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No audio when bridging two trunk calls - Asterisk SIP ...
https://community.asterisk.org/t/no-audio-when-bridging-two-trunk-calls/85714
Hi all, I am facing one weird issue with asterisk. I am using manager api to originate calls. I am successfully able to do that between one direct channel (on webrtc) and mobile number (through my sip provider) while everything works perfectly (able to hear voice, can end the call etc). But when I am trying to do the same between two mobile numbers (through …
Calls with no audio in both sides - Asterisk SIP ...
https://community.asterisk.org/t/calls-with-no-audio-in-both-sides/73244
I’m trying to implement a new service on my Asterisk server network. We have a 3G mobile phone with a softphone app - Linphone, whose number is 9012 and is registered on the Asterisk server with IP # 10.192.124.101 (Sagres). Then there is a Trunksip between this server and another one with IP # 10.192.230.231 (Viriato). The Mobile is associated with a Mobile …
No audio and no rtp traffic - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/no-audio-and-no-rtp-traffic/69420
I have installed magnusbilling on asterisk 11 and sometime no one of both end hear something, sometime I issued one way audio and sometimes I’m able to talk normally. In this third case i can see the output in “rtp set debug on”, otherwise no. ... – Locally bridging SIP/VOIPTEST-000000b8 and SIP/GO_VoIP_1-000000b9
No audio on Asterisk SIP call - Stack Overflow
https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
Call redirection: no audio depending on target number ...
https://community.asterisk.org/t/call-redirection-no-audio-depending-on-target-number/87940
My setup is as follows: Asterisk 13.38.1 (PJSIP) behind a router SIP from Telekom Germany IP desk phone (Snom D375) connected to the Asterisk that does call forwarding In certain cases, call forwarding does not work: the call is signalled as successfully established, but no audio is transferred. This depends on to which (public) number number the call is …
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