Need help with inbound routes...getting "No DID or CID Match"

hi, i am setting up a new FreePBX 14.0.3.17 install. i have set up my sip trunk and it is registering, however, when i put my did on an extension and test, i get “all circuits busy” and the logs show

[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:1] NoOp(“PJSIP/1VOIP-00000026”, “No DID or CID Match”) in new stack

I am sure i am doing something wrong but i don’t see it. I have a packet capture and confirmed that the provider is sending 10 digits, that is what i have as the DID on the extension.

any help would be appreciated!

full log:

[2018-09-20 10:54:34] VERBOSE[14396] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.168.18’
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:1] NoOp(“PJSIP/1VOIP-00000026”, “No DID or CID Match”) in new stack
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:2] Answer(“PJSIP/1VOIP-00000026”, “”) in new stack
[2018-09-20 10:54:34] WARNING[30330][C-00000017] chan_sip.c: This function can only be used on SIP channels.
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:3] Log(“PJSIP/1VOIP-00000026”, "WARNING,Friendly Scanner from ") in new stack
[2018-09-20 10:54:34] WARNING[30330][C-00000017] Ext. s: Friendly Scanner from
[2018-09-20 10:54:34] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:4] Wait(“PJSIP/1VOIP-00000026”, “2”) in new stack
[2018-09-20 10:54:36] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:5] Playback(“PJSIP/1VOIP-00000026”, “ss-noservice”) in new stack
[2018-09-20 10:54:36] VERBOSE[30330][C-00000017] file.c: <PJSIP/1VOIP-00000026> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/1VOIP-00000026”, “hangupcall,”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/1VOIP-00000026”, “1?theend”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/1VOIP-00000026”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/1VOIP-00000026”, " monior file= ") in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:5] AGI(“PJSIP/1VOIP-00000026”, “attendedtransfer-rec-restart.php,”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] res_agi.c: <PJSIP/1VOIP-00000026>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Executing [[email protected]:6] Hangup(“PJSIP/1VOIP-00000026”, “”) in new stack
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/1VOIP-00000026’ in macro ‘hangupcall’
[2018-09-20 10:54:39] VERBOSE[30330][C-00000017] pbx.c: Spawn extension (from-pstn, h, 1) exited non-zero on ‘PJSIP/1VOIP-00000026’

The INVITE does not contain the DID, perhaps it’s in the To: header of the SIP INVITE. If so, your trunk needs to be set to the built in context, from-pstn-toheader

Lorne, that fixed it, thanks for your help!

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