Incoming calls not working

I have outgoing calls working but I cannot get incoming calls working. I have it working in Freebeepbx using sip & not pjsip.

In Freepbeex my sip setting for incoming are
User Context = from-trunk
Register String =


From here Grandstream GXP Series Setup Guide – Goldfish Telecoms Help Centre it says use and sbc will only work in VitalPBX. If i use in VitalPBX outgoing does not work so I presume that is related to sip and pjsip.

Here is my settings in VitalPBX

Clean the fields:

  • Contacts
  • Server URI
  • Client URI

Those values are automatically generated by the system. Additionally, the format of those values are incorrect in your settings. So, Let the system to generate then for you!

Ok calls are now reaching my server but are rejected

[2023-11-04 17:25:28] NOTICE[7577] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“RDS” <sip:0851******>’ failed for ‘’ (callid: 08d540ac02e6e3c52755c9853267135a@sbc.soho66.c - Failed to authenticate

“RDS” is the name of my account in Goldfish

I do not know who are?

Here are my trunk settings

Goldfish & Soho66 are same company I think. The Soho66 website is identical to Goldfish’s old website.

Use sngrep via ssh and look what happens.
Maybe different number format on incoming?

You can also leave CID and DID empty for a catch all filter.
Then use Incoming route Module with Verify DID. Make an incoming call.

I get a 484 Address Incomplete. Ougoing uses and incoming is useing


  • Looks like and resolve to the same IP: It seems like they updated the domain on their end.
  • Try adding as well as ``` to the match field and try again.
  • If it doesn’t work, please share a screenshot from your updated trunk configuration.
  • Also, in the Asterisk CLI, run pjsip set logger on, reproduce the issue, upload the output to and share the logs here.

A couple of more questions:

  • What is the DID that is being called? It looks like the INVITE is to s
  • It looks like Goldfish is using a VERY outdated version of Asterisk… Do the have support that can guide you on how to setup a simple PJSIP Trunk?

Hi Here is logger

sip -

I also found this. Not sure if its related but sounds similar.

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