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
-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
I have a situation that I need help with. I have 2 phone systems, 1 Asterisk
and 1 Avaya. All voicemail is kept on the Avaya system. Whenever a call comes
into an extension that the Asterisk server owns, I re-direct it to a different
number that is owned by the Avaya System. If that Avaya
I am trying to compile the certified-asterisk-11.6-cert5 code and when I try to
start it and then go into the console I am getting the error message asterisk
dead but subsys locked. Can anyone help with why this is happening? I have
never seen this before.
This is a fresh install on a new
Had to re-install and change selinux to disable. Works now.
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: Friday, September 12, 2014 1:00 PM
To: Asterisk Users Mailing List - Non
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Thursday, July 17, 2014 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
Thanks AJ, this sounds like what I need.
Thanks,
Scott Haley
] On Behalf Of A J Stiles
Sent: Friday, July 18, 2014 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
On Friday 18 Jul 2014, Haley,Scott A wrote:
I have this working but I have one problem. I need to grab values from
variables that I have
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Simultaneous Ring
On Wednesday 16 Jul 2014, Haley,Scott A wrote:
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial another set of
numbers while continuing to ring, or interrupting the first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is
I am trying to run an agi script and asterisk is not finding it. The output of
the cli is as follows:
-- Executing [s@tbs-utils:7] AGI(SIP/7002-001a, tbsdial.agi) in new
stack
[Apr 28 12:00:05] WARNING[21812][C-000f]: res_agi.c:1681 launch_script:
Failed to execute
of Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
I
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
It runs but hangs with the output of:
perl tbsdial.agi 81101
GET
One more thing. I have this exact same script working on an Asterisk 1.8 box.
This is a new Asterisk 11.7 box.
Thanks,
Scott Haley
5-2244
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent
: Monday, April 28, 2014 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue
if that is the case then check again Perl Asterisk AGI.
On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A
scott.ha...@edwardjones.commailto:scott.ha
:58 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Trunk issue
On 28-04-14 19:49, Haley,Scott A wrote:
Now I am getting Permission denied.
Have you checked if SELinux is blocking the app? Any blockage should show up as
an 'AVC' in /var/log/audit/audit.log You can temporarily
a trunk between
Asterisk and CM this morning, and it works great providing that you allow
for anonymous calls.
-Original Message-
From: Haley,Scott A scott.ha...@edwardjones.com
Sent: Wednesday, April 23, 2014 9:36am
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
I have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I
try to send a call over it, the call gets rejected. Here is the sip debug
trace. Could anyone tell me what may be going wrong?
nxdasterisk-2*CLI
[Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable
:172.17.184.46:31285 ---
INVITE sip:51...@edj.devjones.com SIP/2.0
From: Haley, Scott
sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00
To: sip:51...@edj.devjones.com
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Max-Forwards: 71
Via: SIP/2.0/TCP 172.17.184.46;branch
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