Strange Outbound call connections

Hi Everyone,
I have just setup a new FreePBX install in our office (FreePBX 13.0.167) and am having a few connection issues with certain phone numbers.

I think it relates to calling other PABX systems as 90% of the phone calls we are making are working without an issue. The problem seems to be when I make a call to these particular numbers, everything appears to work as normal, but there is no phone

I have emailed our client (who runs the other PBX) and he can see the call (in their logs - Avaya IP Office) but their system does not ring, just silence - They are able to call our PABX without an issue.
This situation, ironically, also happens when I try to ring my VOIP providers support number.

I have looked through the log and the issue appears to only happen when this is shown in the log

-- Channel SIP/EnginPace-00000005 joined 'simple_bridge' basic-bridge <0c8cc13e-8954-4213-823f-9ab6e60f1d14>
-- Channel PJSIP/409-00000004 joined 'simple_bridge' basic-bridge <0c8cc13e-8954-4213-823f-9ab6e60f1d14>

All the other successful calls don’t appear to have this message.

I am hoping its something I have missed in the settings as our old system did not have this problem (it was running FreePBX 2)

If someone could point me in the right direction to get this fixed, that would be fantastic.

Thanks in advance.

with the information you have provided it would be fantastic if anyone could point you in the right direction

I was hoping that somebody may have seen this issue and could offer some assistance. I could post log entries and settings all day long but I don’t know which parts are relevant and which are not.

You are bridging a SIP connection to a PJ-SIP connection, so this is perfectly normal. Your connection to EnginPace is going out over a SIP connection and your phones are connecting via PJ-SIP. This would be the same message you would get with a DAHDI or SCCP connection to a SIP.transport.

There are many things that can jam you up, but the one I would look at first is codec interconnection. I’d guess (without any more information than we have) that you are failing in codec negotiation.

Hi Dave, That is exactly what was happening - Instead of chasing my tail I have just switched all the handsets back to a SIP connection and everything is working as normal. I originally went with PJSIP because all my phones were already configured to port 5060 (CHAN_SIP on our new Asterisk is 5061)

Thanks for the input, much appreciated.