Hello,
when I try to make an outbound call via a pjsip trunk via sipgate, the SIP log shows some weird behavior:
INVITE sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP {{local-ip}}:5060;rport;branch=z9hG4bKPj541705dc-fe7a-4edf-9d0b-7817394589c7
To: <sip:{{dialednumber}}@sipconnect.sipgate.de>
Route: <sip:{{dialednumber}}@sipconnect.sipgate.de:5060>
Answer:
SIP/2.0 404 Not found (no match)
As you can see: the dialed number is not present in the first line: INVITE sip:sipconnect.sipgate.de SIP/2.0
The number and the following @ sign are missing!
In the To and Route header, both are as expected.
This leads to an answer: 404 Not found from the provider…
If I take this SIP packet and send it to the provider via the linux netcat command, I get the same error: 404 Not found.
BUT:
If I add the dialed number to the first INVITE line of the packet, so that it looks now like that:
INVITE sip:{{dialednumber}}@sipconnect.sipgate.de SIP/2.0
it works! I get an answer 407 Proxy Authentication Required
I don’t understand, why FreePBX or Asterisk or pjsip or whoever is responsible does not include the number I want to dial in the INVATE header…
Could anyone please help me here…?
Thanks.