note The port1 and port2 of IP04 are slotted with FXO modules Always click Apply Changes in the right top corner when you do some changes Step 2 Create Outgoing Calling Rules 31 www atcom cn Go to page Outgoing Calling Rules Hew CallingRule utg Calling Rule Name OUT_PSTN a Pattern O _9 Send to Local Destination Destination Send this call through trunk Use Trunk Strip a digits from front and Prepend these digits before dialing Use Faildver Trunk fail over Trunk Serip digits from front and Prepend these digits before dialing 9 Cancel Save The pattern _9 and strip 1 digits means all calls start with 9 will be cut the first digit and sent out via this rule for example if you dial 983018049 the ip04 will send 83018049 to porti or portz Step 3 A
26. ious Digita Advanced Settings Module Name wetdm24xxp Opermode gt B us a a law override O ulaw fxs honor mode E apply opermode to fzo modules only boostringer O normals fastringer gt O lowpower A Co ring detect A standara MWI mode O Hone Mancel Changes Update Settings Note Hover on the i and you can see the comment of every settings 2 4 Trunks Trunks are used to make outbound call to the real world There are different trunks we can set here tanase Anator trames Service Providers VOIP Trunks T1 E1 BRI Trunks Analog Ports Ports 1 Edit MDelete Ports 2 Edit Delete Ports 3 Edit X Delete Ports 4 Edit Delete l We have put the IP04 with four FXO ports so there are four analog trunks in this setting page VoIP trunks SIP amp IAX2 are also available in the IPXX Manage SIP amp TAX trunks 0 0O OO Analog True L E uaa C A N S ES i s e Type SIP Provider Name voipbuster Hostname Sip voipbuster com Username test Password More info about how to set up the trunks please refer the application notes 6 www atcom cn 2 5 Outgoing Calling Rules Outgoing Calling Rules defines the calling permission sand the routing rules when making calls JHanage Calling Rules eee Hew CallingRule Calling Rule Name E Pattern Send to Local Destination Destination l
27. irectory Extension Also read the extension number O Use first name instead of last name FI Save gt Directory Extension Extension to dial for accessing the Name Directory gt Read Extension number In addition to the name also read the extension number to the caller before presenting dialing options gt Use first Name instead of Last Name Allow the caller to enter the first name of a user in the directory instead of using the last name 2 18 Voicemail Group Define VoiceMail Groups to leave a voicemail message for a group of users by dialing an extension Hew oice Kail Group VYoiceMail Group s Extension AAO Label User MailBoxes goo1 C lgon2 CJ Cancel Save 2 19 Voice Menu prompts Record or Upload custom VoiceMenu prompts Custom Voice Nenu Prompts List of Custom Voice Menu Prompts Upload a Voice Menu prompt No custom Voice Henu prompts found You can record a new VYoiceMenu Prompt by clicking on the Record a new Voice Menu prompt or click on the Upload a Voice Menu prompt button to upload a custom voice menu 21 www atcom cn 2 20 BackUp Backup or Restore the configure files f Baarap Restore contigarataons O U Manage Configuration Backups s Create New Backup List of Previous Configuration Backups 1 backup_2009feb04_ 103930 Feb 04 2009 Download from Unit Restore Previous Config Delete 2 backup_2
28. is case if you dial 983018806 in your extension 83018806 will be sent via port1 2 Cut the first digit for all dialstring start with 0 and prepend 86 then dial via voipbuster trunk T www atcom cn Calling Rule Name Out_voipbuster Pattern _0 Use Trunk Voipbuster Strip 1 digits from the front And prepend these digits 86 In this case if you dial 075583018806 in your extension 8675583018806 will be sent via voipbuster trunk 2 6 Dial Plans A Dial Plan is a collection of Outgoing Call Rules Dial Plans are assigned to Users to specify the dialing permissions they have For example you might have one Dial Plan for local calling that only permits users of that Dial Plan to dial local numbers via the local outgoing calling rule Another user may be permitted to dial long distance numbers and so would have a Dial Plan that includes both the local and longdistance outgoing calling rules DialPlans Edit DialPlan DialPlan Name DialPlanl Include Outgoing Calling V port1 Mlporti_8 M port1_17909 Jport2_8 Clport2_9 port2_17909 port3_8 port3_9 Jport3_17909 C Rules porta 8 port4_9 port4_17909 Include Local Contexts V default V parkedcalls V conferences V ringgroups V yoicemenus V queues V vyoicemailgroups V directory pagegroups m page_an_extension QO Cancel Save yolcemenus queues voicemailgroups directory pagegroups page_an_extension C DialPl
29. n to the Specific Queue This is an extension that all the Agents can Call to Login to their specified Queues Agent Callback Login Extension Extension to be dialed for the Agents to Login to the Queues they are apart of Same as Agent Login Extension except you do not have to remain on the line 13 www atcom cn 2 11 Voice Menu IVR Menus allow for more efficient routing of calls from incoming callers Also known as IVR Interactive Voice Response menus or Digital Receptionist Nanage Voice Menus Create New YoiceNenu Name Extension 7000 E Allow Dialing Other Extensions Actions Add new Step Select an Option E Allow KeyPress Events gt Name Name ofthis Voice Menu gt Extension If you want this Voicemenu to be accessible by dialing an extension then enter that extension number gt Dial other Extensions Is the caller allowed to dial extensions other than the ones explicitly defined gt Actions A sequence of actions performed when a call enters the menu gt Add anew step Add additional steps performed during the menu gt Keypress Events Allow key press events will cause the system to listen for DTMF input from the caller and define the actions that occur when a user presses the corresponding digit 14 www atcom cn 2 12 Time Intervals Time Intervals are defined ranges of time that will be used by call routing features Tine me Hew Time Interval
30. no limit When the limit has been reached a caller will hear a busy tone and advance to the next calling rule after attempting to enter the queue AutoFill Defining this option causes the Queue when multiple calls are in it at the same time to push them to Agents simultaneously Thus instead of completing one call to an Agent at a time the Queue will complete as many calls simultaneously to the available Agents AutoPause Enabling this option pauses an agent if they fail to answer a call This means that the agent is still logged into the queue but they will not receive calls from the queue Once paused an agent can unpause by logging into the queue using the regular agent login extension Report Hold Time Enabling this option causes Asterisk to report to the Agent the hold time of the caller before the caller is connected to the Agent KeyPress Events If a caller presses a key while waiting in the queue this setting selects which voice menu should process the key press Queues Agent Login Settings Agent Login Extension Agent Callback Login Extension To logout of Agent Login Hangup your phone To Logout of Agent Callback Login Dial the Same extension used to login specify your Agent Logout extension and password when prompted and hit when asked for your callback extension This will successfully log you out of all queues you are a part of Agent Login Extension Extension to be dialed for the Agents to Logi
31. pr R IY Analog Station mae D man D 7H rxfiash D 1250 citac Preferente First wiwe o Second cm e Third mee Fourth mee Fifth i meme y VoIP Settings r Address D Line Mader 1 D LineKeys 3 2 D swr tat Password 6001 JD mat Z D can Reinvite D DNF Mode wera D insecure 2e t Other Options Vay Calling D In Directory t Call Waiting D CTI D Ie Agent D Pickup Group l e Qoa trte Use the same method to add the extensions 6002 6003 and 6004 In our system picture we use softphone x lite to register on 6001 and 6002 Use AT 530 IP phone to register on the extensions 6003 and 6004 Then these four extensions can communicate with each other use the numbers 6001 6002 6003 and 6004 30 www atcom cn 3 4 Make outbound calls to PSTN There are many kinds of trunking you can use to make outgoing calls It includes Analog FXO trunk Digital E1 T1 BRI Trunk SIP trunk AX trunk etc 3 4 1 Analog FXO trunking For the IP01 04 08 you can install FXO module and use the FXO trunking to make outgoing call via your local PSTN line The set up is as per below PSTN Connect to PSTh V IPF 192 168 1 101 lite extensian 6002 IP 192 168 1 102 IPO4 192 168 1 100 extension 6003 extension 6004 IF 192 168 1 103 IP 192 168 1 104 Outgoing Call To PSTN Network Step 1 Create FXO trunk Go to page Trunks gt Add New Analog Trunk Nanage Analog trunks _ Analo
32. r User conf Checking a User here makes them a member of the current Queue gt Music On Hold Select the Music on Hold Class for this Queue Music on Hold classes can be managed from the Music On Hold panel gt LeaveWhenEmpty This option controls whether callers already on hold are forced out of a queue that has no agents There are three options Yes Callers are forced out of a queue when no agents are logged in No Callers will remain in a queue with no agents Strict Callers are forced out of a queue with no agents logged in or if all logged in agents are unavailable gt Join Empty This option controls whether callers can join a call queue that has no agents There are three options Yes Callers can join a call queue with no agents or only unavailable agents No Callers cannot join a queue with no agents Strict Callers cannot join a queue with no agents or if all agents are unavailable Queue Options 12 www atcom cn gt gt Yy Yy gt gt Timeout How many seconds an Agent s phone will ring before the Queue tries to ring the next Agent Wrapup Time How many seconds after the completion of a call an Agent will have before the Queue can ring them with a new call The default is 0 which is no delay MaxLen How many calls can be queued at once This count does not include calls that have been connected with Agents it only includes calls that have not yet been connected Default is 0 which is
33. r setting page 3 8 2 Call Park The default call park extension is 700 The call park features code can be found in the file etc asterisk features conf Park a call on eye beam Press XFER button then it will shows Enter Number press XFER Enter the default park extension 700 and press the XFER button again The call will be parked to the extension range 701 720 dial 701 720 to get the parked call in another extension 3 9 Cron Cron is the name of program that enables Linux users to execute commands or scripts groups of commands automatically at a specified time date It is normally used for sys admin commands like makewhatis which builds a search database for the man k command or for running a backup script but can be used for anything You can start the cron service for IP04 by etc init d cron enable etc init d cron start The crontab file locats in etc config More info for how to use cron in linux please search in the internet 3 10 Backup and restore file 1 In the page back up to create new back up 2 Youcan restore this back up file to your computer 3 to transfer this back up file to another IPxx 1 Putthe file in your tftp server 2 Use putty to connect to the IP04 via SSH Run below commands 3 cd persistent var lib asterisk gui_backups 4 tftp g r YOUR_BACKUP_FILE YOUR_TFTTP_IP 5 goto the backup page and select the file to restore Reboot the device 42 www atcom cn 4
34. s When you have entered the IP04 setting page the system status will be showed and you can see the system status as below System Status stem otatus Uptime 15 38 26 up 22 44 load average 0 00 0 00 0 00 ly Trunks Trunk Username Port Hostnane IP Ports 1 Ports 1 Ports 2 Ports 2 Ports 3 Ports 3 Ports 4 Ports 4 Conference Rooms Parked Calls 6060 No Parked Calls Not in use Extensions v Free P Busy a Un vailable v Ringing Extension Name Label Status Type 7001 test1 Messages O SIP IAX User 7002 7002 Messages 0 0 SIP IAX User 7m9 wells Messages 0 0 SIP IAX User 7469 Grace Messages 0 0 SIPZIAX User 7569 Forrest Messages O 0 SIPZIAX User 7789 peter Messages 0 0 SIP IAX User 7806 edwin Messages SIPZIAX User 7969 Gilly Messages O 0 SIP TAX User 6090 Support Call Queue 6000 test Voice Menu 6050 Check Yoicemails YoiceMailMain No Extension assigned Dial by Hames Directory 5 www atcom cn 2 3 Configure Hardware The Configure Hardware page lists the available telephony ports in your system You can configure the hardware to comply with your local telephony environment Analog Hardware a FES Ports 1 2 3 4 Edit Tone Eegion Please choose your country or your nearest i neighboring country for default Tones Ez dialtone busy tone FXO Ports Tone Region ring tone etc Reset al s Information Prev
35. s E sone eavnee aout acaesqucseuaseuties E E 22 Active Channels ccc dine daaicas cnc setcacdaauvsae sasua es taaesee vais eciace asus AA a AA eas 22 Advance OPHONS rirerire iaa a ds a E A a 23 Advance Options Network Settings ccccccccssccccccccceccceceeseseesssssssseeeeeeeeeseeeeeeseeeeeeeas 23 Advance Options Call Detail Records ccc cccccccccccceeceeessssssessnnsaaeeeeeeeeeeeeeeeesseeeeeegs 24 Advance Options Firmware Update cccccccccccccccscceecceeeeseseesssssssseeeeeeeeeeeeeeeeeeeeseeeeas 24 Advance Options File Editor cseesssccsscceeecessseessssssncnnenscccccsececeeesessssesssssnaaes 24 Application NOTES i254 Gerace tetas a a a hae iG ata a see eek ee oso weet re 26 Hetal Tear wy alk Caz cts Tn R A A AA E AA ENE 26 Different methods to access the IPXX 00000000oneresoessssssssssssssssssoeeeeecessssssssssssssreeeeeceesessss 2a Make free internal calls is saidadseosccecdescadecs eats vvegugiestndeswde tidliiaandds sadsgslagudanduaaddtaietacdancneneesngaes 29 Make outbound calls to PSTN 2 00 00 cscsesssssessssseeeccccccceesesessesessssasssssccenscccccecesseesees 31 Analogy EXO MUNKIN O raoa essa aac ce nceccich ocioass oes Gein ceausawide coca A 31 VOIP Cr 9 sriorenerrae a sac edsaiteee ees ee es 33 Combine the IP04 with exist traditional PBX cc ccccccccceeecceeeseeesesssssceeeeeeeeeeeeeees 35 Intercommunication between two IPXX ccccsssssssscceeeeeeeeeceeceeess
36. sage SiVM_MSGNUM The message number in your mailbox BiVM_DATE The date and time the message was left gt Send messages by e mail only If this option is set then voicemails will not be checkable using a Phone Messages will be sent via e mail only Note You need to have an smtp server configured for this functionality gt Attach recording to e mail This option defines whether or not voicemails are sent to the Users e mail addresses as attachments Note You need to have an smtp server configured for this functionality SMTP server setting General Settings Email Settings for VoiceMails SMTP Settings Smtp server Fort CJ Cancel F Save gt SMTP Server The IP address or hostname of an SMTP server that your Astfin box may connect to without authentication in order to send e mail notifications of your voicemails i e mail yourcompany com gt Port The port number on which the SMTP server is running generally port 25 Note for Setup example for the Voicemail to Email please refer the application note 18 www atcom cn 2 15 Conferencing MeetMe conference bridging allows quick ad hoc conferences with or without security Manage Conference Rooms li isik New Conference Bridge Extension 6300 Marked Admin user Extension Password Options Pin Code Admin PinCode Conference Room Options a Play hold music for first E Close conference when last marked user c
37. ssseesssssaeeeeeeeeeeeeeees 36 Link two IPO4s in the same netWwork seccssccccccccceeesesssessessssnnnecesccceceereeseses 36 Link two IPOA4s in different location ccc ccceeeessssccecceeeeeeececeeeeeeeeeessssssaaeeeeeeseeeeeees 39 Voicemail to Email Configure exaMmple ccccccccccccccccccceeeceeeesesseesssssseeeeeeeeeeeeeeeeseeseeeeeas 4 Call FEITU Sie ties ob tact atti eb aad thee a t al es creel ol a ao nmtelah at ctl 42 CAMP YC Ee UP eaa E acess sac E A Gad na gear ios EE ee cso I 42 www atcom cn 3 8 2 CAM PANS eean A tices tne acta a eiaa aint sl eotaiieds 42 S HTM eset aac a T 42 3 10 Backup and restore Met 025 ascees2e onne nS sds en E E A Se 42 A KNOWN ISS UC aE EEEN EEE ENEE ENE E rere er erry terry 43 AA Gall Detail Record resns E a lean 43 4 2 FXO FXS port doesn t work in special CaS S ccccccccecccsseeeeeeseesessncceeeeeeeeeeeeeeeseeseeeeeas 43 4 2 1 Case 1 modification of tonezone causes corrupt file problem ccccccccccceeeeees 43 Ao 1826 EISS Eaa ERCP eer a caer A a er Ra oe 43 3 www atcom cn 1 Introduce This Article This article is the user manual for the IP XX series products It also includes the application notes for how to use ATCOM products to build a telephony system for small office Through this article we hope that users can build the IP telephony system via IP XX series products The IP XX series PBXs include IP01 IP02 IP04 IP08 and IPBRI
38. utbound CallerID Specify the public CallerID for outbound calls it is only available when your digital or voip provider support this feature VoiceMail Voicemail Access PIN code The password of your voicemail box Email address The email address for the voicemail to email function Technology SIP enable this option so the extension can be a SIP device IAX2 enable this option so the extension can be an IAX2 device Analog Station If you have analog FXS ports in your IP04 you can select the port here for your extension flash rxflash flash parameter for the users Codec preference specify the preference codec for the users Voip Setting MAC address used for polycom phone provisioning Line Number used for polycom phone provisioning Linekeys used for polycom phone provisioning SIP IAX2 password user password for SIP IAX2 registration NAT enable this when you use the IP04 in public network and the sip devices are in 9 www atcom cn private network Can Reinvite enable this and the IP04 will try to negotiate the endpoints to route the media string directly not through IP04 This can reduce the CPU load of the IP04 and you will get better voice performance because the media string are sent directly from endpoint to endpoint DTMF mode DTMF uses on conversation the RFC2833 is the most common Insecure method of authentication Other Options 3 Way Calling enable disable 3 way calling In directory
39. www atcom cn IP XX User Manual www atcom cn 1 2 2 1 2 2 2 3 2 4 2 2 6 2 7 2 8 2 9 2 10 2 11 2 12 2 13 2 14 2 15 2 16 2 17 2 18 2 19 2 20 2 21 2 22 2 23 2 24 2 25 2 26 3 3 1 3 2 3 3 3 4 3 4 1 3 4 2 3 5 3 6 3 6 1 3 6 2 3 7 3 8 3 8 1 MEOdU E oinen nE EN E NE EN ea tees aa eamaeaanonen vasa AN 4 Contioure the Gevice Vid GU eenn EEEE E EA EAA 5 Access the GU lesen ciiaeeeseccnleieuonc ace saiatacgvacscensenaiemniass alec aaa 5 DVS Ce lo CUS a cacasenanesaaastoiesatanacesateel E E 5 Connoure AGOW Al Cries ccs ieee A I E 6 MUTATING occ cece senses ances care Soh E O E SE 6 Outgoine Callin S RUES sosar tarn NAE eesti ect ite love cash eine OTERA 7 PALPI a cedar vanes ae etc eee ne 8 US CLS rn E S arises acess came acease 9 Rine GROUPS reana E I na E anaeaseaancen ahem A ENA 10 Mosicon Holdeni A S a O NA 11 Cali OENES nanna a a a ANA 12 Voice Men IVR Joren a ee N 14 T EI V E A E A O OE N E N A 15 Mcomime Calling RUICS rrsan n sate eR naa ie E AS 16 AV oh Cele v0 12 1 presence E rr Rane creme Ri ane E A cea nee een ore 17 GORTEPEH CIAO oactcc can ctccersnsaatenas N TTO 19 POON MO eee ae ne PRR Pee ea ae eee 20 DIV CCL ORY wi saete re citesass bale rans esac E Sandee accea meer 21 WOIC Cll GTO Diran es arasavetacas evarenevesataksasaranavanucent A eeu hae ee 21 VOICE Menu PLOM US acc sacs acce cies fos cheaancs caeas aaatacntns ne Aes saracheaees eles san dun see Wan Gsacnatineetulosmarmorsies 21 Backe
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