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No Audio PJSIP - Endpoints - FreePBX Community Forums

    https://community.freepbx.org/t/no-audio-pjsip/65068#:~:text=As%20a%20quick%20test%2C%20set%20all%20the%20Recording,does%20give%20you%20audio%2C%20add%20this%20to%20%2Fetc%2Fasterisk%2Fpjsip_custom_post.conf
    none

[pjsip] Audio problem

    https://pjsip.pjsip.narkive.com/PZQgcBbG/audio-problem
    pjproject/pjsip-apps/bin I can play an audio file using the commands:./pjsua --clock-rate=48000 --play-file=[file.wav] Then I issue the command cc 1 0 And I hear the sound file playing. However I noticed this as output: pjsua_app.c ..Turning sound device ON This is something I don't notice in my own application, there the only output I see is

Distro and PJSIP: No Audio - FreePBX Community Forums

    https://community.freepbx.org/t/distro-and-pjsip-no-audio/52675
    The PJSIP port is for signaling, the RTP ports (which you aren’t changing) are for audio. If you were unable to register then sure, the PJSIP port didn’t work. But you’ve both said you made calls and there was no audio. Therefore PJSIP signaling port is working. Considering PJSIP’s default is 5060, some providers block this port.

No Audio PJSIP - Endpoints - FreePBX Community Forums

    https://community.freepbx.org/t/no-audio-pjsip/65068
    If you still don’t get audio, report whether dialing *43 (echo test) from each of the extensions works. If setting Force does give you audio, add this to /etc/asterisk/pjsip_custom_post.conf disable_direct_media_on_nat=yes. and if that doesn’t work, try direct_media=no. Restart (not just reload) Asterisk after changing the config files.

[SOLVED i.e. the problem never existed...] PJSIP with no ...

    https://community.asterisk.org/t/solved-i-e-the-problem-never-existed-pjsip-with-no-audio-on-out-and-incoming-calls/78546
    Hi! I started to use pjsip to connect to a SIP trunk (German Telekom). Internal calls are no problem. In- and outgoing signalling is also no problem, but there is no audio with these calls. The trunk is a bit special as it uses tcp for SIP, a proxy, and requires NAPTR and SRC to work for the name resolutions. My PBX is a bit special as it is sitting on the router and it sees the …

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