I can't make any call

Asterisk Asterisk SIP
1

Hi everybody,

I install vicidial wich come with Asterisk 13.38.2-vici, I create my campain in predictive but when I make a call, the called phone dont ring. He does not get any call in his phone.

This is the log in asterisk when trying to make a call.

[quote][Nov 11 04:27:04] – Called 910747948770@default
[Nov 11 04:27:04] – Executing [910747948770@default:1] AGI(“Local/910747948770@default-0000004e;2”, “agi://127.0.0.1:4577/call_log”) in new stack
[Nov 11 04:27:04] – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=CAMP01))
[Nov 11 04:27:04] – <Local/910747948770@default-0000004e;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 11 04:27:04] – Executing [910747948770@default:2] Dial(“Local/910747948770@default-0000004e;2”, “SIP/10747948770@MTN,tTor”) in new stack
[Nov 11 04:27:04] == Using SIP RTP CoS mark 5
[Nov 11 04:27:04] – Called SIP/10747948770@MTN
[Nov 11 04:27:04] – Local/910747948770@default-0000004e;1 is ringing
[Nov 11 04:27:04] > 0x7fbd24082bd0 – Strict RTP learning after remote address set to: 10.50.51.112:10242
[Nov 11 04:27:04] – SIP/MTN-00000039 is making progress passing it to Local/910747948770@default-0000004e;2
[Nov 11 04:27:04] == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 11 04:27:04] – Called 917747948770@default
[Nov 11 04:27:04] – Executing [917747948770@default:1] AGI(“Local/917747948770@default-0000004f;2”, “agi://127.0.0.1:4577/call_log”) in new stack
[Nov 11 04:27:04] – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=CAMP01))
[Nov 11 04:27:04] – <Local/917747948770@default-0000004f;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 11 04:27:04] – Executing [917747948770@default:2] Dial(“Local/917747948770@default-0000004f;2”, “SIP/17747948770@MTN,tTor”) in new stack
[Nov 11 04:27:04] == Using SIP RTP CoS mark 5
[Nov 11 04:27:04] – Called SIP/17747948770@MTN
[Nov 11 04:27:04] – Local/917747948770@default-0000004f;1 is ringing
[Nov 11 04:27:04] > 0x7fbd2800a260 – Strict RTP learning after remote address set to: 10.50.51.112:10244
[Nov 11 04:27:04] – SIP/MTN-0000003a is making progress passing it to Local/917747948770@default-0000004f;2
[Nov 11 04:27:04] > 0x7fbd2800a260 – Strict RTP switching to RTP target address 10.50.51.112:10244 as source
[Nov 11 04:27:06] == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 11 04:27:06] == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 11 04:27:09] > 0x7fbd2800a260 – Strict RTP learning complete - Locking on source address 10.50.51.112:10244
[Nov 11 04:27:34] == Spawn extension (default, 910747948770, 2) exited non-zero on ‘Local/910747948770@default-0000004e;2’
[Nov 11 04:27:34] – Executing [h@default:1] AGI(“Local/910747948770@default-0000004e;2”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0-----CANCEL---------------SIP 183 Session Progress)”) in new stack
[Nov 11 04:27:34] == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 11 04:27:34] – <Local/910747948770@default-0000004e;2>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0-----CANCEL---------------SIP 183 Session Progress) completed, returning 0
[Nov 11 04:27:34] == Spawn extension (default, 917747948770, 2) exited non-zero on ‘Local/917747948770@default-0000004f;2’
[Nov 11 04:27:34] == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 11 04:27:34] – Executing [h@default:1] AGI(“Local/917747948770@default-0000004f;2”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0-----CANCEL---------------SIP 183 Session Progress)”) in new stack
[Nov 11 04:27:34] – <Local/917747948770@default-0000004f;2>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0-----CANCEL---------------SIP 183 Session Progress) completed, returning 0[/quote]

What’s wrong?

Can anyone help me? I am a beginner with asterisk.

Thanks and regards

2

You will find more relevant assistance on a Vicidial forum. This forum is a more ‘low level,’ ‘command line Asterisk’ group. Vicidial log messages mean close to nothing here.

Ultimately, a packet trace says what’s really going on.

3

Also the version of Asterisk quoted is a customised version of a version of Asterisk that went end of life over a week ago.

4

Make a SIP trace to track the calls between your Asterisk and MTN carrier, because calls looks fine, and the output on the console is normal for Asterisk and VICI

5

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