[asterisk-users] Dial Plan Issue

Haley,Scott A scott.haley at edwardjones.com
Tue Feb 10 15:14:33 CST 2015

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it.

The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail  playback and it does not leave the voicemail. Here is the printout of the log file for both boxes.

Free PBX:
[2015-02-10 12:12:34] VERBOSE[10502] pbx.c:     -- Executing [XXXXXXXXXX at subMachine:4] Playback("SIP/trunk503out-00009728", "temp/0250002") in new stack
[2015-02-10 12:13:29] VERBOSE[10502] pbx.c:     -- Executing [XXXXXXXXXX @subMachine:5] Wait("SIP/trunk503out-00009728", "1") in new stack
[2015-02-10 12:13:30] VERBOSE[10502] pbx.c:     -- Executing [XXXXXXXXXX @subMachine:6] Playback("SIP/trunk503out-00009728", "temp/0250002") in new stack

Standard Asterisk Build:
[2015-02-10 15:01:12] VERBOSE[32567][C-0000000f] pbx.c:     -- Executing [xxxxxxxxxx @subMachine:1] SendDTMF("SIP/SMtrunk1-0000000f", "w1wwwww") in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-0000000f] pbx.c:     -- Executing [xxxxxxxxxx @subMachine:2] Set("SIP/SMtrunk1-0000000f", "IVR_MSG=temp/0250002") in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-0000000f] pbx.c:     -- Executing [xxxxxxxxxx at subMachine:3] System("SIP/SMtrunk1-0000000f", "/bin/echo -e "xxxxxxxxxx,RUN2,i9,02102015145822,MACHINE,SIP/SMtrunk1-0000000f,02.10.2015 15.01">>log/outbound.txt") in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-0000000f] pbx.c:     -- Executing [xxxxxxxxxx @subMachine:4] Playback("SIP/SMtrunk1-0000000f", "temp/0250002") in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-0000000f] file.c:     -- <SIP/SMtrunk1-0000000f> Playing 'temp/0250002.slin' (language 'en')
[2015-02-10 15:01:50] VERBOSE[32567][C-0000000f] pbx.c:   == Spawn extension (subMachine, xxxxxxxxxx, 4) exited non-zero on 'SIP/SMtrunk1-0000000f'

I copied the context from the FreePbx box over to the new box so the code should be the same. Any help would be appreciated.


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