AMI generated calls with intermittent One-way audio issue

General Help
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1

I have been looking all over the internet and I can’t seem to find anyone with exactly the same issue, however I did find a forum post here that was similar:

Basically I am using a chrome extension (Jolt) click to call which executes a PHP script passing the outbound number as well as the local SIP extension to ring via AMI script. This had worked for years but in upgrading to a more recent version of FreePBX for the time groups calendar function it is now intermittent, I would estimate approximately 10% of the calls using the AMI script complete correctly and get 2 way audio, however the majority the person on the local SIP phone cannot be heard by the remote party, however the ‘called’ person (remote over SIP trunk) via voicepulse can be heard on the local sip phone. All calls that made from the SIP phones have 2 way audio 100% of the time and there are no issues with RTP time outs or reinvite issues when calling from the phone.

It has been driving my crazy as like any intermittent problem it has been very difficult to troubleshoot.

I do not believe it is an AMI script or configuration issue as the call does complete and 10% of the time works as expected. I think it has to do with bridging the call from the AMI script to the SIP → SIP trunk for some reason most of the time the audio channels are not getting bridged correctly leading to one way audio. The SIP phones / asterisk server are on the same LAN no NAT between them there is a firewall in front of the asterisk box. But I don’t think this can be an NAT issue as ALL calls directly from all 12 SIP phones work all of the time, it is only with AMI scripts are involved that there are these issues.

Ideals? I am losing my mind here.

2

I have ideals but no ideas. First, you have to post a log of a failing call, SIP trace included.
At the Asterisk command prompt, type
pjsip set logger on
and/or
sip set debug on
as appropriate, make a failing call, get the relevant section of the Asterisk log, redact it as you feel necessary, paste it at pastebin.freepbx.org and post the link here.

3

Are both channels outside of a NAT router? Is the full RTP port range being forwarded to the PBX? I like doing this test to rule out a router issue:

channel originate local/xxxx@from-internal application echo

Where the xxx’s are an extension number or a pstn number. When answered, you must be able to hear your own voice echoed back.

4

Thank you for the replies so far and yes. Ideas :slight_smile:

Here is the paste bin link, I hope that I am capturing enough information this from a failed call (one way audio today)

https://pastebin.freepbx.org/view/b7892c9c

… Igaetz I did try that echo test that worked fine I could hear the echo does not appears to be an issue with RTP ports (I have 10000 - 20000 forwarded to the asterisk box and no firewall issues with my upstream voip provider

To be clear the call via AMI originates from a computer on the local network calling a PHP script on the local asterisk box on the same LAN 10.1.10.X / 24 The SIP phone that is called as the internal phone “rings first” is on the same 10.1.10.X/24 lan. Once they answer the call it calls out to the external number via our upstream provider over the internet.

I am hoping this will help.

Thank you so far.

5

I had posted a log were you able to look at that? It appears to me that it is a bridging issue

6

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