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Distro and PJSIP: No Audio - FreePBX Community Forums
https://community.freepbx.org/t/distro-and-pjsip-no-audio/52675#:~:text=The%20PJSIP%20port%20is%20for%20signaling%2C%20the%20RTP,default%20is%205060%2C%20some%20providers%20block%20this%20port.
No Audio PJSIP - Endpoints - FreePBX Community Forums
https://community.freepbx.org/t/no-audio-pjsip/65068
Normally, Asterisk relays audio between the parties. If A calls B, then A sends audio to Asterisk and Asterisk sends it to B, and vice-versa. However, when possible, pjsip attempts to get the parties to communicate directly. This reduces …
PJSIP no audio on calls - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-no-audio-on-calls/85165
No audio but calls are placed without any errors. I moved to PJSIP because my WebRTC peer wouldn’t qualify and always be unreachable in SIP, but in PJSIP, it is reachable although no audio UPDATE: I turned off direct_media(of 200) and and now I can hear one side audio on WebRTC(100)-Softphone(200).
no audio with asterisk 13 pjsip - Stack Overflow
https://stackoverflow.com/questions/37849325/no-audio-with-asterisk-13-pjsip
Show activity on this post. I cannot playback any wav files from the dialplan or get any audio at all with asterisk 13 pjsip. Server is not behind a nat, endpoints (clients) are. Chan_sip works perfectly just not chan_pjsip pjsip.ini: It seems that the rtp 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.
[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 …
No audio devices are found · Issue #2521 · pjsip/pjproject ...
https://github.com/pjsip/pjproject/issues/2521
Describe the bug Fail to detect audio devices. To Reproduce Launch pjsystest-armv6l-unknown-linux-gnueabihf Expected behavior List of detected audio devices Logs/Screenshots Fail Audio Device Detection.txt Desktop/Smartphone (please comp...
How to get the audio stream from PJSIP when there is no ...
https://stackoverflow.com/questions/46243029/how-to-get-the-audio-stream-from-pjsip-when-there-is-no-audio-hardware-device
I want to use PJSIP's C API to record the incoming audio to a file on a machine with no hardware sound device . I'm unsure about the details, but the sparse documentation for PJSIP suggests it should be possible through the pjsua_set_null_snd_dev() call.
audio - Asterisk 16 LTS & PJSIP; hello world works but no ...
https://superuser.com/questions/1465895/asterisk-16-lts-pjsip-hello-world-works-but-no-sound-coming-from-endpoints
Asterisk 16 LTS & PJSIP; hello world works but no sound coming from endpoints. Ask Question Asked 2 years, 6 months ago. Active 2 years, 6 months ago. Viewed 949 times 2 1. I have recently set up an Asterisk server with version 16.4.0 LTS. I am completely new to asterisk yet I have managed to set up the server with the service and it runs ...
Asterisk PJSIP Troubleshooting Guide - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide
Run the CLI command pjsip show aor <aor name>. If no AoR is displayed, then that means the AoR was not loaded. Ensure that the configuration section has type = aor specified. Ensure that all configuration options specified on the AoR are options that Asterisk recognizes. Ensure that the res_pjsip_registrar.so module is
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