I am trying to get my DID to work in Asterisk Version 13.12.1 this is what my SIP provider said was happening.
My Invite message on my test call from my cell phone:
Ask your provider if they are using IP authentication or registration. If the former, you can lose the username and password on the inbound side. Note that you may need to set up different incoming and outgoing (peer and user) sections in your trunk if that’s the case.
I just got an email from my provider. They stated that they can see calls going to and coming from the server now. It is IP Authentication. I can now see the calls in the asterisk -rvvv log but I now get a busy signal and it shows this.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [4302@from-trunk:1] Set(“SIP/ctctelcom.net1-0000001d”, “__FROM_DID=4302”) in new stack
– Executing [4302@from-trunk:2] NoOp(“SIP/ctctelcom.net1-0000001d”, “Received an unknown call with DID set to 4302”) in new stack
– Executing [4302@from-trunk:3] Goto(“SIP/ctctelcom.net1-0000001d”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/ctctelcom.net1-0000001d”, “”) in new stack
– Executing [s@from-trunk:3] Log(“SIP/ctctelcom.net1-0000001d”, “WARNING,Friendly Scanner from 209.212.xxx.xxx”) in new stack
[2017-12-04 13:23:20] WARNING[6054][C-0000002f]: Ext. s:3 @ from-trunk: Friendly Scanner from 209.212.xxx.xxx
– Executing [s@from-trunk:4] Wait(“SIP/ctctelcom.net1-0000001d”, “2”) in new stack
– Executing [s@from-trunk:5] Playback(“SIP/ctctelcom.net1-0000001d”, “ss-noservice”) in new stack
– <SIP/ctctelcom.net1-0000001d> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-trunk:6] SayAlpha(“SIP/ctctelcom.net1-0000001d”, “4302”) in new stack
– <SIP/ctctelcom.net1-0000001d> Playing ‘digits/4.ulaw’ (language ‘en’)
– <SIP/ctctelcom.net1-0000001d> Playing ‘digits/3.ulaw’ (language ‘en’)
– <SIP/ctctelcom.net1-0000001d> Playing ‘digits/0.ulaw’ (language ‘en’)
– <SIP/ctctelcom.net1-0000001d> Playing ‘digits/2.ulaw’ (language ‘en’)
== Spawn extension (from-trunk, s, 6) exited non-zero on ‘SIP/ctctelcom.net1-0000001d’
– Executing [h@from-trunk:1] Macro(“SIP/ctctelcom.net1-0000001d”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/ctctelcom.net1-0000001d”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/ctctelcom.net1-0000001d”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/ctctelcom.net1-0000001d”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/ctctelcom.net1-0000001d’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/ctctelcom.net1-0000001d’
Ok, I figured it out. The Asterisk server can use what looks like any DID inbound format. My SIP provider sends just the DID4 format so All I needed to do was fix the inbound route to have the last four #'s of the DID and it worked. Thanks for all the reply and help.