I have never worked with ITSP VOIP service before. SO this may be a simple over-site on my part.
Here is my sip file with the configs I have set:
register => 050XXXXXXXX@ocn.ne.jp:XXXXXXX:XXXXXXX@voip-caXXXX.ocn.ne.jp/050XXXXXXXXX
When I do a sip show peers I see the connection.
jpykastXXXXX*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
voip-ca3XXX3.ocn.ne.jp/5XX5 211.123.XXX.1XX N 5XXX Unmonitored
But when I call one of my assigned numbers I do not see them come into the PBX.
You have presumably failed to register. Plese provide the SIP trace for the register attempt.
(I’m not sure that @ is allowed in a user field.)
Can you define the process of getting a “SIP trace for the register attempt”.
I goggled SIP trace and I see that “SIP Tool” and wireshark can be used to complete a sip trace.
Is a “SIP trace for the register attempt” just a packet capture? If so how can I force the registration attempt during my packet capture?
I think i got closer…
I did: sip debug ip
Nothing comes up on the console…
And I cannot find a full log under /var/log/asterisk/full
sip set debug peer name_of_peer
sip set debug on