Asterisk One Way Audio Sip Trunk

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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 :

Asterisk one way audio issue | DIDforSale

https://www.didforsale.com/asterisk-one-way-audio-issue

asterisk sip trunk provider. Some of our key SIP Trunking features:- ... The Second possible reason for causing one way audio could possibly be Codec, This often happens when a call comes in with ULAW and the system tries to accept with other codecs which can cause superfluous codec negotiation. To avoid this we need to remove the unwanted ...

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 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 …

How to fix one way audio in asterisk free pbx easily

https://asteriskneeds.com/one-way-audio-asterisk-free-pbx/

If you don’t have Web access to PBX, you can do it by SSH, SSH your server run these commands cd /etc/asterisk vi sip.conf look for [general] and type externip = 111.111.111.111 replace 111.111.111.111 with your Static IP. save it and run service asterisk restart That is all, your one way audio should be fixed now.

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