Incoming calls not reaching extension?

I have having some issues with a cloud hosted PBX. I have tried placing the inbound route to a “put on hold forever” just to verify calls are reaching the extension. However, this is what appears on the log when a call is placed to my inbound route:

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [[email protected]:1] NoOp("SIP/VOIP-00000006", "No DID or CID Match") in new stack
    -- Executing [[email protected]:2] Answer("SIP/VOIP-00000006", "") in new stack
       > 0x7f4560007370 -- Probation passed - setting RTP source address to
    -- Executing [[email protected]:3] Log("SIP/VOIP-00000006", "WARNING,Friendly Scanner from") in new stack
[2017-01-31 16:05:43] WARNING[5206][C-00000006]: Ext. s:3 @ from-trunk: Friendly Scanner from
    -- Executing [[email protected]:4] Wait("SIP/VOIP-00000006", "2") in new stack
    -- Executing [[email protected]:5] Playback("SIP/VOIP-00000006", "ss-noservice") in new stack
    -- <SIP/VOIP-00000006> Playing 'ss-noservice.ulaw' (language 'en')
    -- Executing [[email protected]:6] SayAlpha("SIP/VOIP-00000006", "") in new stack
    -- Executing [[email protected]:7] Hangup("SIP/VOIP-00000006", "") in new stack
  == Spawn extension (from-trunk, s, 7) exited non-zero on 'SIP/VOIP-00000006'
    -- Executing [[email protected]:1] Macro("SIP/VOIP-00000006", "hangupcall,") in new stack
    -- Executing [[email protected]:1] ExecIf("SIP/VOIP-00000006", "0?Set(CDR(recordingfile)=.)") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/VOIP-00000006", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [[email protected]:4] ExecIf("SIP/VOIP-00000006", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [[email protected]:5] Hangup("SIP/VOIP-00000006", "") in new stack
  == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/VOIP-00000006' in macro 'hangupcall'
  == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/VOIP-00000006'

Any idea what can be going wrong? I’ve double checked trunk settings, user/pass, and settings with my provider. Everything seems to be correct and I can’t pin point why this may be happening.

what does your inbound route look like?

You aren’t hitting your inbound route.

Like this

remove the did from the route so that you have no cid or did - this will allow any inbound call to use this route

Now its working. What can be the issue? On my other company PBX I have multiple inbound routes all with the DIDs listed in the number field.

Thank you so much for the help! :slight_smile:

my guess is that what you were receiving from your ITSP is not what you had in the route. check with your provider as to the format they are sending

There should be a line in the logs near the “No DID or CID match” line in your /var/log/asterisk/full that tells you what your ITSP is sending you for your DID.

Create an incoming Route that matches that and you’re in business. You can also ask your ITSP what they are sending and ask them to modify it so that it matches what you are expecting.

I realized what the problem was. For some reason my provider configured my setup to be compatible with soft-phones instead of a PBX server. The issue has been fixed and now everything is working properly.

Thank you again for the help! Thanks to Dave as well!