Voiptalk outbound sip trunk

Asterisk Asterisk Support
1

Hi There,

I’m new for freepbx and i don’t how it work and to troubleshoot.
I need your kind help to solve voiptalk outbound issue.

I cant make any outgoing call it said ‘the number you have dialed is not in service please check your number and try again’

Below is the logs:-

[2013-01-24 15:49:46] VERBOSE[3376] netsock2.c: == Using SIP RTP TOS bits 184
[2013-01-24 15:49:46] VERBOSE[3376] netsock2.c: == Using SIP RTP CoS mark 5
[2013-01-24 15:49:46] VERBOSE[6463] pbx.c: – Executing [+441355593400@from-trunk:1] Set(“SIP/710-000000be”, “__FROM_DID=+441355593400”) in new stack
[2013-01-24 15:49:46] VERBOSE[6463] pbx.c: – Executing [+441355593400@from-trunk:2] NoOp(“SIP/710-000000be”, “Received an unknown call with DID set to +441355593400”) in new stack
[2013-01-24 15:49:46] VERBOSE[6463] pbx.c: – Executing [+441355593400@from-trunk:3] Goto(“SIP/710-000000be”, “s,a2”) in new stack
[2013-01-24 15:49:46] VERBOSE[6463] pbx.c: – Goto (from-trunk,s,2)
[2013-01-24 15:49:46] VERBOSE[6463] pbx.c: – Executing [s@from-trunk:2] Answer(“SIP/710-000000be”, “”) in new stack
[2013-01-24 15:49:46] VERBOSE[6463] pbx.c: – Executing [s@from-trunk:3] Wait(“SIP/710-000000be”, “2”) in new stack
[2013-01-24 15:49:49] VERBOSE[6463] pbx.c: – Executing [s@from-trunk:4] Playback(“SIP/710-000000be”, “ss-noservice”) in new stack
[2013-01-24 15:49:49] VERBOSE[6463] file.c: – <SIP/710-000000be> Playing ‘ss-noservice.alaw’ (language ‘en’)
[2013-01-24 15:49:51] VERBOSE[6463] pbx.c: == Spawn extension (from-trunk, s, 4) exited non-zero on ‘SIP/710-000000be’
[2013-01-24 15:49:51] VERBOSE[6463] pbx.c: – Executing [h@from-trunk:1] Macro(“SIP/710-000000be”, “hangupcall,”) in new stack
[2013-01-24 15:49:51] VERBOSE[6463] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/710-000000be”, “1?theend”) in new stack
[2013-01-24 15:49:51] VERBOSE[6463] pbx.c: – Goto (macro-hangupcall,s,3)
[2013-01-24 15:49:51] VERBOSE[6463] pbx.c: – Executing [s@macro-hangupcall:3] ExecIf(“SIP/710-000000be”, “0?Set(CDR(recordingfile)=)”) in new stack
[2013-01-24 15:49:51] VERBOSE[6463] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/710-000000be”, “”) in new stack
[2013-01-24 15:49:51] VERBOSE[6463] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/710-000000be’ in macro ‘hangupcall’
[2013-01-24 15:49:51] VERBOSE[6463] pbx.c: == Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/710-000000be’

Please do let me know if you requires further information, your kind help on this matter is much appreciated.

2

Freepbx questions should be directed to the forums on the freepbx.org web site.

My guess is that FreePBX doesn’t expect DID numbers to start with +, or you failed to include the + when you configured it.

3

Sorry for posting in your forum, i dont know where i can get help…
Where i need to add +?

4

Wherever you specify the incoming digits that you expect to match. I don’t know the structure of the dialplan supplied by FreePBX, nor the details of the user interface used to configure it.

However, if you have a true direct in dialing line (some or all of the called number is forwarded to you on incoming calls) you need to include rules in the dialplan that match the exact form in which those numbers are forwarded, in the context invoked for calls from that peer.