[SOLVED] Unitel Inbound Route Setup Issue

Hello everyone,

I am yet again having trouble with FreePBX.

We just switched to Unitel using a PJSIP trunk with them. Outbound routes are working just fine. However inbound keeps giving the “not in service” message.

Their instructions for inbound routing are:

  • Navigate to Connectivity > Inbound Routes
  • Click button [+ Add Incoming Route]
  • Description: {Descriptive Name} i.e. 15556661234
  • DID Number: 15556661234 **( must use leading “1”
  • Caller ID Number: 15556661234
  • Set Destination: {Set to your desired destination. i.e. Ring Groups %set to appropriate destination%

I have tried almost every variation including +1 1 etc…

It seems to be a Caller ID Number issue, attached are the logs.

[2019-07-23 20:36:11] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:1] NoOp(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “No DID or CID Match”) in new stack
[2019-07-23 20:36:11] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:2] Answer(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “”) in new stack
[2019-07-23 20:36:12] WARNING[24915][C-0001322a] chan_sip.c: This function can only be used on SIP channels.
[2019-07-23 20:36:12] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:3] Log(“PJSIP/UniTel_SIP_Trunk_001-00000572”, "WARNING,Friendly Scanner from ") in new stack
[2019-07-23 20:36:12] WARNING[24915][C-0001322a] Ext. s: Friendly Scanner from
[2019-07-23 20:36:12] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:4] Wait(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “2”) in new stack

[2019-07-23 20:36:14] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:5] Playback(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “ss-noservice”) in new stack
[2019-07-23 20:36:14] VERBOSE[24915][C-0001322a] file.c: <PJSIP/UniTel_SIP_Trunk_001-00000572> Playing ‘ss-noservice.ulaw’ (language ‘en’)

[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:6] SayAlpha(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “”) in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:7] Hangup(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “”) in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Spawn extension (from-pstn, s, 7) exited non-zero on ‘PJSIP/UniTel_SIP_Trunk_001-00000572’
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “hangupcall,”) in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “1?theend”) in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/UniTel_SIP_Trunk_001-00000572”, " montior file= ") in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:5] GotoIf(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “1?skipagi”) in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Executing [[email protected]:7] Hangup(“PJSIP/UniTel_SIP_Trunk_001-00000572”, “”) in new stack
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/UniTel_SIP_Trunk_001-00000572’ in macro ‘hangupcall’
[2019-07-23 20:36:19] VERBOSE[24915][C-0001322a] pbx.c: Spawn extension (from-pstn, h, 1) exited non-zero on ‘PJSIP/UniTel_SIP_Trunk_001-00000572’

Though I know nothing about UniTel, it’s clear that they are not sending the DID in the SIP URI.

Try changing the Context for the trunk to
from-pstn-toheader
then look at what is shown for DID in your CDRs and set your Inbound Route to match that format.

If you have only one DID on the trunk, an easier fix is to set From User for the trunk to whatever number you have for the Inbound Route DID.

1 Like

Alright! Looks like it was because PJSIP wasn’t playing nicely! CHAN_SIP is working perfectly after the modification! Yay!

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