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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>

...>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>

2017-06-08 12:54 GMT-04:00 Javier Marcuzzi <javier.ruben.marcuzzi en gmail.com> : > ¿me hice entender? ?No. Para salir del escollo lo convertiré a gráficos? base y continuaré con mi vida ?... ?Au revoir.? -- «Pídeles sus títulos a los que te persiguen, pregúntales cuándo nacieron, diles que te demuestren su existencia.» Rafael Cadenas [[alternative HTML version deleted]]

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>

...or the research project on multilevel logistic regression. There is confusion about bugs() function in R and BUGS software. Is there any relation between these two? Is there any comprehensive package for Multilevel Logistic modelling in R? Please guide in this regard. Thank You RAZA --------------------------------- Boardwalk for $500? In 2007? Ha! [[alternative HTML version deleted]]

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

...outube.com/watch?v=MID2RvgdD7s http://www.youtube.com/watch?v=_LbDUdAGfSY http://www.youtube.com/watch?v=J9Chkrg7E-M http://www.youtube.com/watch?v=MsC12wc9ZnU http://www.youtube.com/watch?v=Y12exIN1soY More You can find here http://www.youtube.com/supertecacademy Thank You -- Regards, Zuhair Raza Asst. Manager Technical Support Phone: +1-850-433-8555 ext 101064 Website: www.didx.net Skype: zuhairraza What is DIDX.net? <http://www.youtube.com/watch?v=mIgGTGkTZns> <http://www.youtube.com/watch?v=mIgGTGkTZns> -------------- next part -------------- An HTML attachment was scrubbed...

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();

...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

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I have P4 2.50GHz RAM 4GB, Asterisk how many calls handle using H.323 to SIP conversion on this server? Regards, --------------------------- Muhammad Asif Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090210/6d5cb26b/attachment.htm

...oblems like: when i connect my analog phone with the card there is no dial tone, but i can dial any extension... but after that i can't hear any voice from my receiver i have used different phone sets but still i cant communicate with other extension. Please help me out. Thank you Regards Ali Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101023/9cc54b4a/attachment.htm

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

Hi, I need help to build an email server using LDAP phpldapadmin,postfix,dovecot,spamassin,amavis,claimav and horde webmail. please share documents related to it. Thanx Musawwir

...before T.38 ports are configured when T.38 is found but no audio stream is found. (Closes issue #16318. Reported by bird_of_Luck. Tested by vrban, mihaill. Patched by vrban, mnicholson.) * Avoid crashes with large numbers of MeetMe conferences. (Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Change in 'sip show channels' display format allowing more digits for CID. (Closes issue #16459. Reported, Patched by Rzadzins. * Revise documentation on disposition values to the actual values used. (Closes issue #16289. Reported by wdoekes.) A...

...377, #16376. Reported by bcnit. Patched by dant. * If EXEC only gets a single argument, don't crash when the second is used. (Closes issue #16504. Reported by bklang. Patched by tilghman.) * Avoid a crash with large numbers of MeetMe conferences. (Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces (for Solaris 10). (Patched by seanbright.) * Allow "REMAINDER" to function properly in expressions. (Closes issue #16427. Reported, Patched by wdoekes.) * Shut down the SIP...

...# "h" (always), as the partcipant's phone does not hang up. So, when the following command is issued meetme kick h 1 the participant's phone finally hangs up. I don't know why is this behaviour shown by the meetme module ? Could anybody help me in this regard ? Thanx, Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070608/f5f953a7/attachment.htm

...before T.38 ports are configured when T.38 is found but no audio stream is found. (Closes issue #16318. Reported by bird_of_Luck. Tested by vrban, mihaill. Patched by vrban, mnicholson.) * Avoid crashes with large numbers of MeetMe conferences. (Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Change in 'sip show channels' display format allowing more digits for CID. (Closes issue #16459. Reported, Patched by Rzadzins. * Revise documentation on disposition values to the actual values used. (Closes issue #16289. Reported by wdoekes.) A...

...377, #16376. Reported by bcnit. Patched by dant. * If EXEC only gets a single argument, don't crash when the second is used. (Closes issue #16504. Reported by bklang. Patched by tilghman.) * Avoid a crash with large numbers of MeetMe conferences. (Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces (for Solaris 10). (Patched by seanbright.) * Allow "REMAINDER" to function properly in expressions. (Closes issue #16427. Reported, Patched by wdoekes.) * Shut down the SIP...