WARNING[15823][C-0000000f]: channel.c:4862 ast_prod: Prodding channel 'SIP/1001-0000000f' failed

Hi Everyone,

I am glad to have reached this stage. This was my first installation and it has been great so far. I am trying to setup trunks and outbound routes and extensions on this new server. I have tried asterisk -vvvr to find out what could be the issue the outbound calls are not made. Please find the error below when tried to make a call.

Asterisk 11.15.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.15.0 currently running on Elastix (pid = 14368)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] ResetCDR(“SIP/1001-0000000f”, “”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/1001-0000000f”, “”) in new stack
– Executing [[email protected]:3] Progress(“SIP/1001-0000000f”, “”) in new stack
– Executing [[email protected]:4] Wait(“SIP/1001-0000000f”, “1”) in new stack
– Executing [[email protected]:5] Progress(“SIP/1001-0000000f”, “”) in new stack
– Executing [[email protected]:6] Playback(“SIP/1001-0000000f”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/1001-0000000f> Playing ‘silence/1.gsm’ (language ‘en’)
– <SIP/1001-0000000f> Playing ‘cannot-complete-as-dialed.gsm’ (language ‘en’)
– <SIP/1001-0000000f> Playing ‘check-number-dial-again.gsm’ (language ‘en’)
– Executing [[email protected]:7] Wait(“SIP/1001-0000000f”, “1”) in new stack
– Executing [[email protected]:8] Congestion(“SIP/1001-0000000f”, “20”) in new stack
[2015-02-05 21:18:57] WARNING[15823][C-0000000f]: channel.c:4862 ast_prod: Prodding channel ‘SIP/1001-0000000f’ failed
== Spawn extension (from-internal, 00XXXXXXX22240, 8) exited non-zero on ‘SIP/1001-0000000f’
– Executing [[email protected]:1] Hangup(“SIP/1001-0000000f”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1001-0000000f’

You need a working outbound route for 00XXXXXXX22240

– Executing [[email protected]:6] Playback(“SIP/1001-0000000f”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack

That being said, If you are starting out, I would recommend you use a supported Distro, Elasix is not such a beast. Select an appropriate one from:-

I would suggest you stick with Asterisk 11 for now.

Then spend some time at:-

http://wiki.freepbx.org/

To get you up to speed.