Archives : December-2017
Im having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following example (sorry for its length):INVITE sips:bob@biloxi.example.com SIP/2.0 Via: SIPS/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bK74..
I am using AMI to issue a BackgroundDetect on a channel.Everything works great, I receive the result and the variables on the channel.I am running into one issue though. After calling that function on AMI, when I send the next command on AMI for t..
Briefly: I want to be able to have press or say (number), with Asterisk listening for a spoken number, but accepting a DTMF digit, too. Im posting everything I found so far, here, partly to show working, but also in case anyone else finds it usef..
I carefully read [1] which details how backtrace files can be produced.Maybe this seems natural to some, but how can I go one step futher, and check that produced XXX-thread1.txt, XXX-brief.txt, … files are OK ?In other words, where can I find an exam..
Everyone, How to configure PJSIP to reply 200 OK from upstream sip proxy on keepalive packet ?proxy ~> Keepalive OPTIONS ~>..
I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation. No matter what I try I alw..
The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15.The available security releases are released as versions 13.13-cert8, 13.18.3,14.7.3 and 15.1.3.These releases are available for immedi..
The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15.The available security releases are released as versions 13.13-cert8, 13.18.3,14.7.3 and 15.1.3.These releases are available for immedi..
Hello!Im currently using Asterisk 11 (due to the fact that Debian Wheezy has Ast 11 in backports – so that I can have security updates from my distribution).I recently played a little with Asterisk to be able to hear internet-radios over the local pho..
I want to disable autoload in modules.conf and only enable the necessary modules for voicemail.I am using ODBC to connect to a MySQL database for authenticating voicemail users.Please let me know which modules are necessary only for the voicemail funct..