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Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
SIP transformations are known to corrupt some of the SIP headers resulting in issues with the transfer of the voice traffic correctly. Follow these basic steps to isolate the cause of one-way audio: Connect the ATA/IAD or other SIP device …
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.
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 Trunk, No NAT, One Way Audio - Asterisk Support ...
https://community.asterisk.org/t/sip-trunk-no-nat-one-way-audio/33583
Hi, Asterisk 1.6.2.11. SIP trunk from an operator. Outgoing call : signal is OK, audio is only one way. I can hear the distant person, but she can’t hear me. Networking : [IP Phone] ------- [Asterisk] ------- [VoIP Provider's Router] ------- [VoIP Provider's PABX] 192.168.4.107 eth0 : 192.168.4.1 eth1 : 37.1.1.1 37.1.1.2 212.39.140.250 pabx*CLI> sip show peers altitude-output …
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 ...
One way audio help - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/one-way-audio-help/88526
I have a FreePBX 15.0.16.38 and asterisk 16.15.0, I have pjsip extensions and a sip trunk, randomly when I call, I have only one way audio, they hear me but I don’t hear anything. I thought it was a nat problem and I have configured the trunk as follows. type = peer secret = ECOxxxxxxxxx qualify = yes progressinbound = yes nat = force_rport, comedy insecure = port, …
One-way audio during outbound calls - Asterisk SIP ...
https://community.asterisk.org/t/one-way-audio-during-outbound-calls/82347
Hello All, I have an Asterisk 16 installation which is running behind a TD-LTE modem-router. I have a DID from my ISP which is configured as a SIP trunk (chan_sip). I have set RTP port range to 7000-20000 in Asterisk also have forwarded this port range in my router. Incoming calls from trunk work well. However when dial an outside number, only outgoing …
PJSIP One Way Audio - Asterisk SIP - Asterisk Community
https://community.asterisk.org/t/pjsip-one-way-audio/84926
Hi, After spending an hour on the phone with my providers, it seems that I have a problem with my settings for my endpoint or my transport. Where ? we did not find… So we have a concern for one way audio. If I call an external number from my asterisk, the person hears me, but I cannot hear the person. Config : [transport-udp] type=transport protocol=udp bind=0.0.0.0 …
Pfsense and Asterisk….one-way audio problem with SIP ...
https://forum.netgate.com/topic/465/pfsense-and-asterisk-one-way-audio-problem-with-sip
This shows up on voicemail as well...Asterisk hangs up immediately as it cannot hear any audio. The inbound caller can hear me speaking. My thoughts are that this is the classic 'one-way audio over SIP' problem, which can be found on certain firewall types/configurations ie symmetric firewalls. The static port feature may be the way around this.
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
SIP with NAT or Firewalls. 1.1. Problem Description: Most conventional voip protocols (SIP, h323, …) are not programmed with NAT in mind, on itself they only carry call signaling (call setup, teardown,… and use RTP to carry the audio samples. The signaling usually uses fixed and standardized ports, but the RTP uses random ports to exchange ...
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