[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.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
Re: [asterisk-users] Dial Plan Issue
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
Re: [asterisk-users] MWI issue
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 extension does not answer it, I send it to the voicemail on the Avaya Messaging system for the extension that it came in on the Asterisk box. Once that happens, I need to send a MWI indicator to an application on the desktop of the Avaya User that there is a voicemail for that mailbox. I see the SIP Notify come in from Avaya for the extension (I did this with a tcpdump). My question is how do I configure Asterisk to act on that request and call an agi program to do what I want. Any help would be appreciated. Thanks, Scott Haley 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_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
[asterisk-users] compiling Asterisk
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 server CentOS 6.5. Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha...@edwardjones.commailto:scott.ha...@edwardjones.com 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_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
Re: [asterisk-users] compiling Asterisk
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-Commercial Discussion Subject: [asterisk-users] compiling Asterisk 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 server CentOS 6.5. Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha...@edwardjones.commailto:scott.ha...@edwardjones.com 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
Re: [asterisk-users] Simultaneous Ring
I have this working but I have one problem. I need to grab values from
variables that I have set in the calling context to dial. How would I do that.
[tbs-utils]
exten = s,1,NoOp(Entering tbs-utils for extension ${ARG1})
;Set local variables to be used in the call
same = n,Set(NUMBER=${ARG1})
same = n,Set(GLOBAL(DIALGROUP1)=)
same = n,Set(GLOBAL(DIALGROUP2)=)
same = n,Set(_VM=)
same = n,Set(_TIMER1=)
same = n,Set(_TIMER2=)
same = n,Set(BRANCH=)
same = n,Set(_TO_VM=0)
;Check to see if the Primary SIP trunk is up
same = n,Set(NETWORKSTATUS=${SIPPEER(${GLOBAL(TRUNK1)},status)})
;Setting the TRUNK variable based upon the status of whether Trunk1 is reachable
same =
n,Set(TRUNK=${IF($[$[NETWORKSTATUS=UNREACHABLE]]?${GLOBAL(TRUNK2)}:${GLOBAL(TRUNK1)})})
;Calling the agi script
same = n,AGI(agi://localhost/tbs.agi)
;Displaying the values of the variables set in the agi script
same = n,NoOp(Branch number is: ${BRANCH})
same = n,NoOp(DIALGROUP1 is: ${DIALGROUP1})
same = n,NoOp(DIALGROUP2 is: ${DIALGROUP2})
same = n,NoOp(TIMER1 is: ${TIMER1})
same = n,NoOp(TIMER2 is: ${TIMER2})
same = n,NoOp(VM is: ${VM})
same = n,NoOp(TO_VM is: ${TO_VM})
;Check to see if we should go straight to VM
same = n,Gotoif($[${TO_VM} = 1]?200:)
;Dial the primary number and to to the return status
same = n,Dial(Local/Group1-101@DelayLocal/Group2-101@Delay,30)
same = n,Hangup();
[Delay]
;Dial Group 1
exten = Group1-101,1,Verbose(2,Dialing Group 1 set of phones
${GLOBAL(DIALGROUP1)})
same = n,Dial(${DIALGROUP1},20,t)
;Dial Group 2
exten = Group2-101,1,Verbose(2,Dialing Group2 set of phones)
same = n,Verbose(2, Waiting 10 seconds before dialing)
same = n,Wait(10)
same = n,Dial(${DIALGROUP2},${TIMER2},t)
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: 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
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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 A J Stiles
Sent: Thursday, July 17, 2014 2:57 AM
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 to dial
another set of numbers while continuing to ring, or interrupting the
first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
This sounds exactly like the sort of situation for which local channels were
invented .
Dial(${DIALGROUP1}LOCAL/foo@bar) with a longer timeout than 10 seconds. Then
in your local channel, wait 10 and Dial(${DIALGROUP2}). The first Dial() will
be satisfied when someone answers either a phone in dial group 1, or a phone in
dial group 2 set ringing by the Dial() in the local channel.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] Simultaneous Ring
That worked. I had to use the *two* underscores in the agi script where I was setting the values. Thanks. Thanks, Scott Haley 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 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 set in the calling context to dial. How would I do that. I think you need to prefix your variable names with *two* underscores, to make them indefinitely heritable down the succession of channels. If they are prefixed with a single underscore, then they only get inherited *once*; so if the child channel spawns a grandchild, then any _VARS it inherited from the parent channel won't exist in the grandchild, but any __VARS will. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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
Re: [asterisk-users] Simultaneous Ring
Thanks AJ, this sounds like what I need.
Thanks,
Scott Haley
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.
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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
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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 A J Stiles
Sent: Thursday, July 17, 2014 2:57 AM
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 to dial
another set of numbers while continuing to ring, or interrupting the
first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
This sounds exactly like the sort of situation for which local channels were
invented .
Dial(${DIALGROUP1}LOCAL/foo@bar) with a longer timeout than 10 seconds. Then
in your local channel, wait 10 and Dial(${DIALGROUP2}). The first Dial() will
be satisfied when someone answers either a phone in dial group 1, or a phone in
dial group 2 set ringing by the Dial() in the local channel.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
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asterisk-users mailing list
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asterisk-users mailing list
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[asterisk-users] Simultaneous Ring
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 there a way to do this without interrupting the first call?
Thanks,
Scott Haley
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.
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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_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.
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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? 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. -- _ -- 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
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 -- _ -- 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
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 -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
Re: [asterisk-users] Trunk issue
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 -- _ -- 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
Re: [asterisk-users] Trunk issue
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 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. -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun
Re: [asterisk-users] Trunk issue
That seemed to fix it. Thanks to everyone. 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 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 -- _ -- 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
Re: [asterisk-users] Trunk issue
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
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
[asterisk-users] SIP 603 Declined error message
I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a SIP/2.0 603 Declined message. Here is the output. Any reason that this might be happening? It has been working up until now this week. I rebooted the machine on Tuesday. --- SIP read from TCP: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=z9hG4bK8066eb6f589ce3126b652973b4b00 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Supported: 100rel,histinfo,join,replaces,sdp-anat,timer Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH,UPDATE User-Agent: Avaya CM/R016x.02.0.823.0 Contact: Haley, Scott sip:3145152244@172.17.184.46;transport=tcp Route: sip:192.168.122.51;transport=tcp;lr;phase=terminating Accept-Language: en;q=1 Alert-Info: cid:internal@edj.devjones.com;avaya-cm-alert-type=internal History-Info: sip:51...@edj.devjones.com;index=1 History-Info: 51104 sip:51...@edj.devjones.com;index=1.1 Min-SE: 1200 P-Asserted-Identity: Haley, Scott sip:3145152...@edwardjones.com Record-Route: sip:172.17.184.46;transport=tcp;lr Session-Expires: 1200;refresher=uac Content-Type: application/sdp Content-Length: 257 v=0 o=- 1393419743 1 IN IP4 172.17.184.46 s=- c=IN IP4 172.17.184.93 b=AS:64 t=0 0 a=avf:avc=n prio=n a=csup:avf-v0 m=audio 28196 RTP/AVP 0 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 - --- (23 headers 13 lines) --- Sending to 172.17.184.46:31285 (NAT) Using INVITE request as basis request - 8066eb6f589ce3125b652973b4b00 Found peer 'trunk503in' for '3145152244' from 172.17.184.46:31285 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 127 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 127 Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) Peer audio RTP is at port 172.17.184.93:28196 Looking for 51104 in from-trunk-sip-trunk503out (domain edj.devjones.com) list_route: hop: sip:172.17.184.46;transport=tcp;lr --- Transmitting (NAT) to 172.17.184.46:31285 --- SIP/2.0 100 Trying Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Record-Route: sip:172.17.184.46;transport=tcp;lr From: Haley, Scott sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00 To: sip:51...@edj.devjones.com Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Server: FPBX-2.8.1(1.8.13.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1200;refresher=uac Contact: sip:51104@192.168.122.51:5060;transport=TCP Content-Length: 0 -- Executing [51104@from-trunk-sip-trunk503out:1] Set(SIP/trunk503in-010b, GROUP()=OUT_1) in new stack -- Executing [51104@from-trunk-sip-trunk503out:2] Goto(SIP/trunk503in-010b, from-trunk,51104,1) in new stack -- Goto (from-trunk,51104,1) -- Executing [51104@from-trunk:1] Set(SIP/trunk503in-010b, __FROM_DID=51104) in new stack -- Executing [51104@from-trunk:2] Gosub(SIP/trunk503in-010b, app-blacklist-check,s,1) in new stack -- Executing [s@app-blacklist-check:1] GotoIf(SIP/trunk503in-010b, 0?blacklisted) in new stack -- Executing [s@app-blacklist-check:2] Set(SIP/trunk503in-010b, CALLED_BLACKLIST=1) in new stack -- Executing [s@app-blacklist-check:3] Return(SIP/trunk503in-010b, ) in new stack -- Executing [51104@from-trunk:3] Gosub(SIP/trunk503in-010b, cidlookup,cidlookup_1,1) in new stack -- Executing [cidlookup_1@cidlookup:1] GotoIf(SIP/trunk503in-010b, 1?cidlookup,cidlookup_return,1) in new stack -- Goto (cidlookup,cidlookup_return,1) -- Executing [cidlookup_return@cidlookup:1] ExecIf(SIP/trunk503in-010b, 0?Set(CALLERID(name)=)) in new stack -- Executing [cidlookup_return@cidlookup:2] Return(SIP/trunk503in-010b, ) in new stack -- Executing [51104@from-trunk:4] ExecIf(SIP/trunk503in-010b, 0 ?Set(CALLERID(name)=3145152244)) in new stack
