Re: [asterisk-users] Trunk issue

2014-04-28 Thread Haley,Scott A 
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 '/var/lib/asterisk/agi-bin/tbsdial.agi': File does not exist.

The file is in that directory and is owned by the user asterisk. Why does it 
say the file does not exist?

Thanks,
Scott Haley
5-2244





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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Eric Wieling 
Does the script generate an error when run outside of Asterisk?   An AGI should 
simply wait for input when run outside 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 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 '/var/lib/asterisk/agi-bin/tbsdial.agi': File 
does not exist.

The file is in that directory and is owned by the user asterisk. Why does it 
say the file does not exist?
-- 
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Trunk issue

2014-04-28 Thread Haley,Scott A 
It runs but hangs with the output of:
perl tbsdial.agi 81101
GET VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = DIALGROUP1;
my $dialgroup2 = DIALGROUP2;
my $vmvariable = VM;
my $timer = TIMER;
my $branch = BRANCH;
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi-get_variable(astexten);

#$agi-answer();
#$agi-stream_file(welcome);






$agi-set_variable($dialgroup1, $dg1value);
$agi-set_variable($dialgroup2, $dg2value);
$agi-set_variable($vmvariable, $vmvalue);
$agi-set_variable($timer, $timervalue);
$agi-set_variable($branch, $branchvalue);

Thanks,
Scott Haley
5-2244





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rights reserved.




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk?   An AGI should 
simply wait for input when run outside 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 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 '/var/lib/asterisk/agi-bin/tbsdial.agi': File 
does not exist.

The file is in that directory and is owned by the user asterisk. Why does it 
say the file does not exist?
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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Asghar Mohammad 
file is executable?
can you show ls -l /var/lib/asterisk/agi-bin


On Mon, Apr 28, 2014 at 7:12 PM, Haley,Scott A
scott.ha...@edwardjones.comwrote:

 It runs but hangs with the output of:
 perl tbsdial.agi 81101
 GET VARIABLE astexten


 Right now, it is a simple perl script. Here is the entire script.

 #!/usr/bin/perl


 use Asterisk::AGI;

 my $agi = new Asterisk::AGI;

 my $dialgroup1 = DIALGROUP1;
 my $dialgroup2 = DIALGROUP2;
 my $vmvariable = VM;
 my $timer = TIMER;
 my $branch = BRANCH;
 my $input;
 my $dg1value;
 my $dg2value;
 my $vmvalue;
 my $branchvalue;



 $input = $agi-get_variable(astexten);

 #$agi-answer();
 #$agi-stream_file(welcome);






 $agi-set_variable($dialgroup1, $dg1value);
 $agi-set_variable($dialgroup2, $dg2value);
 $agi-set_variable($vmvariable, $vmvalue);
 $agi-set_variable($timer, $timervalue);
 $agi-set_variable($branch, $branchvalue);

 Thanks,
 Scott Haley
 5-2244





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 attachments), or if you have received this message in error, immediately
 notify us and delete it and any attachments.

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 administrative communications, please email this request to
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 For important additional information related to this email, visit
 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.




 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Monday, April 28, 2014 12:07 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 Does the script generate an error when run outside of Asterisk?   An AGI
 should simply wait for input when run outside 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 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 '/var/lib/asterisk/agi-bin/tbsdial.agi':
 File does not exist.

 The file is in that directory and is owned by the user asterisk. Why
 does it say the file does not exist?
 --
 _
 -- 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
 --
 _
 -- 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

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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Eric Wieling 

Odd.  AGI scripts should hang waiting for input when run from the CLI.  They 
should not output anything.  If the script is not set as executable you'd get 
an error.

If you were not running it as the same user as asterisk runs as you should 
still get a different error.


-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 1:13 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 VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = DIALGROUP1;
my $dialgroup2 = DIALGROUP2;
my $vmvariable = VM;
my $timer = TIMER;
my $branch = BRANCH;
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi-get_variable(astexten);

#$agi-answer();
#$agi-stream_file(welcome);






$agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, 
$dg2value); $agi-set_variable($vmvariable, $vmvalue); 
$agi-set_variable($timer, $timervalue); $agi-set_variable($branch, 
$branchvalue);

Thanks,
Scott Haley
5-2244





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.com along with the email address you wish to unsubscribe.

For important additional information related to this email, visit 
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.




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk?   An AGI should 
simply wait for input when run outside 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 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 '/var/lib/asterisk/agi-bin/tbsdial.agi': File 
does not exist.

The file is in that directory and is owned by the user asterisk. Why does it 
say the file does not exist?
--
_
-- 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
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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Haley,Scott A 
Here is the directory listing:

[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x.  2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x.  1 asterisk asterisk  590 Apr 28 11:55 tbsdial.agi

Thanks,
Scott Haley
5-2244


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue


Odd.  AGI scripts should hang waiting for input when run from the CLI.  They 
should not output anything.  If the script is not set as executable you'd get 
an error.

If you were not running it as the same user as asterisk runs as you should 
still get a different error.


-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 1:13 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 VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = DIALGROUP1;
my $dialgroup2 = DIALGROUP2;
my $vmvariable = VM;
my $timer = TIMER;
my $branch = BRANCH;
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi-get_variable(astexten);

#$agi-answer();
#$agi-stream_file(welcome);






$agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, 
$dg2value); $agi-set_variable($vmvariable, $vmvalue); 
$agi-set_variable($timer, $timervalue); $agi-set_variable($branch, 
$branchvalue);

Thanks,
Scott Haley
5-2244





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.com along with the email address you wish to unsubscribe.

For important additional information related to this email, visit 
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.




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk?   An AGI should 
simply wait for input when run outside 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 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 '/var/lib/asterisk/agi-bin/tbsdial.agi': File 
does not exist.

The file is in that directory and is owned by the user asterisk. Why does it 
say the file does not exist?
--
_
-- 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
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New

Re: [asterisk-users] Trunk issue

2014-04-28 Thread Haley,Scott A 
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 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Here is the directory listing:

[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x.  2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x.  1 asterisk asterisk  590 Apr 28 11:55 tbsdial.agi

Thanks,
Scott Haley
5-2244


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue


Odd.  AGI scripts should hang waiting for input when run from the CLI.  They 
should not output anything.  If the script is not set as executable you'd get 
an error.

If you were not running it as the same user as asterisk runs as you should 
still get a different error.


-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 1:13 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 VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = DIALGROUP1;
my $dialgroup2 = DIALGROUP2;
my $vmvariable = VM;
my $timer = TIMER;
my $branch = BRANCH;
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi-get_variable(astexten);

#$agi-answer();
#$agi-stream_file(welcome);






$agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, 
$dg2value); $agi-set_variable($vmvariable, $vmvalue); 
$agi-set_variable($timer, $timervalue); $agi-set_variable($branch, 
$branchvalue);

Thanks,
Scott Haley
5-2244





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.com along with the email address you wish to unsubscribe.

For important additional information related to this email, visit 
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.




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Does the script generate an error when run outside of Asterisk?   An AGI should 
simply wait for input when run outside 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 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 '/var/lib/asterisk/agi-bin/tbsdial.agi': File 
does not exist.

The file is in that directory and is owned by the user asterisk. Why does it 
say the file does not exist?
--
_
-- 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
--
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   http://www.asterisk.org/hello

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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Asghar Mohammad 
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.comwrote:

 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 12:32 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 Here is the directory listing:

 [root@nxdasterisk-3 agi-bin]# ls -al
 total 12
 drwxr-xr-x.  2 asterisk asterisk 4096 Apr 28 12:11 .
 drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
 -rwxrwxr-x.  1 asterisk asterisk  590 Apr 28 11:55 tbsdial.agi

 Thanks,
 Scott Haley
 5-2244


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Monday, April 28, 2014 12:25 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue


 Odd.  AGI scripts should hang waiting for input when run from the CLI.
  They should not output anything.  If the script is not set as executable
 you'd get an error.

 If you were not running it as the same user as asterisk runs as you should
 still get a different error.


 -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 1:13 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 VARIABLE astexten


 Right now, it is a simple perl script. Here is the entire script.

 #!/usr/bin/perl


 use Asterisk::AGI;

 my $agi = new Asterisk::AGI;

 my $dialgroup1 = DIALGROUP1;
 my $dialgroup2 = DIALGROUP2;
 my $vmvariable = VM;
 my $timer = TIMER;
 my $branch = BRANCH;
 my $input;
 my $dg1value;
 my $dg2value;
 my $vmvalue;
 my $branchvalue;



 $input = $agi-get_variable(astexten);

 #$agi-answer();
 #$agi-stream_file(welcome);






 $agi-set_variable($dialgroup1, $dg1value);
 $agi-set_variable($dialgroup2, $dg2value);
 $agi-set_variable($vmvariable, $vmvalue); $agi-set_variable($timer,
 $timervalue); $agi-set_variable($branch, $branchvalue);

 Thanks,
 Scott Haley
 5-2244





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 messa...@edwardjones.com along with the email address you wish to
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 d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward
 Jones. All rights reserved.




 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Monday, April 28, 2014 12:07 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Trunk issue

 Does the script generate an error when run outside of Asterisk?   An AGI
 should simply wait for input when run outside 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 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 '/var/lib/asterisk/agi-bin/tbsdial.agi':
 File does not exist.

 The file is in that directory and is owned by the user asterisk. Why
 does it say the file does not exist?
 --
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 asterisk-users mailing list
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 asterisk

Re: [asterisk-users] Trunk issue

2014-04-28 Thread Haley,Scott A 
Now I am getting Permission denied.

-- Executing [4000@phones:1] NoOp(SIP/7001-003a, Starting TBS Dailer 
App) in new stack
-- Executing [4000@phones:2] NoOp(SIP/7001-003a, 4000) in new stack
-- Executing [4000@phones:3] Gosub(SIP/7001-003a, 
tbs-utils,s,1,(4000)) in new stack
-- Executing [s@tbs-utils:1] NoOp(SIP/7001-003a, Entering tbs-utils 
for 4000) in new stack
-- Executing [s@tbs-utils:2] Set(SIP/7001-003a, DIALGROUP1=) in new 
stack
-- Executing [s@tbs-utils:3] Set(SIP/7001-003a, DIALGROUP2=) in new 
stack
-- Executing [s@tbs-utils:4] Set(SIP/7001-003a, VM=) in new stack
-- Executing [s@tbs-utils:5] Set(SIP/7001-003a, TIMER=) in new stack
-- Executing [s@tbs-utils:6] Set(SIP/7001-003a, BRANCH=) in new 
stack
-- Executing [s@tbs-utils:7] AGI(SIP/7001-003a, tbsdial.agi) in new 
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/tbsdial.agi
tbsdial.agi: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': 
Permission denied

Thanks,
Scott Haley
5-2244

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad
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...@edwardjones.com wrote:
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.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Haley,Scott A
Sent: Monday, April 28, 2014 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue

Here is the directory listing:

[root@nxdasterisk-3 agi-bin]# ls -al
total 12
drwxr-xr-x.  2 asterisk asterisk 4096 Apr 28 12:11 .
drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 ..
-rwxrwxr-x.  1 asterisk asterisk  590 Apr 28 11:55 tbsdial.agi

Thanks,
Scott Haley
5-2244


-Original Message-
From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Eric Wieling
Sent: Monday, April 28, 2014 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trunk issue


Odd.  AGI scripts should hang waiting for input when run from the CLI.  They 
should not output anything.  If the script is not set as executable you'd get 
an error.

If you were not running it as the same user as asterisk runs as you should 
still get a different error.


-Original Message-
From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.commailto: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 VARIABLE astexten


Right now, it is a simple perl script. Here is the entire script.

#!/usr/bin/perl


use Asterisk::AGI;

my $agi = new Asterisk::AGI;

my $dialgroup1 = DIALGROUP1;
my $dialgroup2 = DIALGROUP2;
my $vmvariable = VM;
my $timer = TIMER;
my $branch = BRANCH;
my $input;
my $dg1value;
my $dg2value;
my $vmvalue;
my $branchvalue;



$input = $agi-get_variable(astexten);

#$agi-answer();
#$agi-stream_file(welcome);






$agi-set_variable($dialgroup1, $dg1value); $agi-set_variable($dialgroup2, 
$dg2value); $agi-set_variable($vmvariable, $vmvalue); 
$agi-set_variable($timer, $timervalue); $agi-set_variable($branch, 
$branchvalue);

Thanks,
Scott Haley
5-2244





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or if you have received this message in error, immediately notify us and delete 
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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.




-Original Message-
From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun

Re: [asterisk-users] Trunk issue

2014-04-28 Thread Patrick Laimbock 

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 set 
SELinux to permissive with 'setenforce 0' and check if the problem goes 
away.


HTH,
Patrick

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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Haley,Scott A 
That seemed to fix it. Thanks to everyone.

Thanks,
Scott Haley
5-2244





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or if you have received this message in error, immediately notify us and delete 
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Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All 
rights reserved.




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Monday, April 28, 2014 12:58 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 set SELinux to 
permissive with 'setenforce 0' and check if the problem goes away.

HTH,
Patrick

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Re: [asterisk-users] Trunk issue

2014-04-28 Thread Patrick Laimbock 

On 28-04-14 20:13, Haley,Scott A wrote:

That seemed to fix it. Thanks to everyone.


https://bugzilla.redhat.com/show_bug.cgi?id=1092150

HTH,
Patrick

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Re: [asterisk-users] Trunk issue

2014-04-24 Thread Haley,Scott A 
It is just plain Asterisk. I solved the original problem of it not being in the 
from-pstn context, now I am getting a rejected error I believe from the CM.

Thanks,
Scott Haley
5-2244

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
richard.seg...@marisec.ca
Sent: Wednesday, April 23, 2014 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Trunk issue

Are you using freeswitch, or just plain asterisk?  I just setup 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
Subject: [asterisk-users] Trunk issue

--
_
-- 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-usersI 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 to set 
utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio 
is at 18380 Adding codec 14 (alaw) to SDP Adding codec 100012 (g722) to SDP 
Adding codec 13 (ulaw) to SDP Reliably Transmitting (no NAT) to 
192.168.175.135:5060:
INVITE sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Contact: sip:3145152000@192.168.122.57:5060
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 23 Apr 2014 13:20:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 
192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

--- SIP read from UDP:192.168.175.135:5060 ---
SIP/2.0 100 Trying
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

-
--- (7 headers 0 lines) ---

--- SIP read from UDP:192.168.175.135:5060 --- INVITE 
sip:913145152...@devjones.com SIP/2.0
P-AV-Message-Id: 1_1
Route: sip:192.168.122.57;lr;phase=terminating
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Date: Wed, 23 Apr 2014 13:20:59 GMT
Contact: 
sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68
Via: SIP/2.0/UDP 
192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 
192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 
192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 
192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr
Record-Route: 
sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr
Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr
P-Charging-Vector: icid-value=d13ae820-caef-11e3-9b9c-6c3be5a59e68
User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: Edward Jones sip:3145152...@devjones.com
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Max-Forwards: 66
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 229
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
P-Location: 
SM;origlocname=Asterisk-2;origsiglocname=Asterisk-2;origmedialocname=Asterisk-2;termlocname=Asterisk-2;termsiglocname=Asterisk-2;smaccounting=true

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 
192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000

[asterisk-users] Trunk issue

2014-04-23 Thread Haley,Scott A 
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 to set 
utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted
Audio is at 18380
Adding codec 14 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 13 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.175.135:5060:
INVITE sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Contact: sip:3145152000@192.168.122.57:5060
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 23 Apr 2014 13:20:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

--- SIP read from UDP:192.168.175.135:5060 ---
SIP/2.0 100 Trying
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

-
--- (7 headers 0 lines) ---

--- SIP read from UDP:192.168.175.135:5060 ---
INVITE sip:913145152...@devjones.com SIP/2.0
P-AV-Message-Id: 1_1
Route: sip:192.168.122.57;lr;phase=terminating
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Date: Wed, 23 Apr 2014 13:20:59 GMT
Contact: 
sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68
Via: SIP/2.0/UDP 
192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 
192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 
192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 
192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr
Record-Route: 
sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr
Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr
P-Charging-Vector: icid-value=d13ae820-caef-11e3-9b9c-6c3be5a59e68
User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: Edward Jones sip:3145152...@devjones.com
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Max-Forwards: 66
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 229
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
P-Location: 
SM;origlocname=Asterisk-2;origsiglocname=Asterisk-2;origmedialocname=Asterisk-2;termlocname=Asterisk-2;termsiglocname=Asterisk-2;smaccounting=true

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
-
--- (27 headers 11 lines) ---
Sending to 192.168.175.135:5060 (no NAT)
Sending to 192.168.175.135:5060 (no NAT)
Using INVITE request as basis request - 
504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Found peer 'SMtrunk' for '3145152000' from 192.168.175.135:5060
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw|alaw|g722), peer - 
audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - 
(ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.122.57:18380
Looking for 913145152244 in from-pstn (domain devjones.com)

--- Reliably Transmitting (no NAT) to 192.168.175.135:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 
192.168.175.135;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4;received=192.168.175.135;rport=5060
Via: SIP/2.0/UDP 
192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP

Re: [asterisk-users] Trunk issue

2014-04-23 Thread Administrator TOOTAI 

Hello

Le 23/04/2014 15:36, Haley,Scott A a écrit :


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?




[...]

Here

[Apr 23 08:20:59] NOTICE[19026][C-0003]: chan_sip.c:25450 
handle_request_invite: Call from 'SMtrunk' (192.168.175.135:5060) to 
extension '913145152244' rejected because extension not found in 
context 'from-pstn'.

[...]

Regards

--
Daniel

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Re: [asterisk-users] Trunk issue

2014-04-23 Thread richard . seguin 
Are you using freeswitch, or just plain asterisk?  I just setup 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
Subject: [asterisk-users] Trunk issue

-- 
_
-- 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-usersI 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 to set 
utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted
Audio is at 18380
Adding codec 14 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 13 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.175.135:5060:
INVITE sip:913145152244@192.168.175.135 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Max-Forwards: 70
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Contact: sip:3145152000@192.168.122.57:5060
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Wed, 23 Apr 2014 13:20:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

--- SIP read from UDP:192.168.175.135:5060 ---
SIP/2.0 100 Trying
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
CSeq: 102 INVITE
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Content-Length: 0

-
--- (7 headers 0 lines) ---

--- SIP read from UDP:192.168.175.135:5060 ---
INVITE sip:913145152...@devjones.com SIP/2.0
P-AV-Message-Id: 1_1
Route: sip:192.168.122.57;lr;phase=terminating
Supported: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Date: Wed, 23 Apr 2014 13:20:59 GMT
Contact: 
sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68
Via: SIP/2.0/UDP 
192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4
Via: SIP/2.0/UDP 
192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967
Via: SIP/2.0/UDP 
192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947
Via: SIP/2.0/UDP 
192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135
Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632
Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr
Record-Route: 
sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr
Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr
P-Charging-Vector: icid-value=d13ae820-caef-11e3-9b9c-6c3be5a59e68
User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: Edward Jones sip:3145152...@devjones.com
From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f
To: sip:913145152244@192.168.175.135
Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Max-Forwards: 66
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 229
Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68
P-Location: 
SM;origlocname=Asterisk-2;origsiglocname=Asterisk-2;origmedialocname=Asterisk-2;termlocname=Asterisk-2;termsiglocname=Asterisk-2;smaccounting=true

v=0
o=root 424150695 424150695 IN IP4 192.168.122.57
s=Asterisk PBX 11.7.0
c=IN IP4 192.168.122.57
t=0 0
m=audio 18380 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
-
--- (27 headers 11 lines) ---
Sending to 192.168.175.135:5060 (no NAT)
Sending to 192.168.175.135:5060 (no NAT)
Using INVITE request as basis request - 
504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060
Found peer 'SMtrunk' for '3145152000' from 192.168.175.135:5060
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description