[asterisk-users] Dial Plan Issue

2015-02-10 Thread Haley,Scott A
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 
[XX@subMachine:4] Playback(SIP/trunk503out-9728, temp/0250002) 
in new stack
[2015-02-10 12:13:29] VERBOSE[10502] pbx.c: -- Executing [XX 
@subMachine:5] Wait(SIP/trunk503out-9728, 1) in new stack
[2015-02-10 12:13:30] VERBOSE[10502] pbx.c: -- Executing [XX 
@subMachine:6] Playback(SIP/trunk503out-9728, temp/0250002) in new stack

Standard Asterisk Build:
[2015-02-10 15:01:12] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx @subMachine:1] SendDTMF(SIP/SMtrunk1-000f, w1w) in new 
stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx @subMachine:2] Set(SIP/SMtrunk1-000f, IVR_MSG=temp/0250002) 
in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx@subMachine:3] System(SIP/SMtrunk1-000f, /bin/echo -e 
xx,RUN2,i9,02102015145822,MACHINE,SIP/SMtrunk1-000f,02.10.2015 
15.01log/outbound.txt) in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx @subMachine:4] Playback(SIP/SMtrunk1-000f, temp/0250002) in 
new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] file.c: -- 
SIP/SMtrunk1-000f Playing 'temp/0250002.slin' (language 'en')
[2015-02-10 15:01:50] VERBOSE[32567][C-000f] pbx.c:   == Spawn extension 
(subMachine, xx, 4) exited non-zero on 'SIP/SMtrunk1-000f'

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.

Thanks,
Scott



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Re: [asterisk-users] Dial Plan Issue

2015-02-10 Thread Haley,Scott A
One follow-up. At the end of the call, after it dis-connects I get the 
following error:

[2015-02-10 15:33:42] NOTICE[4524]: pbx_spool.c:402 attempt_thread: Call 
completed to SIP/SMtrunk1/xx

Thanks,
Scott Haley
5-2244

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Tuesday, February 10, 2015 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dial Plan Issue

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 
[XX@subMachine:4] Playback(SIP/trunk503out-9728, temp/0250002) 
in new stack
[2015-02-10 12:13:29] VERBOSE[10502] pbx.c: -- Executing [XX 
@subMachine:5] Wait(SIP/trunk503out-9728, 1) in new stack
[2015-02-10 12:13:30] VERBOSE[10502] pbx.c: -- Executing [XX 
@subMachine:6] Playback(SIP/trunk503out-9728, temp/0250002) in new stack

Standard Asterisk Build:
[2015-02-10 15:01:12] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx @subMachine:1] SendDTMF(SIP/SMtrunk1-000f, w1w) in new 
stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx @subMachine:2] Set(SIP/SMtrunk1-000f, IVR_MSG=temp/0250002) 
in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx@subMachine:3] System(SIP/SMtrunk1-000f, /bin/echo -e 
xx,RUN2,i9,02102015145822,MACHINE,SIP/SMtrunk1-000f,02.10.2015 
15.01log/outbound.txt) in new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing 
[xx @subMachine:4] Playback(SIP/SMtrunk1-000f, temp/0250002) in 
new stack
[2015-02-10 15:01:16] VERBOSE[32567][C-000f] file.c: -- 
SIP/SMtrunk1-000f Playing 'temp/0250002.slin' (language 'en')
[2015-02-10 15:01:50] VERBOSE[32567][C-000f] pbx.c:   == Spawn extension 
(subMachine, xx, 4) exited non-zero on 'SIP/SMtrunk1-000f'

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.

Thanks,
Scott



If you are not the intended recipient of this message (including attachments), 
or if you have received this message in error, immediately notify us and delete 
it and any attachments.

If you do not wish to receive any email messages from us, excluding 
administrative communications, please email this request to 
messa...@edwardjones.commailto:messa...@edwardjones.com along with the email 
address you wish to unsubscribe.

For important additional information related to this email, visit 
www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure.
 Edward D. Jones  Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. 
Louis, MO 63131 © Edward Jones. All rights reserved.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users