Archives : August-2018
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With chan_sip there is the variable SIP_MAX_FORWARDS to set Max-Forwards. This counter is persistant after a redirect. I cant find the equivalent for PJSIP, so I went the way of header manipulation. Only to find out that any headers added to the outbo..
Guys Found the solution for this…! https://wiki.asterisk.org/wiki/display/AST/New+in+12#Newin12-channels_chan_a gent and https://reviewboard.asterisk.org/r/2657/diff/1/ and https://blogs.asterisk.org/2016/02/10/converting-from-chan_agent-to-app_a..
there; Im trying to dial into a Zoom conference, send some digits, wait, send a name, and be in the room, as it were. I thought this would work: same => n,Dial(PJSIP/02036950088@voipfone-205,12,r(callWaiting)D(WWW12345W#WW::)) But it didnt, so I tr..
All With the below config, I just keep gettings this in the Asterisk 13.22.0 CLI: WARNING[15872][C-00000051]: channel.c:6343 ast_request: No channel type registered for Agent whenever a caller gets sent to that agent queue with logged in agents wait..
list, Hope you all doing fine!!I am using chan_sip of Asterisk 13.6.0 and eventually I have a carrier which replies with 400 to some INVITES which happen to timeout…. I know the SIP reply code is not correct, but anyway I want to understand what..
all,I found the volume function. I am wondering if that works for ConfBridge ?The web page mentions works on a channel – but what about adding volume to a ConfBridge.How do I do that?Th..
I sent the requested information. I always get this responde:Response: Success Message: Timeout Set But keep the old timeout, interestingly, decreasing the timeout works perfectly. The problem is increasing.Version Asterisk 1.8.32.$this->setTimeout($ch..
guys, I sent a dial to asterisk with a specific timeout, I want to increase it for some users if it is approaching to the end, but when I send AbsoluteTimeout action and change it timeout I get success but hangup at initial timeout, other words, it doesnâ..
Daniel Thanks for the reply! Yes, turns out it was all my fault, I had a line feed character (0x0a a.k.a printf(\n)) in one of the Asterisk channel variables passed via system() / shell() to my target script. It seems 13.22.0 (Im using the same vers..
Last question for today, I promise! The problem: In order to disconnect calls after x minutes, I need to do this: [setup] exten => setup,1,Answer() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reach..