Archives : August-2018
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..