Incoming number from sip provider not correct

I’ve gotten into a dispute with a client of ours. This client is using a SIP provider for some time now and has some blocks of numbers and they are all configured in our Asterisk server and that’s working well.

a few weeks ago the client requested to add a new incoming for a newly formed team. Since they didn’t have any free numbers left they ordered new numbers from the SIP provider.

Now, any of those numbers don’t work and in Asterisk CLI I can see the provider sends an incomplete number so I can’t route it to the correct team. I obviously also get a diconnected tone as well.
Now the provider keep pointing the finger to us, saying all is ok at their side. Pretty strange as all their other numbers (almost 100) function well and show up correctly in the Asterisk CLI. Except any new number the provide us.

I do get a pseudoID that shows the correct number, but the actual SIP string shows the wrong number which we use for the incoming route.

My knowledge doesn’t go far enoug, what is the PseudoID? And what can I do else? IMO I can’t do anything to fix it before the SIP provider does.

– Executing [313@from-trunk-pjsip:1] NoOp(“PJSIP/pjsip-trunk-2-RouteIT-000002f3”, “Fixing DID using information from SIP TO header”) in new stack
– Executing [313@from-trunk-pjsip:2] Set(“PJSIP/pjsip-trunk-2-RouteIT-000002f3”, “pseudodid=“31318886327 31318886327” sip:313@stmaan.2mobileconsultancy.voipit.nl”) in new stack
– Executing [313@from-trunk-pjsip:3] Set(“PJSIP/pjsip-trunk-2-RouteIT-000002f3”, “pseudodid=“31318886327 31318886327” <sip:313”) in new stack
– Executing [313@from-trunk-pjsip:4] Set(“PJSIP/pjsip-trunk-2-RouteIT-000002f3”, “pseudodid=313”) in new stack
– Executing [313@from-trunk-pjsip:5] AGI(“PJSIP/pjsip-trunk-2-RouteIT-000002f3”, “agi://localhost:4574/asteriskCIDLookupScript”) in new stack
– <PJSIP/pjsip-trunk-2-RouteIT-000002f3>AGI Script agi://localhost:4574/asteriskCIDLookupScript completed, returning 0
– Executing [313@from-trunk-pjsip:6] Set(“PJSIP/pjsip-trunk-2-RouteIT-000002f3”, “CALLERID(name)=0651008002”) in new stack
– Executing [313@from-trunk-pjsip:7] Goto(“PJSIP/pjsip-trunk-2-RouteIT-000002f3”, “from-trunk-conversion,313,1”) in new stack
– Goto (from-trunk-conversion,313,1)

Whatever the writer of your dialplan meant it to be! It has no special meaning in SIP and no special meaning to Asterisk. It looks like it may be the user part of the incoming To header, but without the actual dialplan, that is only a guess, based on the intermediate values.

Actually, it is pseudodid, not PseudoID, as case matters for variables, and so does spelling. DID, which really stands for direct in dialling, has been adopted, by the SIP community, a meaning the number the original caller actually dialled.

When providing the dialplan and logs, please do not post them as images; please post them as plain text, taken from the relevant files, and marked up as pre-formatted text.

pseudodid it is indeed! can’t believe I read it as ID!

pls paste your dialplan “from-trunk-pjsip” here

SIP packet capture might be useful as evidence.