[asterisk-users] calling peer from server
Hi everybody, This is the console output of the asterisk server. debian-te410*CLI sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have tried diff ways but can't seem to get it work. Can please some one suggest me anything in this regard. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calling peer from server
does that phon has a static IP? does it register with the server? posting your SIP.con and extensions.conf related to this issue could help us to understand what you are doing. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: niksingha...@gmail.com Date: Mon, 14 Jun 2010 17:49:37 +0530 To: asterisk-users@lists.digium.com Subject: [asterisk-users] calling peer from server Hi everybody, This is the console output of the asterisk server. debian-te410*CLI sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have tried diff ways but can't seem to get it work. Can please some one suggest me anything in this regard. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] calling peer from server
Thanks for the reply. Actually my problem is not related to sip.conf and extensions.conf. I have used only standard files from martin pdf which are given as example. I am able to call some system connected over LAN, when each has a softphone and are registered on a asterisk server. But now what i want is instead of using the softphone I write a function in my file which will be executed when the call is placed. In that file i wrote $agi=new AGI(); $agi-exec_dial(SIP,2002,NULL,NULL,NULL); I have used the exec_dial function found in phpagi.php . It is built above basic dial function. Here 'agi' is an instance of class AGI, which has a method exec_dial. When i execute the php file, over command line on my unix machine, I am expecting a call on the softphone which I have registered on the asterisk server. For ip clarification: all have static ip: server ip:172.26.48.208:5060 i have configured twinkle as a softphone client on 172.26.48.113:5061, since on the asterisk server cli when i use 'sip debug set peer 2002' it shows the registered ip, as i said earlier, but again on 'sip show registry' no value is displayed. I don't know what's going on. In the softphone I give domin information as the ip of the asterisk server i.e. 172.26.48.208. In the softphone it shows registration successful. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
