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One way audio on our TrixBox - ThinkTel
https://support.thinktel.ca/kb/article/46-One_way_audio_on_our_TrixBox
One way audio on our TrixBox. Running a TrixBox behind a firewall can cause one way audio issues. On a Trixbox, you will need to specify the external IP as well as the internal network. in sip.conf. ie: [general] externhost = asterisk.company.com externip=123.123.123.123 ; the ip address your ISP gives you
Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
Connect the ATA/IAD or other SIP device directly to the first device on the LAN such as the modem. Check for normal operation and good two-way audio. If you experience one-way audio or do not receive dial tone, then the issue will most likely be SIP unfriendly NAT, or a firewall present that is not allowing the correct VoIP packets to cross.
One way audio during outgoing calls - Asterisk SIP ...
https://community.asterisk.org/t/one-way-audio-during-outgoing-calls/91154
One way audio during outgoing calls. rgi January 7, 2022, 10:10am #1. Hello, we are using an asterisk system which is running on version 18.8.0. During some outgoing calls we are experiencing the issue of one-way audio. We can hear the callee, but the callee can not hear us. We then took a capture of a flawed call and noticed that the issue ...
[SOLVED] Remote extension, one-way audio on sip client ...
https://community.freepbx.org/t/solved-remote-extension-one-way-audio-on-sip-client/65376
If you still have trouble, at the Asterisk command prompt, type sip set debug peer 1234 (replace 1234 with the extension number of the Linphone user, with Linphone configured without STUN) then make a test call that fails with one-way audio. Paste the relevant section of the Asterisk log at https://pastebin.freepbx.org and post the link here.
One way audio problem in trunk - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/one-way-audio-problem-in-trunk/72567
Hello. I got some problem for Asterisk SIP trunk. It’s Asterisk 11.17.1. SIP trunk from an operator. Outgoing call : Everything is ok Incoming call : Call signaling is ok but audio RTP is one way. Peer A,B and C can send audio packet to Peer X but it isn’t able to receive audio packet from Peer X. Network :
sip - Asterisk one way audio - Stack Overflow
https://stackoverflow.com/questions/55643399/asterisk-one-way-audio
Trying to call from a sip client to a normal phone or exetension. This results always in a one way audio connenction. I use the odbc database, and can't really find the problem. Can anybody help me in the right direction. There seems to be no errors at all. Have tried several things, and searched on the net, coudn't find the correct solution.
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