One way audio in SIP trunk calls - Freepbx 17 on Ubuntu 22.04

FreePBX Endpoints
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1

Hello Everyone.

I just did a fresh installation of Freepbx -17 with asterisk 21 on Ubuntu 22.04.

Installed it from scratch successfully, was able to upload extensions and setup SIP trunks.

Extension to extension calls work fine with audio both ways,

SIP trunk connectivity to another (bare) asterisk server with asterisk 12.8.0 is connected both ways, and inbound and outbound calls are successful. However, for this sip trunk calls we have the issue of oneway audio.

We have troubleshooted, put off ufw etc. but one way audio persists. Outside caller can hear people in the pbx, but those behind the pbx cannot hear anything from outside.

The SIP trunking is configured IP peer to peer trunk with the PBX in a LAN behind a NAT so the Public IP is forwarded to the local IP.

Is there any one who has had this kind of issue before? Bear in mind that asterisk 21 uses pjsip, while asterisk 12 uses sip. Is that supposed to be an issue? Any help would be appreciated. Thank you

2

With reasonable settings, you would expect the two implementations of SIP, from the same supplier, work well together.

You need look at the SDP and make sure the c= lines are correct, in both directions, or provide us with the SIP logs, so we can check that.

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For clarification, The calls you are having problems with, are they only outbound callas or inbound too ? Are they involving cell phones or land lines ? Have you narrowed it down to any particular provider ? We are in Hawaii and are having problems with 1 way audio. But it is only on calls that interact with a T Mobile account. So far we have tested this against Verizon, H2o, AT&T and a few others, but so far 100% of the failed calls (one way audio) are T-Mobile. We also have an open ticket with Sangoma about this.

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Hi
This sounds like a nat issue please make sure you have forwarded rtp ports 10000-20000 including sip port 5060 too and you have set your external ip as public ip under asterisk sip settings also try these too under trunk advance
Rewrite Contact=yes
RTP Symetric=yes
i hope this will be helpful

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