search for: engineerzuhairraza

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 0000000000533c19 sp 00007f7db9ce3af0 error 4 in asterisk[400000+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Hi List, Which will be the appropriate variable to get called peer IP address? I tried following channel variables peerip, recvip, URI, from and following SIP channel variables: SIPURI,SIPDOMAIN They all return calling peer IP but not the destination/called peer IP. unfortunately set(CDR(calledip)=${CHANNEL(to)}) doesn't work Regards, Zohair Raza -------------- next part --------------

Hello all, I am running asterisk on VMs with standby heartbeat configuration, Heartbeat assigns a virtual IP 172.20.255.40 on machine afterwards asterisk is started. In the sip.conf, I have explicitly define bindaddr=172.20.255.40 but sometimes I see packets coming from physical IP 172.20.255.41 I have both tcp and udp transport enabled Here is the lsof -ni :5060 output asterisk 2878 asterisk

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(start) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011-12-16 18:34:48) <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(end) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011 12-16 18:34:48)