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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 …

No audio on Asterisk SIP call - Stack Overflow

    https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
    No audio on Asterisk SIP call. Ask Question Asked 10 years, 10 months ago. Active 1 year, 10 months ago. Viewed 34k times 9 I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not ...

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

Bridges - Asterisk Project - Asterisk Project Wiki

    https://wiki.asterisk.org/wiki/display/AST/Bridges
    A local native bridge occurs when the media between two channels is handled by the channel drivers/protocol stacks themselves, but the media is still sent from each device to Asterisk. In this case, Asterisk is merely proxying the media back and forth between the two devices. Most types of native bridging in Asterisk are local.

asterisk: IP address order may cause no audio · Issue …

    https://github.com/irontec/ivozprovider/issues/511
    Kaian commented on Jun 4, 2018. Hi @manfer. If I understand correctly your last note, the order of the addresses in configured interfaces may lead to no-audio scenarios. If that's the case, it may be a bug related to asterisk multihomed module, which has been reworked in the latest asterisk 13 versions.

No Sound on External SIP in Asterisk|FreePBX

    https://www.sysfix.co.uk/Blog/No-sound-on-external-sip-asterisk.html
    If you have No Audio - It's to do with your ROUTER or FIREWALL. If you've configured the ports as I've shown above, there are only a few things left to check. 1. Internal firewall. If you're using a soft-phone on a machine, make sure the firewall is disabled for testing.

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