AX4E-Centos5.4-Dahdi-User Manual-V1.0-EN - VoIP
Contents
1. 16 LE 17 ED EU 18 SWA cic 19 CNE Piu i cc 21 3 9 VOICE MENUS c 22 31O IME INTERVALS C Xce 23 ILT INCOMING CALLING RULES EE EAE N 24 2 M 25 3 13 CONFERENCING 26 PAFON ME er 27 BS VOICE IVI SIE GROUPS OC 29 5 10 VOICE MENU PROMPTS 30 SYSTEM NE 31 AU 33 ILIAC CHANNELS 34 OO 35 JSTERNSKLOGS e c 37 SNP IIL p ME HR 38 9 FILE EOITOR M 39 40 Sin xen co dad cc HH 40 OFRIRRIWWAREOPDRATEE M 41 3 26 1 Download the Latest Firmware File and Set up TFTP Server 41 www atcom cn 1 3 26 2 Update for P0O8 from Web Page rrr re2. 11 M conferences V ringgroups V voicemenus 7 queues V voicemailgroups M directory Save After configuring please click on Save button and click on Apply Changes button in up right corner of the main page In configuration screens of 6030 please select dialplanl in the DialPaln drop down list Now you can call from 6030 to 6001 and 6005 by dialing with prefix 9 www atcom cn 60 ATCOM 4 6 How to Transfer Files between Windows PC and IP08 Using WinSCP software it is the most convenient way to transfer files between windows PC and IPO8 Open your WinSCP software enter the IP Address username password of IPOS like the following screen FanSCP Login session Session Stored sessions Host name Fort number User name Password Private ker file Hint Vse Stored sessions tab to save your settings Freferences aAdvanced options Local Mark Commands Session Options Remote Help a dil i ec m 2 121 GA i Changed Size Changed 2010 5 4 documents and Sett 2009 10 i 2010 5 6 1 17 rwrr zr z los 2010 1 2 2010 5 10 2 rwrir zr x Intel 2009 7107 2010 5 10 2 MSOCache 2010 2 4 2010 5 6 1 07 rwzr xr x Program Files 2010 5 1 2010 5 6 1 17 2009 10 2006 12 31
3. Figure IPPBX IPOS ATCOM IPPBX Product Guide Version 1 0 2010 05 10 Content M 3 CHAPTER 1 THEINIRODUCTION OF IPOS 4 CHAPTER 2 ACCESS TO THE POS rq 0628s ec ESOS Saras a kaa 7 2 VV EB PAGE ACCESS BY BROW SEE 7 2 2 SOTWACCESS BY PUTIN 8 2 3 ACCESS BY BROWSER WITH FALLBACK ADDRESS pp 10 2 4 CONSOLE PORT ACCESS TO IPOS 10 CHAPTER 3 CONFIGURE IPOS BY WEB GUD ee 12 SANI ig 12 swAWelllci dulce 12 TUNKS 12 TOO pra detta d andes oa NU MU mre e ba 13 14 JA VUTGOING CALUNG RULES eee ee ee E ere ee 14 CP UD HUP ee 16 Os SERS m T
4. Registration for 500 192 168 1 213 timed out trying again Attempt 6 Apr 21 03 50 36 NOTICE 211 chan Registration for 500 192 168 1 213 timed out trying again Attempt 7 Apr 21 03 51 26 NOTICE 211 chan sip Registration for 5008192 168 1 213 timed out trying again Attempt 8 Apr 21 03 52 16 NOTICE 211 chan sip Registration for 5008192 158 1 213 timed out trying again Attempt 9 Apr 21 03 53 06 NOTICE 211 chan sip c Registration for 500 192 168 1 213 timed out trying again Attempt 10 Apr 21 03 53 56 NOTICE 211 chan Registration for 50068192 158 1 213 timed out trying again Attempt 11 Apr 21 03 54 46 NOTICE 211 chan sip c Registration for 500 192 168 1 213 timed out trying again Attempt 12 Apr 21 03 55 36 NOTICE 211 chan sip Registration for 5008192 168 1 213 timed out trying again Attempt 13 Apr 21 03 56 26 NOTICE 211 chan sip c Registration for 5008192 158 1 213 timed out trying again Attempt 14 Apr 21 03 57 16 NOTICE 211 chan sip c Registration for 500 192 168 1 213 timed out trying again Attempt 15 3 22 Bulk Add Using bulk add you can add multi users one time You can define the number of the users you want to create After click on Options gt Advanced Options gt Show Advanced Options please select the Bulk Add option from the vertical menu on the left then you can get the following screen B
5. VoiceMail Access FIN code D Mailbox 600 Email Address Technology v SIP IaX D nalog Station D flash rxflash Codec Preference First Second SS Third Hone Fourth Hone Fifth VoIP Settings Address D Line Number SIP TAR Password NAT D Can Reinvite D Mode D Insecure very D Other Options j Way Calling D In Directory D call Waiting LJ err D Is Agent D Pickup Troup J Cancel Update In General component you have to set up Extension CallerID Name OutBound CallerID parameters and choose a DialPlan for the extensions Here I set up user 6003 and select DialPlan1 for the user I select Enable Voicemail for this User option so the user has voicemail function In the Technology component you have to select SIP or IAX Here I want to configure a SIP user so I select SIP For the Codec Preference only the first two types of code you set are available In the Other Options component I select Is Agent which will be listed in Call Queues as a selectable member for call queue At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 17 3 6 2 Create Analog User Click on Create New User button the following screen is where you create and set up user Edit User Extension 6005 General Extension 6005 Name 6005
6. VoiceMail Access PIN code D Mailbox Email Address Technology v SIF v IAX D nalog Station flash reflash Codec Preference First second Third Jone Fourth Fifth VoIP Settings Address D Line Number D SIP IAJ Password NAT D Can Reinvite D DTMF Mode D insecure very D Other Options LJ j Way Calling In Directory D call Waiting 24 Is Agent D Pickup G Cancel Update At last please click on Update button and click on Apply Changes button in up right corner of the main page 3 Add up an Analog user 6005 After logging into the web page of IP08 please click on Users gt Create New User I add a user 6005 like the following www atcom cn 46 Edit User Extension 6005 General Extension 60045 vane 6005 D DialPlan CallerID 6005 outbound CallerID 6005 D Enable Voicemail for this User VoiceMail Access PIN cade D Mailbox 6008 D Email Address robert ao atcon D Technology D TAX D Analog Station D flash D T5 rxflash Codec Preference First ulan Second ow Third Fourth Hone Fifth VoIP Settings Address 60045 Line Number SIP IAX Password D Can Reinvite D Mode RFC2833 D Insecure very D Other Options j Way Calling In Directo
7. Through the above settings someone calls 6006 but 6006 does not answer the call will be transferred to 6001 automatically 3 15 VoiceMail Groups Define VoiceMail Groups to leave a voicemail message for a group of users by dialing an extension Please select the VoiceMail Groups option from the vertical menu on the left of the main page then you can get the following screen 7 System Status VoiceNail Groups der Voiceil Grow No VoiceMail Groups defined La Ring Groups Music On Hold Queues C3 voice Menus Time Intervals La Incoming Calling Rules Directory L3 Call Features ta Groups Define Groups to leave a voicemail message for a group of users by dialing an extension www atcom cn 29 Click on New VoiceMail Group button on the illustration above Below is what my VoiceMail Group configuration page looks like Yoice Kail Group VoiceMail Group s Extension 64600 Label notice User MailBexes 6006 Cancel cave From the above settings I can dial 6600 to leave message for user 6005 and 6006 3 16 Voice Menu Prompts This component is used for recording custom voice menu Please select the Voice Menu Prompts option from the vertical menu on the left of the main page then you can get the following screen Custom Yoice Menu Prompts INO Record a new Voice M
8. rwrir zr z Ce yste 1 2010 1 1 persistent 2010 5 6 1 08 temp 2010 2 1 proc 2006 12 31 rczrczr z usr 2010 3 3 O root 2010 5 6 1 07 rwzr zr z 2010 5 7 fq sbin 2010 5 6 1 17 rwzrurruz rnd 1 024 20107371 2006 12 31 rwrr xr x E abandon all hope ulaw 24 640 2008 9 9 tmp 2010 5 10 2 rwrrwzrwt avrozxzc BAT MS DOS 2009 107 usr 2010 5 6 1 17 rwzr xr x S bar emf 393 612 EMF 2010 4 2 var 2010 5 6 1 07 rwzr zr z i 242 RRE 2010 1 2 font bi 322 730 BIN 2008 4 1 sys 0 2009 10 HLI 224 821 X4t 2010 1 2 7 Install log 197 2009 10 0 0 2009 10 1iDSRTP_log tzt 0 Xx 2010 5 4 0 2003 10 MS DOS 2008 4 1 mt 2008 4 1 efile sys 2010 5 1 RHDSetup log 2003 10 E service log 2010 5 1 2 047 in 0 of 27 0 B of 0 B in 0 of 14 2 F2 Rename 5 rove CFT Create directory Delete 636 KB 3 183 B G Bees The left part of the screen displays directories and files of your windows PC the right part of the screen displays directories of 8 www atcom cn 61 ATCOM If you want to transfer a file from windows to IP08 you just need to choose the file and drag it to the directory of IPO
9. DialPlan CallerID 6005 D outBound CallerID 6005 D Enable Woicemail for this User WoiceMail Access FIN code QD Mailbox 6005 D Email Address robert ao atcom D Technology sir D LJ tax D Analog Station D flash TO rxflash Codec Preference First ulan Second Third Fourth Hone Fifth VoIP Settings MAC Address 6OO5 Line Number SIP IAE Password 833 D lnzecure very D N T une D DINF Mode Other Options O j Way Calling D LJ In Directory D ios Waiting D D Is Agent Pickup Group G Cancel Update In the General component you have to setup Extension CallerID Name OutBound CallerID parameters and choose a dialplan for the phone Here I set up user 6005 and select DialPlanl for the user I select Enable Voicemail for this User option so the user has voicemail function In the Technology componet you have to select the port in which the analog phone will be plugged from the drop down list of Analog Station I select Enable Voicemail for this User option so the user have voicemail function In the Other Options component I select Is Agent which will be listed in Call Queues as a selectable member for call queue At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 18 Attension in textbox of Extension
10. Extension 6005 Name 6005 DialPlan CallerID 6005 outBound CallerID 6005 D Enable Voicemail for this User D WoiceMail Access PIN code D Mailbox 5005 D Email Address robert D Technology 5 D D Analog Station D flash 750 reflash Codec Preference First ular Second J C Third Jone Fourth Fifth VoIP Settings Address 6005 D Line Number SIP IAR Password NAT D Can Reinwite D Made RFC2533 D insecure very D Other Options Te Agent A Pickup Group CJ Cancel Update 3 13 Conferencing The conferencing function of Asterisk 15 similar to a Tele conference call where multiple callers can call in and participate in a two way conference like in a party room where everyone can talk and listen to one another or just to listen to a Tele presentation Please select the Conferencing option from the vertical menu on the left of the main page then you can get the following screen Conference Rooms IN New Conference Bridge No Conference rooms defined Ring Groups Music On Hold Call Queues r1 Voice Menus Time Intervals La Incoming Calling Rules r Voicemail La Conferencing MeetMe conference bridging allows quick ad hoc conferences with or without security www atcom cn 26 AN ATCOM Click on New Conference Bri
11. value you set is limited to a range you can adjust range in the following screen to meet your requirement Please select the Options option from the vertical menu on the left then you can get the following screen General Preferences Language Change Password Factory Reset Reboot Advanced Options Operator Extension Ring Timeout 20 Extension preferences User Extensions 6001 Conference Extensions 6300 VoicelMenu Extensions 7001 RingGroup Extensions A400 Queue Extensions 6500 VoiceMail Group Extensions ABO Reset to defaults Cancel Save 3 7 Ring Groups Define Ring groups to dial more than one extension simultaneously or to ring more than one phone sequentially This feature may also be called Hunt groups Please select the Ring Groups option from the vertical menu on the left of the main page then you can get the following screen 7 System Status Nanage RingGroups No RingGroups defined Ring Groups Define Ringgroups to dial more than one extension simultaneously or to ring more than one phone sequentially This feature may also be called Huntgroups El www atcom cn 19 Click New RingGroup button on the illustration above the following screen is where you create and set up ring group RingGroup RingGroup Mame rinzzroupl Extension for this ring group 6400 King Group A
12. 08 you have to set it in web GUI After you enter into the web GUI of IP 08 you can try to configure IP address according to the www atcom cn 40 ATCOM following steps After click on Options gt Advanced Options gt Show Advanced Options please select Network Settings option from the vertical menu on the left of main page the following screen is where you configure the network DHCP Hostname Domain IP address 182 185 1 151 Subnet mask 255 255 250 0 Gateway 192 168 1 1 DNS 192 169 1 1 pool org In the drop down list of DHCP you can see following three options 1 DHCP yes IP08 will obtain the dynamic IP address from your router 2 DHCP auto IPO8 will use the static IP specified below and ping the default gateway When there is no response from the default gateway the IP 08 will switch to dynamically obtain the IP address from your router 3 DHCP no IP08 will use the static IP address set below If you want to get static and permanent IP address please do not select yes after configure other parameters please click save in the bottom of your page to save your setting 3 26 Firmware Update You can update to the latest version for IP08 by TFTP 3 26 1 Download the Latest Firmware File and Set up TFTP Server 1 Download the 5 file from http www atcom cn downloads index php folder S VBQOIgvZmlybXdhcmU then put it in your server root
13. D digits from front and Prepend these digits D before dialing G Cancel save At last please click on Save button and click on Apply Changes button in up right corner of the main page 4 2 3 Create a Dial Plan After logging into the web page of IP08 please click on Dial Plans gt New DialPlan I configure a dial plan like the following Edit DialPlan DialPlan Name DialPlan2 Include Outgoing Calling Rules outgoing Include Local Contexts v ldefault V parkedcalls M conferences M ringgroups V voicemenus V queues V voicemailgroups v directory Save At last please click on Save button and click on Apply Changes button in up right corner of the main page www atcom cn 49 4 2 4 Create a User I will use the user 6001 I created before here I need to reselect a dial plan for 6001 here I need to use DialPlan2 so I select DialPlan2 in the DialPlan drop down list Now I can call out with prefix 2 if the caller number is 10086 I will dial 210086 4 3 How to Get an Incoming Call from outside In order to get an incoming call from outside with IP08 you need an analog trunk an incoming calling rule a destination here I use IVR Here I will give the simple configuration steps which show how to get an incoming call from outside for detail configuration you can refer to chapter 3 4 3 1 Create an Analog Trunk I use the trunk2 I created in 4 2 1 4 3 2 Create an Inco
14. Options gt Advanced Options gt Show Advanced Options please select the Call Detail Records option from the vertical menu on the left then you can get the following screen CDR Viewer CDR CS CDR viewer lt lt prev next gt gt Wiewing 1 25 of 357 View 26 most recent first Account Caller Answer Billeble Source Destination Dest Context Channel Last data Start time 5 End Time Duration Disposit Code ID Time seconds l B i 5 d i i id 6001 Local 6001 defaul t 6553 ents Dial 1 6001 amp 2 6001 20 2010 04 2010 04 1 6005 6001 DielPlent ES 1 012 0004 SIP 60018IAX2 8001 20 2010 04 21 05 31 21 21 21 ANSWERE 05 32 01 05 32 04 default Local 6001 default Tell tExt Be et ad sud era oc era c 005 004 xten 05 31 09 05 31 29 05 31 47 B pd a d gt i IMS d 2010 04 12010 04 21 21 ANSWERED 05 23 50105 24 05 um 22 i You can click on the prev to look up the last page for call record and click on the next to look up the next page for call record you can also set the value from the drop down list of view which means how many calls will be displayed in one page Eee www atcom cn 43 Chapter 4 an Application Case of IP08 IP Address 172 16 1 2 Extension 6020 IP Address 192 168 1 10 AT 610 AT 620 IP04 IP Address IP Address
15. Time Interval Time Interval Name timeintervall By day of week Man ta Fri Days of a Month Date Month M Time Entire Start Time 09 00 End Time 08 30 Cancel Update Time Interval Name a unique label to help you identify the time interval when listed in incoming calling rules I set up timeintervall as time interval name By day of week I select it from Monday to Friday the incoming call rule only works from Monday to Friday Time I set up it from 09 00 AM to 06 30 PM the incoming call rule only works from 09 00 AM to 06 30 PM At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 23 3 11 Incoming Calling Rules This is where the behavior of incoming calls from all trunks is being handled When an incoming call from PSTN or VoIP trunk is received asterisk needs to know where to direct it It can be directed to a ring group an extension digital receptionist voice menu or queue For this purpose Incoming Calling Rules need to be set up Please select the Incoming Calling Rules option from the vertical menu on the left of the main page then you can get the following screen Incoming Calling Rules Note If you have multiple SIP trunks from the same provider you ll need to make Incoming Calling Rules for each Contact Extension destination on ALL trunks from that provide
16. directory 2 Run your TFIP server and I set up it like the following www atcom cn 41 ATCOM Iftpd32 by Ph Jounin Curent Directory E upgrade Browse Server interface 192 1681 111 Show Server Client DHCP server Syslog server Current Action Listening an port 69 TTE TEN E Nupgrade is the root directory of my TFTP server 192 168 1 111 15 the IP Address of my server 3 26 2 Update for IP08 from Web Page After click on Options Advanced Options Show Advanced Options please select Firmware update option from the vertical menu on the left of main page the following screen is where you update for IPOS Download image from a HTTF URL 9 Server Server File Name E Reset Configs Server enter the Address of your TFTP server in this textbox File Name enter the update file name Reset Configs if you choose reset Configs it will delete all of your configuration you have done before After setting up please click on Go button to update for IP08 Power off and power the 08 wait for several minutes When the TEL port LEDs light up it means the update is finished and you have the latest firmware www atcom cn 42 3 27 Call Detail Records This component provides the record of all incoming and outgoing calls including the channels used and duration of calls After click on
17. 1 10 is the IP address of 2 Add an AX user 6020 in IPOS After logging into the web page of 8 please click on Users gt Create New User I configure 6020 like the following Edit User Extension 6020 General Extension 6020 0 Name 6020 DialPlan CallerID 6020 D outBound CallerID 6020 D Enable Voicemail for this User D VoiceMail Access PIN code D Mailbox 6020 Email Address Technology v SIF D v Lax D nalog Station D flash reflash Codec Preference First Law Second SS Third Fourth Fifth VoIP Settings MAC Address Line Number D SIP IAE Password 6020 D NAT Can Reinvite 1 D DTMF Mode D insecure very D Other Options FEES Mies OS err 1 Agent Pickup G Cancel Update At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 53 ATCOM 3 Set up AT 620 and register an IAX2 user 6020 After logging into web page of IP Phone AT 620 please select Network option to enter the screen of configuring IP Address I set up a static IP Address 172 16 1 2 Netmask 255 255 0 0 Gateway 172 16 1 254 After finishing the configuration please click on the Apply button You can refer to the following screen IP Phone ATCOM Current Status Network V
18. 1 4 1 2010 05 06 01 01 49 EDT Built in Enter help for a list of built in commands ee www atcom cn Shell imah ATCOM 2 3 Access by Browser with Fallback IP Address This way only be supported by the latest version IP08 0 3 6 of IPO8 If you forget the IP Address of IPO8 you have set up you can use the fallback IP Address 172 31 255 254 30 Before logging into IPOS please set up the IP Address of your PC 172 31 255 253 and SubMask 255 255 255 252 At last you can open your browser and enter 172 31 255 254 to log into the web page of IPO8 2 4 Console Port Access to IP08 If you do not have network connection between IP08 and PC you can try to access to IP08 by console port Please try to do as the following steps 1 Please connect the console port of IPO8 to your PC s console port with RS232 console cable you can refer to the following illustration 2 Please run your Hyper Terminal and set up the console port like the following Bits per second 115200 Data bits 8 Parity None Stop bits 1 Flow control None www atcom cn 10 _ o ATCOM Change the IP Address by Hyper Terminal The default IP address of IP 08 15 192 168 1 100 Your network may have a different IP address range such as 192 168 10 xx In this situation you can not access to IP 08 by putty and browser if you do not change the IP 08 IP address So you have to change the IP address for IP 08 by Hyper Terminal to mak
19. 50 Check 15 VoicelMailMain xNo Extension assigned Dial by Names Directory Call Features www atcom cn 7 2 2 SSH Access by Putty Logging into IP08 by SSH you can configure IP08 by Linux command 1 Please open your putty software and input the IPOS IP address in the Host Name textbox input port number in the Port textbox click the SSH Connection type then click open button Please refer to the following screen i Pull Configuration Category j Session Basic options for your Put TY session mu Logging Specify the destination vau want to connect to E Terminal Keyboard Host Name or IP address Port Features Connection type Window ORaw Telnet C2Rlogin 9 55H Serial os Load save or delete stored session Behaviour Saved Sessions Selection Colours E Connection Default Settings Load Prony Telnet Delete SSH Serial Close window on exit Always Never 9 Only on clean exit Cancel 2 Please input username root and the default password 12xerXes16 in the following screen you can access to IPOS successfully www atcom cn 8 login root frootWi92 168 1 100 s password When you log into IPOS successfully you can get the following illustration roothi92 168 1 100 s BusyBox
20. 8 pb 61 CHAPTER S REFERENCE D URS UR CURA 63 www atcom cn 2 ATCOM Contact ATCOM The Introduction of Founded in 1998 ATCOM technology has been always endeavoring in the R amp D and manufacturing of the internet communication terminals The product line of ATCOM includes IP Phone USB Phone IP PBX VoIP gateway and Asterisk Card Contact Sales Address FL2 Block 3 HuangGuan Technology Park 21 Tairan 9 Rd Chegongmiao Futian District Shenzhen China 518040 86 755 23487618 86 755 23485319 sales atcomemail com Contact Technical Support Tel 86 755 23481119 Support atcomenal on Website Address http www atcom cn ATCOM Wiki Website http www openippbx org index php title Main_Page Download Center http www atcom cn download html www atcom cn 3 Chapter 1 the Introduction of IP08 Overview of the IP08 The 8 is a complete Asterisk Appliance with four Dual ports FXO or FXS modules It is an embedded open source Linux system with built in SIP IAX2 proxy server and NAT features It provides a solid uniform platform for traditional PSTN communications as well as VoIP communications Targeting for SOHO user and SMB market with an easy to use graphical interface IP08 provides a cost saving solution on their telecommunication data needs With IP08 company with branch office
21. After logging into the web page of IP08 please click on Trunks gt Analog Trunks I configure an analog trunk like the following Edit Trunk E Channels Trunk Name trunk CallerID Normally you should not have to adjust your analog ports beyond the initial Porro calibration Should you still need to fine tune your audio settings please use the adjustments at the right Advanced Options Busy Detection D Busy Count D 5 Ring Timeout D 2000 Answer on Hangup Ho Polarity Switch D Polarity Switch Call Progress Ho Progress fone Use CallerId D Caller ID Start CallerID g As Received Pulse Dial 8 CID Signalling mailbox Flash Timing D Tau Receive Flash Timing Cancel Update At last please click on Update button and click on Apply Changes button in up right corner of the main page a www atcom cn 48 4 2 2 Create an Outgoing Calling Rule After logging into the web page of IP08 please click on Outgoing Calling Rules gt New Calling Rule I configure an outgoing calling rule like the following Edit Calling Eule Calling Rule Name D outgoingl Pattern D 2X Send to Local Destination D send this call through trunk Use Trunk Strip D 1 digits from front and Prepend these digits D before dialing Use FailOver Trunk D fail over Trunk D Strip
22. IP Address 192 168 1 3 192 168 1 8 192 168 1 20 Extension 6001 Extension 6030 Analog Phone Extension 6005 Figure Network Topology In the network topology above user 6020 and user 6001 will be registered to IP08 user 6030 will be registered to IP04 analog phone 6005 is connected to FXS port of IP08 After configuration it will realize the following function 1 The internal user 6005 and user 6001 can call each other directly 2 6005 and 6001 can dial out through 8 to PSTN 3 6005 and 6001 can get incoming calls from PSTN by IPOS 4 6030 can call out to PSTN and get incoming call from PSTN through IP04 5 User 6001 and 6030 can call each other through VoIP trunk although they are registered to different IP PBX 6 User 6020 6005 and 6001 can call each other directly although they are not in the same network segment FE www atcom cn 44 4 1 How to Make Internal Calls through IP08 4 1 1 Access to the Web Page of IP08 by Browser After connecting IP08 to LAN please open your browser of PC with windows OS and input the IP Address of IP08 the default IP address is 192 168 1 100 then you can get the following screen Please login Asterisk Configuration Engine Username Password Login Please input the default Username admin Password atcom in the presented screen above When you login successfully you can get the configuration web page as below 2 Logout
23. Now you can call from 6001 and 6005 to 6030 by dialing 96030 Eee www atcom cn 57 4 5 2 Call from IP04 to IP08 In order to call from IP04 to IPOS I will create a SIP user in IPOS for the SIP trunk in IP04 create a SIP trunk an outgoing call rule and dial plan in IP04 1 Add a user 6008 in Add SIP user 6008 after logging into the web page of IP08 please click on Users Create New User I add a user 6008 like the following Create New Uzer General Extension 6008 0 Name 6008 DialPlan CallerID 6008 D outBound CallerID 6008 D Enable Voicemail for this User D VoiceMail Access PIN code D Mailbox G Email Address Technology v SIF v TAX D Analog Station flash T50 reflash 1250 Codec Preference First sula Second SS Third Fourth W Fifth VoIP Settings Address Line Number 51 Password 6008 D Can Reinvite D DIMF Mode D lnzsecure very D Other Options 2 j Way Calling D In Directory D ad Waiting D A cuu D Is Agent D Pickup is raup G Cancel Update At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 58 2 Create a SIP trunk in IPO4 Add a VoIP trunk in IP04 after logging into the webpage of please click Trunks gt VOIP Trun
24. OIP Advanced Dial peer Config Manage Update System Manage Static gt o C Primary DNS 202 96 134 133 iic co z o D N m e lt s LES n gt D a ES D o w Alter DNS 202 96 128 68 Please select the VOIP option then select the A X2 option I register the IAX2 user 6020 as the following illustration IP Phone Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Local Voice Mail Number Voice Mail Text Enable G 729 After configuring please click on the APPLY button Attention here you must register IAX2 user instead of SIP user because the user 6020 is not in the same network segment as IP08 If you use SIP user you can not get sound when the communication is established Now you can call each other among 6020 6001 and 6005 directly www atcom cn 54 4 5 How to Call through VoIP Trunk 4 5 1 Call from IP08 to IP04 In order to call from IPOS to IP04 I will create a SIP user in IP04 for the SIP trunk in 8 create a SIP trunk an outgoing call rule and a dial plan in 1 Add an SIP user 6035 1 will be used as SIP trunk in IP08 in IP04 after logging into the web page of IP04 please click on Users Create New User add the user 6035 like the following Edit User Extensi
25. S at last click on copy button in the popping up screen like the following www atcom cn 62 Chapter 5 Reference http atcom cn download html http www asteriskguru com http www openippbx org index php title Main Page http www atcom cn www atcom cn 63
26. ash Ignoring disallow Ignoring allow a m m m Unable to create channel of type cause 3 No route to destination Apr 21 03 45 36 NOTICE 211 chan sip c Registration for 5000192 168 1 213 timed out trying again Attempt 1 Apr 21 03 45 40 WARNING 19691 ast expr2 fl ast yyerror syntax error syntax error unexpected expecting end Input Apr 21 03 45 40 WARNING 19691 ast expr2 fl If you have questions please refer to doc channelvariables txt in the asterisk source Apr 21 03 46 06 WARNING 19691 app dial c Unable to create channel of type cause 3 No route to destination Apr 21 03 46 26 NOTICE 211 chan sip c Registration for 5008192 158 1 213 timed out trying again Attempt 2 Apr 21 03 47 16 NOTICE 211 chan Registration for 50068192 168 1 213 timed out trying again Attempt 3 Apr 21 03 47 46 WARNING 211 chan sip c Maximum retries exceeded on transmission 24806208277904 20042119194361 192 168 1 3 for seqno 1 Critical Response Apr 21 03 47 46 WARNING 211 chan sip c Hanging up call 24806208277904 20042119194361 192 168 1 3 no reply to our critical packet Apr 21 03 48 06 NOTICE 211 chan sip Registration for 5008192 158 1 213 timed out trying again Attempt 4 Apr 21 03 48 56 NOTICE 211 chan sip c Registration for 50068192 188 1 213 timed out trying again Attempt 5 Apr 21 03 49 46 NOTICE 211 chan
27. aveWhenEmpt D JoinEmpty Queue Options TimeOut 15 Wrapup Time 0 Len 0 D duto Fill C D Auto Pause D Report Hold Time KeyPrese Events Agents D 16002 6002 6003 6003 Cancel Update Extension a unique label to help you identify the call queue when listed in outgoing calling rules component Agents select the users which you want them to be queue member You can get information of other parameters by putting your mouse on the label At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 21 3 9 Voice Menus Like most organization we would like to redirect all of the incoming calls automatically The voice menu is very handy for these sorts of things The system should allow callers to make the selection according to the voice menu Please select the Voice Menus option from the vertical menu on the left then you can get the following screen Extension Dial Other Extensions Key Press Actions 7001 Edit Delete t3 Voice Menus Menus allow for more efficient routing of calls from incoming callers Also known as IVR Interactive Voice Response menus or Digital Receptionist Click on Create New VoiceMenu button on the illustration above the following screen is where you create and set up voice menu Edit YoiceNenu voicemenu cust
28. d button and click on Apply Changes button in up right corner of the main page www atcom cn 56 3 Create an outgoing calling rule in IPOS after logging into the webpage of IP08 please click on Outgoing Calling Rules gt New Calling Rule I configure an outgoing2 rule like the following Edit Calling Eule Calling Rule Wane D outgoing Pattern Send to Local Destination D Send this call through trunk Use Trunk Strip 1 digits from front and Prepend these digits before dialing Use FailOver Trunk 2 fail over Trunk D strip D digits from front and Prepend these digits D before dialing G3 Cancel cave After configuring please click on Save button and click on Apply Changes button in up right corner of the main page 4 Create dial plan in IPOS after logging into the webpage of IP08 please click on Dial Plans New DialPlan I configure a dialplan2 like the following Edit DialPlan DialPlan Name DialPlan2 Include Outgoing Calling Rules outgoing outgoing Include Local Contexts default V parkedcalls V conferences V ringgroups V voicemenus 7 queues V voicemailgroups directory Save After configuring please click on Save button and click on Apply Changes button in up right corner of the main page In configuration screens of 6001 and 6005 please select dialplan2 in the DialPaln drop down list
29. dge button on the illustration above Below is what my conference configuration page looks like Edit Conference Bridge 6300 Extension 00 Marked Admin user Extension Password Options 173 D 156 Conference Room Options D Play hold music for first F Close conference when last marked user caller exits Enable caller menu Announce callers Mode Wait for marked user Cancel Update Naturally there are some options that you may wish to have for the conference room They are entirely up to you The main important things are for you to create the conference room number and the conference pin code for you to know how to enter into the conference The rest of the fields are optional You can get information of other parameters by putting your mouse on the D label This conference number is 6300 the Pin Code is 123 for common member the Pin Code is 456 for Admin So you have to dial 6300 then press the Pin Code if you want to enter the conference I enable the play hold music for option and announce callers option so the first member who enter the conference will listen to a music and the online members will be informed when someone enter the conference At last please click on Update button and click on Apply Changes button in up right corner of the main page 3 14 Follow Me If A calls B B does not answer the call will be transferr
30. e it in the same network segment as your LAN After you have accessed to IP 08 by Hyper Terminal please use the following command to change the IP address for IP 08 root gt ifconfig eth0 192 168 1 151 the IP address you want to set for IP 08 By this way the IP address you set for IP 08 15 temporary it will recover to the original default IP address after rebooting If you want to give a static and permanent IP address for IP 08 you can try to set itin web GUI for detail steps please refer to chapter 3 np 28 www atcom cn 11 Chapter 3 Configure IP08 by Web GUI 3 1 System Status In the system status screen it displays the functions you configured such as trunks extensions conference and so on like the following screen System Status Please click on a panel to manage related features Uptime 05 17 03 up 2 min load average 0 18 0 08 0 02 Trunks Status Trunk siptrunkl trunk2 Unregistered Username 6035 Port Hostname IP 192 168 1 20 Ports 2 Agents Configure Hardware Free Busy Conference Rooms 6300 Not in use Extensions Unavailable Ringing Extension 6001 6002 6003 65004 3 2 Configure Hardware Nane Label Status 6001 6002 6003 6004 Type SIP IAX User SIP User SIP User Analog User Port 3 In the configure hardware page it includes the foll
31. ed to C who 15 set up in follow me Please select the Follow Me option from the vertical menu on the left then you can get the following screen www atcom cn 27 j Follow FollowMe Preferences for Users FollowMe Options Extension Follow Follow Order 6001 Disabled Not Configured 6002 Disabled Not Configured 6003 Disabled Not Configured 6004 Disabled Not Configured 6005 Disabled Not Configured 6006 Enabled You can choose user for which you want to setup follow me function Here taking the user 6006 for an example click on the edit button at the same line as 6006 you can get the following screen Status D Enable Disable Music On Hold Class DialPlan D Destinations Add FollowBe Humber 1 Save Select the enable status and click on Add FollowMe Number button to add a destination phone www atcom cn 28 Status 9 Enable Disable Music On Hold Class D DialPlan Destinations 6001 10 seconds 1 Number D 9 Dial Local Extension Dial Outside Number for SO Seconds 5001 6001 Dial Order D D r Trying previous extension number S004 6004 g with previous extension number 6005 BOOS BOWE 6006 Add Click on Dial Local Extension and select 6001 Click on Add button and click on Apply Changes button in up right corner of the main page
32. enu prompt Upload a Voice Menu prompt No custom Voice prompts found La Ring Groups You can record new VoiceMenu Prompt by clicking on the Record a new Voice Menu prompt c3 Music On Hold 1 2 Hed or click on the Upload a Voice Menu prompt button to upload a custom voice menu L3 Call Queues cJ Voice Menus Time Intervals La Incoming Calling Rules Yoicemail Record or Upload custom VoiceMenu prompts Click on Record a new Voice Menu prompt button on the illustration above Below is what my Record a new Voice Menu prompt configuration page looks like Record a new Yoice Nenu prompt File Mame WelcomToATCOM dial this User Extension to record new voice coni NN prompt G Cancel im Record www atcom cn 30 ATCOM File Name give a filename for the record sound file here I give a name WelcomToATCOM Dial this User Extension to record a new voice dial to a user then the user pick up the phone and speak the voice menu which will be recorded Here I select 6001 I set up before Click on Record button the asterisk will call to 6001 6001 will show like the following Incoming cal from asterisk Answer amp Ignore Click on Answer button then you call speak and start to record what you say The following illustration will be presented after you click on the Answer button Talking asterisk 0 00 23 zm Hang When you want to finis
33. every single parameter by putting your cursor on the D label At last please click on Save button and click on Apply Changes button in up right corner of the main page The way of outgoing calling rules works Every time you dial a number asterisk will do the following in strict order Examine the number you dialed Compare the number with the pattern that you have defined in your first outgoing rule and if www atcom cn 15 ATCOM matches it will initiate the call using that trunk If it does not match it will compare the number with the pattern that you have defined in the second outgoing rule and so on e Pass the number to the appropriate trunk to make call 3 5 Dial Plans A DialPlan is a set of Calling Rules that can be assigned to one or more users Please select the Dial Plans option from the vertical menu on the left of the main page then you can get the following screen DialPlans rip T 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 z one Dial Plan for local calling that only permits users of that Dial Plan to dial local mmbers via the local outgoing calling rule Another user may be A DialPlan is a set of Calling Rules that can be assigned to one or more users No DialPlans defined Click on New DialPlan button on the illustration above the following sc
34. free internal calls Click New SIP IAX Trunk the following screen is where you create and set up VoIP trunk Create New SIP IAE trunk Hostname 192 168 1 213 Username 500 Fromuser Fromdomain Password Contact Ext Insecure CJ Cancel Add The important parameters are Type You can select SIP or AX type to meet your need Provide Name a unique label to help you identify the trunk when listed in outgoing calling rules and incoming calling rules Hostname the IP address or domain name of your service provider s server Username the username that your service provider configured Password the password that your service provider configured for the user 3 4 Outgoing Calling Rules Outgoing calling rules is used to route an outgoing call when you make an external call which trunk and what dial pattern the call used are configured in outgoing calling rules Please select the Outgoing Calling Rules option from the vertical menu on the left of the main page then you can get the following screen www atcom cn 14 Calling Rules New Calling Rule Outgoing Calling Rules Calling Rules define dialing n outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern This allows different patterns to be dialed through different trunks permissions and routing e g local T digit dials through a PRI but long di
35. h the record please click on Hang up button tecord a new Voice Menu prompt a Voice Menu prompt Options 1 WelcomToATCOM gsm Record Again Play Delete After you finish the recording please refresh you webpage and enter into voice menu prompts component again you can see you have had a sound file like the above 3 17 System Info From this component you can easily get the basic system information it includes General Information www atcom cn 31 System Information IM General Network Disk Usage Memory Usage OS Version Linux IFOX 2 6 22 165 ADI 2008Riastfin syn 5 Thu 29 23 10 13 EDT 2010 blackfin unknown Uptime 05 04 05 up 13 win Load Average 0 05 0 18 0 09 Version Details sterisk 1 23 21 2 wersion 2 0 2 Ce Firmware version atcom ce 04 0 3 6 Server Date amp TimeZone Wed May 5 05 04 05 EDT 2010 Hoztname The latest version of 8 is atcom_ce_ip08 0 3 6 You see version that you are using from Version Details in the above illustration Network Information System Information Network Disk Usage Memory Usage Link encap Ethernet Hiaddr 00 09 45 76 59 76 inet addr 192 165 1 151 Beast 192 166 1 255 Mask 255 255 255 0 BROADCAST RUNNI MULTICAST MTU 1500 Metric 1 packets 1272794 errors 0 dropped 0 overruns 0 packets 66865435 e
36. hing Active Channels in 5 Seconds Channel Seconds Application Zap 2 1 34 Zap 3 1 undefined Zap 4 1 41 Dial ARG2 IRINGCTIME f IDIALOPTIONS www atcom cn 34 3 20 Options This component is used for administrator to manage the system it includes the following modules General Preferences Language Change Password Factory Reset Reboot Advanced Options General Preferences you can set up a user to be the operator and the range of extension number for different types extensions like the following screen General Preferences Language Change Password Factory Reset Reboot Advanced Options Global Outbound CID Operator Extension D Mser 001 Ring Timeout D 20 Extension preferences User Extensions 6001 Conference Extensions 6300 VoiceMenu Extensions 7001 RingGroup Extensions 6400 Queue Extensions 6500 VoiceMail Group Extensions 6600 Reset to defaults Cancel Save Language change the sound file language in which they play General Preferences Lan guage Change Password Factory Reset Reboot Advanced Options G Cancel Change Password it is used for customers to change the admin password click on the Change Password button the following illustration will be presented below www atcom cn 35 ATCOM General Preferences Language Change Password Factory Reset Reboot Advanced Options Enter New Password Retype New Password U
37. iceMails SMTP Settings Extension for checking messages D 6750 Music On Hold Direct Voicemail Dial D 0 Queues Max greeting in seconds 30 La voice Menus Time Intervals Dial 0 for Operator t3 Incoming Calling Rules Nessage Options cJ Voicemail Maximum messages per folder 2 General settings for voicemail Max message time D 2 minutes Min message time 1 second Click on General Settings button on the illustration above You can see the following screen General Settings Email Settings for VoiceMails SMTP Settings Extension for checking messages 2750 Direct Voicemail Dial D Max greeting seconds Dial for Operator D Nezzage Options ra cn K m Maximum messages per folder Max message time minutes Min message time 1 second Playback Options nay message Caller ID sav message duration Play envelope Allow usera review C Cancel W Save Extension for checking messages when you dial 6750 you will hear the voice message other people left for you You can get information of parameters by putting your cursor on the D label If you want to set www atcom cn 25 voicemail function for the user you have to enable voicemail component when you set up a user Please refer to the following illustration Edit User Extension 6005 General
38. ions gt Advanced Options Show Advanced Options please select Asterisk Logs option from the vertical menu on the left of the main page then you can get the following screen Asterisk Log messages IN Click the textbox you can get the following screen Asterisk Log messages IN April 2010 Tue Wed Thu Fri 1 2 B T 9 13 14 15 16 20 22 23 2T 28 29 3l You can see a date table and you can select the log to watch by clicking on the date After choosing the date please click on Go button you can see the asterisk log of the day you choosed Here I need to see the asterisk log of April 2172010 I click on 21 in the date table I get the following screen Asterisk Log messages 21 Apr 2010 I click on Go button then I get the log in the following screen www atcom cn 37 Asterisk Log messages 21 Apr 2010 Apr 21 03 44 29 WARNING 19672 chan Apr 21 03 44 29 WARNING 19672 chan Apr 21 03 44 29 WARNING 196T2 chan Apr 21 03 44 29 WARNING 196T2 chan Apr 21 03 44 29 WARNING 19672 chan zap Apr 21 03 44 29 WARNING 19672 chan zap Apr 21 03 44 29 WARNING 19672 chan Apr 21 03 44 29 WARNING 196T2 chan Apr 21 03 44 29 WARNING 196T2 chan Apr 21 03 45 16 WARNING 19880 app dial Ignoring insecure Ignoring signalling Ignoring macaddress Ignoring autoprov Ignoring label Ignoring linenumber Ignoring fl
39. ipreferences conf rc org conf logger cont Sip cont enum conf musiconhold conf dnzmgr conf rip conf laxpraov cont sip notify cant Here I select users conf file so I can see the file and edit to meet my requirement www atcom cn 39 users conf Add Context 500 6001 6003 6004 6005 6006 general trunk_1 3 24 Asterisk CLI These are some of the available CLI commands that can be executed from the console you can input the asterisk CLI commands from the web page directly After click on Options gt Advanced Options gt Show Advanced Options please select the Asterisk CLI option from the vertical menu on the left then you can get the following screen Lommand help Execute a shell command abort halt Cancel a running halt agent logoff Sets an agent offline agent show Show status of agents agent show online Show all online agents agi debug Enable AGI debugging agi debug off Disable AGI debugging agi dumphtml Dumps a list of agi commands in html format agi show List AGI commands or specific help cdr status Display the CUR status core set debug channel Enable disable debugging on a channel core set debug Set level of debug chattiness core set debug off Turns off debug chattiness Here I input help command in the textbox so I can get all the command which I can use in CLI mode 3 25 Network Settings In order to give a static and permanent IP address for IP
40. ired to set up for your specific ISF Please look at the table below to configure the Router for your Internet connection From the web page of your router please configure the IP address subnet mask and DHCP I configure them like the following Network Setup Router IP Network Address Server Settings DHCP Local IP Address Subnet Local DHCP Sewer Start IP Address Mumber of Address DHCP Address Range Client Lease Time WINS 255 255 255 0 Enable Disable 192 168 1 1 254 182 168 1 1 to 192 168 1 254 minutes 0 means ane le l b Save Settings Cisco SYSTEMS Cancel Changes From the webpage of your router please configure port range forwarding like the following E www atcom cn 52 LINKSYS A Division of Cisco Systems Inc Firmware versian 1 05 00 Etherfast Cable DSL Router BEFSR41V3 Applications amp Gaming Setup Security eee Administration Status amp Gaming Port Range Forwarding Port Triggering UPAP Forwarding Port Range Forwarding Port Range Forwarding Port Range Forwarding can used to set up public services an your network When users fram the Internet make certain requests on your network m Both v 1921681110 the Router can forward to jo Both v 19218810 O computers equipped to handia tho enact fax The user 6020 uses IA X2 the port number 15 4569 192 168
41. ks gt New SIP IAX Trunk I configure a SIP trunk like the following Create New SIP IAE trunk Provider Name siptrunktelPoa Hostname 192 166 1 10 Username 6008 Fromuser Fromdomain Password Contact Ext Insecure Type D Cancel Add After configuring please click on Add button and click on Apply Changes button in up right corner of the main page www atcom cn 59 3 Create an outgoing calling rule in 1 4 After logging into the webpage of IP04 please click on Outgoing Calling Rules gt New Calling Rule I configure an outgoing rule like the following Hew CallingRule EUR CENE eee outgoing Pattern D 8 Send to Local Destination Destination send this call through trunk Use Trunk D siptrunktoIP S8 Strip 1 digits from front and Prepend these digits D before dialing Use Fail ver Trunk D 5 fail over Trunk Strip D digits from front and Prepend these digit D before dialinz Cancel cave After configuring please click on Save button and click on Apply Changes button in up right corner of the main page 4 Create dial plan in IP04 After logging into the webpage of IP04 please click on Dial Plans New DialPlan I configure a dialplanl like the following Edit DialPlan DialPlan Name DialPlanl Include Outgoing Calling Rules outgoing Include Local Contexts V default
42. ming Calling Rule After logging into web page of IP08 please click on Incoming Calling Rules New Incoming Rule I configure an incoming calling rule like the following Edit Incoming Calling Rule Irunk trunk 1 Iime Interval none Hone Timelntervals matched Pattern E Trunk Time Interval Destination VaeicsMenu ive Vr Cancel Update At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 50 ATCOM 4 3 3 Create a Voice Menu After logging into the web page of IP08 please click on Voice Menus gt Create New VoiceMenu I create a voice menu like the following Edit YoiceNenu voicemenu custom z Name wvoicemenul Advanced Edit Extension TOO Allow Dialing Other Extensions Actions Answer the call Play record elcomToaATCom amp Listen far KeyPress events Goto User 6001 Hale figu Allow KeyPress Events Goto User 6001 Goto User 6005 When the call comes from port 2 the system will play a record sound file if the caller presses 1 user 6001 will ring if the caller presses 2 user 6005 will ring If the caller does not press any key the call will go to 6001 You can also configure IP04 to let 6030 call outside and get incoming call by the steps are the same as IPOS you can refer to configuration of IPOS 4 4 How to Call Each Othe
43. of the current system configuration Backup Management Click on Create New Backup button on the illustration above you can get the following illustration Create Backup File Name backup 2010apr26_ 115450 GJ Cancel Backup File Name give a file name for the backed up file Click on Backup button once the backup process is completed you will see a screen with the backup filename displayed in illustration below www atcom cn 33 Create New Backup List of Previous Configuration Backups 1 backup_2010apr26_115450 Apr 26 2010 Download from Unit Restore Previous Config Backup itself is not useful if it cannot be restored IPOS8 also has this function This is a very simple procedure All you need to do is to click on the Restore Previous Config option 3 19 Active Channels The channels which are in communication status will be displayed in this component Please select the Active Channels option from the vertical menu on the left then you can get the following screen La System Status Channel HNanagement Trunks Refresh Now Outgoing Calling Rules Dial Plans Refreshing Active Channels in 2 Seconds Ring Groups Music On Hold Call Queues Voice Menus Time Intervals Incoming Calling Rules Voicemail Conferencing Follow Me Directory Call Features Groups Voice Menu Prompts La System Info Backup Refres
44. on 6035 General Extension 60345 pan DialPlan CallerID 6025 outbound CallerID 6035 D Enable Voicemail for this User E 1an Access PIN code D Mailbox 6001 Email Address D Technology v SIP D nalog Station D flash rxflash D Codec Preference First Slew Second esm Third Fourth Fifth NAT D Can Reinvite O D DTMF Mode D insecure very D Other Options LJ j Way Calling D C In Directory D Call Waiting Fo D Is Agent D Enable Call Record D Pickup Group VoIP Settings Address Line Number SIP IAE Password 6035 CJ Cancel y Update At last please click on Update button and click on Apply Changes button in up right corner of the main page Add a SIP user 6030 in IP04 for AT 620 the way is the same as adding 6035 2 Add a VoIP trunk in IPOS after logging into the webpage of IPOS please click on Trunks VOIP Trunks New SIP IAX Trunk I configure a SIPtrunkl like the following SSS CC www atcom cn 55 Edit SIP trunk 6035 Provider Name siptrunkl Hostname 192 169 1 20 Username 6035 Fromuser Fromdomain Password Contact Ext s Insecure Codecs First Second Third GSN vi Fourth G T2B Fifth Callerm ub Enable Remote Cancel v Add After configuring please click on Ad
45. or you to access IPOS 1 Web page access by browser 2 SSH access by putty 3 Access by browser with Fallback IP Address 4 Console port access by RS232 console cable In order to access to by the first three ways you have to check that if your network connection between IP08 and PC is OK If you do not have network connection between IP08 and PC you can try to use the last way to access to 8 and change IP address for 08 2 1 Web Page Access by Browser It is the most convenient and common way to access the IP08 you just need to open your browser and input the IP address of IPOS WAN port the default IP address is 192 168 1 100 You would better use Firefox instead of IE because there are compatible issues Then input the default Username admin Password atcom the password of old version is mysecret in the presented screen like the following ATCOM YolPtel Please login Asterisk Configuration Engine Username Password Login When you login successfully you can get the configuration web page as below 3 2 Logo System Status 9 Please click on a panel to T manage related features Uptime 04 14 57 up 10 min load average 0 04 0 06 0 02 Trunks Status Username Port Hostname IP Configure Hardware Extensions Free Busy Unavailable e Ringing Extension 1 Status Type 67
46. osm Name wolcemenul Advanced Edit Extension TOO D D Allow Dialing Other Extensions Actions Answer the call Play 20046111556565 al amp Listen far KeyPress events Goto User 6001 Add new Step D Allow kevPress Events Goto Operator Goto Ringsroup ringgroupl Goto User 6001 Name a unique label to help you identify the voice menu when listed in incoming calling rules Add new Step select an action from the drop down list I add three steps above so it will answer the call and play a sound file at last go to user 6001 Click on Allow KeyPress Events when the caller is in voice menu they can press some specific numbers which are defined here to enter other destination Here I define three numbers for going to operator ringgroup and user respectively www atcom cn 22 At last please click on Save button and click on Apply Changes button in up right corner of the main page 3 10 Time Intervals Time Intervals defines ranges of working time that will be used by call routing features Please select the Time Intervals option from the vertical menu on the left of the main page then you can get the following screen Time Interval Name Time Intervals are defined ranges of time that will be used by call routing features Click on New Time Interval button on the illustration above the following screen is where you create and set up time interval Hew
47. owing components analog hardware tone region advanced settings Analog Hardware When you boot the IP08 which will detect the FXO and FXS modules automatically the analog hardware component displays the modules which are detected correctly Tone Region You should select the tone region according to your country if it does not have your country s name in the dropdown list please ask your service operator which kind of tone region is used in your area 3 3 Trunks To receive calls from PSTN and make calls to the outside world you have to use trunk Please select the Trunks option from the vertical menu on the left of the main page then you can get the following screen www atcom cn 12 System status Nanage Analog trunks Configure Hardware Trank Analog Trunks Service Providers VOIP Trunks T1 ET BRI Trunks Trunks are outbound lines Hew Analog Trunk used to allow the system to make calls ta the real world Trunks be wolIPR lines No Analog Trunks Defined traditional telephony lines 3 3 1 Create Analog Trunks Analog trunk is associated with FXO port and it will call outside by PSTN line Click on New Analog Trunk in the illustration above the pop up screen is where you create and set up trunk Edit Analog Trunk Xx Channels wa Irunk Name D trunk CallerID Normally you should not have to adjust your analog ports beyond the ini
48. pdate After inputting your new password please click on Update button then click on Apply Changes button on the up right corner of the main page Factory Reset it will help you to recover to the default factory settings Click on Factory Reset button the following illustration will be presented below General Preferences Language Change Password Factory Reset Reboot Advanced Options Reset to Defaults Please click on Reset to Defaults button to recover to default factory setting then click on Apply Changes button on the up right corner of the main page Reboot you can click on Reboot button Reboot Now button to reboot your system Advanced Options in default IPOS web page hides several advanced options in the vertical menu on the left 1f you need to use them you have to display the options by clicking on Show Advanced Options in the following illustration General Preferences Language Change Password Factory Reset Reboot Advanced Options Show Advanced Options After click on Show Advanced Options in the illustration above you can see the advanced options in the vertical menu on the left of the main page like the following www atcom cn 36 a ATCOM Options Admin Settings Ci Asterisk Logs Editor La Asterisk CLI lAs Settings L1 SIF Settings L1 Network Settings La Firmware update L1 Call Detail Records Peta 3 21 Asterisk Logs After click on Opt
49. r Example Trunk siptrunkl Music On Hold Time Interval Pattern Destination cJ Call Queues Trunk trunk2 Time Intervals Time Interval Destination Incoming Calling Rules Create modify prioritize and delete incoming call rules based on Time Intervals Click on New Incoming Rule button on the illustration above the following screen is where you create and set up time interval Hew Incoming Rule Trunk trunk ka Time Interval timeintervall wt Pattern 8 Destination VaoiceMenu voicemenul G3 Cancel Update Trunk select trunk for incoming call to use I select trunk2 I set up before Time Interval determine the time when the incoming call rule works I select timeintervall I set up before Pattern match the destination number I use S which will match any destination number Destination I select voicemenul so the call will be ruled to voice menu At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 24 3 12 Voicemail When you call someone who does not answer the call you can leave a voice message for the called party if the called party supports voice mail Please select the Voicemail option from the vertical menu on the left of the main page then you can get the following screen General 1 Settings IN General Settings Email Settings for Vo
50. r Directly from Different Network Segment Take the user 6020 6005 and 6001 for example I will configure router users and IP08 then the three users can call each other directly 1 Set up router From the web page of your router please configure the IP address subnet mask and default gateway of WAN port I configured a static IP Address 172 16 1 1 Subnet Mask 255 255 0 0 Default Gateway 172 16 1 254 You can refer to the following www atcom cn 51 l INKSYS A Division of Cisco Systems Inc Setup Internet Setup Internet Connection Type Optional Settings required by some ISPs Setup Basic Setup Security Applications amp Gaming DONS Address Static IP IP Address Subnet Mask Default Gateway static OMS 1 Static DMS 2 Static DMS 3 Hast Domain on rp em PIECE EN 2 PIE HOO Enable 9 Disable Size Etherfast Cable DSL Router Administration ATCOM AAA Firmware Version 1 05 00 BEFSR41 V3 Status Advanced Routing Basic Setup The Basic Setup screen is canfiguratian is perfarm internet Service Providers will require that you enter the DANS information These settings can be obtained from your ISP After you have configured these settings should set a router from the Completing the Internet Setup section is all that is requ
51. reen is where you create permitted to dial long distance numbers and so would have a Dial Plan that includes both the local and longdistance outgoing calling rules and set up dial plan Create New DialPlan DialPlan Name DialPlanl Include Outgoing Calling Rules M outgoingl Include Local Contexts default M parkedcalls V conferences V ringgroups V voicemenus V queues V voicemailgroups v directory Cancel Save DialPlan Name a unique label to help you identify the dial plan when listed in user component you have to set up a dial plan name and select outgoing call rule and local context that you want to use 3 6 Users Users component is used to add or remove Analog SIP AX extension Please select the Users option from the vertical menu on the left then you can get the following screen C37 System Status User Extensions on s X Delete Selected Users No users created Users is a shortcut for quickly adding and removing all the necessary configuration components for any new phone www atcom cn 16 ATCOM 3 6 1 Create SIP IAX User Click on Create New User button on the illustration above the following screen is where you create and set up user Edit User Extension 6003 General Extension 6003 0 Name 6003 DialPlan CallerID 6003 D outbound CallerID 6003 D Enable Voicemail for this User
52. rrors 0 dropped 0 overruns 0 carrier 0 eollisions 0 txqueuelen 1000 bytes 14130939 13 4 MiB TH bytes 1815763 30 3 MiB Interrupt 48 Link encap Local Loopback inet addr 12T 0 0 1 Mask 255 0 0 0 UP LOOPBACK RUNNING MTU 16456 Metric 1 Ri packets 2942 errorz dropped 0 overruns 0 frame 0 packets 2942 errors 0 dropped 0 overruns 0 carrier U eollisions O txqueuelen bytes 1272712 1 2 MiB TX bytes 1272712 1 2 Disk Usage Information System Information M General Network Disk Usage Memory Usage Disk Usage Filesyet em lk blocks Used Available Mounted on dev mtdblockl 14327 1 38T4 455 OTH dev mtdblackz 253952 177992 10 persistent www atcom cn 32 Memory Usage Information System Information Network Disk Usage Memory Usage Usage total shared buffers 45978 41504 0 812 3 18 Backup Backup and Restore are two of the mandatory functions of any application 8 is no exception Customers can backup all the files under the etc asterisk directory and restore them Please select the Backup option from the vertical menu on the left of the main page then you can get the following screen Backup Restore Configurations List of Previous Configuration Backups Previous Backup configurations found Please click on the Create New Backup button to take a backup
53. ry D can Waiting D LJ err Is Agent D Pickup Group G Cancel Update At last please click on Update button and click on Apply Changes button in up right corner of the main page Please pay attention to the Technology component there is an Analog Station drop down list I choose port 4 in which port the analog phone plugs 4 1 3 Register a SIP user 6001 in AT610 After logging into the web page of IP Phone AT 610 please select VOIP option I register the 6001 as the following illustration p IP Phone Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage Public SIP Configuation Basic Setting Register status Registered Proxy Server Address NE Server Address 192 168 1 10 Proxy Server Port A Fuse Phone Number Enable Register APPLY Advanced Set www atcom cn 47 After configuring please click on the APPLY button Now you can call each other directly between user 6001 and 6005 4 2 How to Make a Call to Outside through PSTN In order to dial out to PSTN with IPOS you need an analog trunk an outgoing calling rule a dial plan and a user Here I will give the simple configuration steps which show how to make a call to outside for detail configuration you can refer to chapter 3 4 2 1 Create an Analog Trunk
54. s in different countries can be easily combined together to work like a virtual single office through internet Features Open Source Asterisk IP PBX High performance OSLEC Open Source Line Echo Canceller Configurable IVR menu Voice Mail Voicemail to Email Call forward call waiting call transfer Call conference Call queues Ring group SIP trunk IAX trunk PSTN analog trunk Call Detail Record Access via SSH telnet web Firmware upgradable via web page 50 available SIP IA X2 extensions 20 concurrent calls Applications SOHO SMB telephony system Hosted service FAX terminal IVR system Interface 2 RJ45 port 1 Power port 1 MMC SD slot 8 RJII port 5 interchangeable 4 Dual port FXO FXS module slot 1 USB port www atcom cn 4 Hardware CPU 400MHz Blackfin 532 Chip Eight analog FXO FXS module interface NAND flash 256 MB SDRAM 64M System Open Source uClinux Measurement and Weight 225 120 30mm 0 79KG Carton MEAS 456 442 362 mm 21 units CTN G W CTN 18 KG CTN Package IPOS M RS232 module 1 Power Adapter Manual disk UL www atcom cn 5 For the usage of IPOS in VoIP field you can refer to the following network topology Internet Headquarter Branch Office Analog Phone www atcom cn 6 Chapter 2 Access to the IP08 You need a PC to access to IPOS there are four ways f
55. stance 10 digit dials through a low cost SIP trunk You can optionally set a failover trunk to use when rules the primary trunk fails Note that this panel manages only individual outgoing call rules See the Dial Plans section to associate multiple outgoing calling rules to be used for User outbound dialing No CallingRules defined Click on New Calling Rule button on the illustration above the following screen is where you create and set up outgoing calling rule Edit Calling Rule Calling Rule Name outgoing Pattern D 00 R 2 send to Local Destination Destination send this call through trunk Use Trunk Strip 1 digits from front and Prepend these digits D before dialinz 2 Use FailOver Trunk fail over Trunk Strip D digits from front and Prepend these digita D before dialinz The important parameters I configured are below Calling Rule Name a unique label to help you identify the outgoing calling rule when listed in dial plans I use outgoingl as the calling rule name here Pattern it acts like a filter for marching numbers you dialed here I set up _2X it means any number you dial out with prefix 2 will use this outgoing call rule Use Trunk select the trunk for outgoing calling rule here I select the trunk2 I set up before Strip I press 1 here it will strip the first number of the number string you dialed You can get the detail information about
56. system Status System Status IN Upgrade to VoIPtel SE Please click on a panel to manage related features p Uptime 04 14 57 up 10 min load average 0 04 0 06 0 02 Trunks Status Username Port Hostname IP Configure Hardware Extensions Free Busy Unavailable Ringing Extension Hanme Label Status Type 6750 Check 15 VoicelMailMain xNo Extension assigned Dial by Names Directory Call Features 4 1 2 Add up Users from Web Page of IP08 1 Add up a DialPlan Before you add up user you have to add up a DialPlan please click on Dial Plans New DialPlan I add up a DialPlan like the following Create DialPlan DialPlan Name DialPlanl Include Outgoing Calling Rules You do not have any calling Rules defined click here to manage calling rules Include Local Contexts default parkedcalls V conferences V ringgroups V voicemenus 7 queues V voicemailgroups v directory O Cancel Save www atcom cn 45 After configuring please click on Save button and click on Apply Changes button in up right corner of the main page 2 Add up SIP user 6001 After logging into the web page of IP08 please click on Users gt Create New User I configure user 6001 like the following Edit User Extension 6001 General Extension 6001 Nane BOO 6001 D outBound CallerID 6001 Enable Voicemail for this User
57. tial EE calibration Should you still need to fine tune your audio settings please uze the adjustments at the right Advanced Options Busy Detection D Busy Count D Ring Timeout anon Answer on Hangup on Polarity Switch D Polarity Switch Call Progress Ho Progress fone Use CallerID D Caller ID Start As Received Pulse Dial D 3 CID Signalling mailbox Flash Timing Tau Receive Flash Timing G Cancel Update There are many parameters for you to set up I just set the following two parameters Channels select the FXO port you want to use Here I use the port 2 Trunk Name a unique label to help you identify the trunk when listed in outgoing calling rules and incoming calling rules Here I use the trunk2 as my trunk name For the advanced options you can put your cursor on the label you can get the information of the parameter customers have to set these parameters according to your service provider and your need www atcom cn 13 ATCOM 3 3 2 VoIP Trunks A VolP service provider VSP that you have signed up with is also a trunk Via the VoIP trunk you can dial via the VoIP service to reduce your cost when making international calls You can set up the VolP trunk to make calls to the PSTN or other VoIP network depends on the service you use You can also use the VoIP trunk to link headquarter and branch offices for
58. tte ek VES e ases eeu des 42 327 CME DETAIL RECORDS PN 43 CHAPTERS AN APPLICATION CASE OF IPS ooi E eure E 44 4 1 HOW TO MAKE INTERNAL CALLS THROUGH Pent bak a Fox tUa Ires tano 45 4 1 1 Access to the Web Page of IPO8 by BroWse Nt 45 4 1 2 Add up Users from Web Page of1PO8 Nt 45 4 1 3 Register a SIP user 6001 in AT610 47 4 2 HOW TO MAKE OUTSIDE THROUGH PS TIN eek VE Ehe ivi 48 421 Creole on Analog TUNK avec md a Ree 48 4 2 2 Create an Outgoing Calling Rule 49 42 C ete q enata aaa at tet UM eq aM aM IM 49 50 4 3 HOW TO AN INCOMING CALL FROM OUTSIDE Ne 50 4 3 1 Create an Analog JrUnK senses nahen 50 4 3 2 Create an Incoming Calling 50 SA gt i VOCE uc m ee 51 4 4 HOW TO CALL EACH OTHER DIRECTLY FROM DIFFERENT NETWORK SEGMENT eene 51 4 5 HOW CALE THROUGH VOIP T RUDI tab tatu 55 4o L CO Pu RED T 55 58 4 6 HOW TO TRANSFER FILES BETWEEN WINDOWS PC AND IPO
59. ulk Add Create New users from CSV list Create a Range of new users Click on the Create a Range of new users button in the illustration above the following screen 15 where you create bulk users Create New users from CSV list Create a Range of new users Create Users Starting from Extension 6100 Create Users Tip Use the Selected Users button from the Users page to edit options for the created users Here I want to create five users and the extensions starts from 6100 so I select 5 in the Create drop down list and I set 6100 in the textbox of User Starting from Extension www atcom cn 38 http 4 192 168 1 151 Users added Click Ok to reload GUI At last click on button in the pop up screen then click on Apply Changes button on the up right corner of the main page Please select the System Status option in the vertical menu on the left of the main page you can see you have added five users 6100 6101 6102 6103 6104 3 23 File Editor After click on Options gt Advanced Options gt Show Advanced Options please select the File Editor option from the vertical menu on the left then you can get the following screen From the drop down list of config files you can select the file you want to edit or read P F Contig Files extensions conf followme conf meetme conf users cont ziscan cont rapscan cont asterisk conf queues cont applyzap conf gu
60. vailable Users 6003 SIP 6003 6001 IF 6001 6002 SIP 6002 6001 2 6001 Ring Group Options Strategy seconds to ring each member 20 If not answered Goto 3 Cancel Save Set the ring group name and extension for the ring group select ring group members from available users Select strategy for ring group Ring in Order when someone calls the ring group the ring group member will ring in order Ring all simultaneously when someone calls the ring group all of the ring group member will ring at the same time If not answered Goto choose a destination from the drop down list when no one in the ring group answers the call At last please click on Update button and click on Apply Changes button in up right corner of the main page www atcom cn 20 1N ATCOM 3 8 Call Queues Please select the Call Queues option from the vertical menu on the left of the main page then you can get the following screen Queues IN Queues Agent Login Settings C E sc i CSOeoV D EO ODA EELX dX OOALEL92 2J2DaAEKC l 4 Create New Queue No Call Queues defined La Ring Groups r Music On Hold Call Queues Call queues allow calls to be sequenced to one or more agents Click on Create New Queue button on the illustration above the following screen is where you create and set up call queue Edit Queue 6500 Extension 6500 Name queuel strategy Muzic Hold Le
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