search for: zohair

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130215/aaab72a7/attachment.htm>

...0000000>AGI Rx << GET VARIABLE CDR(answer) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011-12-16 18:34:50) In 1.8.6.0, there was no end time and in the other two it's present but neither in correct format nor exact time. Is it something related to system or a bug? Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111216/5876b6e1/attachment.htm>

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 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, 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();

...Mar 6 12:11:15 localhost abrtd: Executable '/usr/sbin/asterisk' doesn't belong to any package Mar 6 12:11:15 localhost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1 *Asterisk was running as root user Any suggestions? Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130306/e64955c4/attachment.htm>

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

...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 -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111218/d18b6482/attachment.htm>

Hi team, On event of no answer in CDR the starttime and endtime of call remains the same. Is there any way how can actually track call originate time and call end time. Thanks Vinod dharashive. Sent from BlackBerry? on Airtel

...erisk 530u IPv4 19211854 0t0 TCP 172.20.255.40: sip->10.100.157.32:49227 (ESTABLISHED) sip show settings Global Settings: ---------------- UDP Bindaddress: 172.20.255.40:5060 TCP SIP Bindaddress: 172.20.255.40:5060 Anyone has idea what could be the reason? Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140820/e28c5fa5/attachment.html>

Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? Regards Bilal

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",