[asterisk-users] Channel names
Hi I'm using asterisk 1.8. How are channel names constructed. I always thought they were technology/peer-hex counter but I've had a lot of instances where a channel name doesn't have the correct peer as part of it. Is it unwise to use channel names to extract the peers involved in a call? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] CBAnn channel not going away in Asterisk 12
It may show up in 'bridge show all' - but I'd actually expect it not to show up there either. Actually, it does. I have a screen full of bridges with 0 channels. I just tried an experiment where all I have is exten = 329,1,Answer(1000) same = n,Confbridge(1234) with absolutely nothing else going on and those leak too. I need to understand why I'm seeing this and nobody else is. -- _ -- 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] Create new channel from dialplan
I'm not sure exactly what your use case is, but you could execute a Dial() and use the M option to execute a Macro (or U to execute a gosub). From there, the call routes into the macro/subroutine, and you can process away. After all of that is completed in the macro/subroutine, you can set MACRO_RESULT or GOSUB_RESULT (depending on which you used) to CONTINUE, so your dialplan continues after everything is complete (or, if you finish everything within the routine, just let it end there). Josh On Wed, Apr 30, 2014 at 10:21 PM, Igor Dvorzhak idm...@gmail.com wrote: Thanks, it almost what I need. But I can't find a way to pass channel variables to Originate cmd in dialplan. Is it possible at all? On Wed, Apr 30, 2014 at 3:13 PM, Richard Mudgett rmudg...@digium.comwrote: On Wed, Apr 30, 2014 at 4:33 PM, Igor Dvorzhak idm...@gmail.com wrote: Hi all, I need a command to originate a new channel from dialplan. I should be able to continue execution of the current context after this command. How to do this? Look at this application: *CLI core show application Originate Richard -- _ -- 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
Re: [asterisk-users] AGI GET DATA behavior
Hello, Yes, it is fairly simple, really. And I found out about the return code of Asterisk enabling the agi debugging, precisely. The documentation of the GET DATA entry specifies that Asterisk should either return -1, nothing, or the actual result of what the user entered, but never 0, am I right ? (http://www.voip-info.org/wiki/view/get+data) The problem is that Asterisk's behavior is not constant : 1 time out of 4 or 5, without ANY change in the behavior of the user, Asterisk simply does not wait for the user input, and returns 0 before the timeout. Le 30/04/2014 13:10, Thorsten Göllner a écrit : Is your script really so simple? Enable agi debugging (agi set debug on) and take look at it when this happens. -- _ -- 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] Putting a notice in the logs from the dialplan
Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] AGI GET DATA behavior
Le 30/04/2014 13:10, Thorsten Göllner a écrit : Is your script really so simple? Enable agi debugging (agi set debug on) and take look at it when this happens. On Thu, 1 May 2014, Hoggins! wrote: Yes, it is fairly simple, really. The problem is that Asterisk's behavior is not constant : 1 time out of 4 or 5, without ANY change in the behavior of the user, Asterisk simply does not wait for the user input, and returns 0 before the timeout. 0) Please don't top-post. 1) Reduce your script to the smallest example that illustrates your issue. 2) Post your script along with console output (with 'core agi set debug on,' 'core set debug 99,' and 'core set verbose 99') of a successful and a failed call. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] CBAnn channel not going away in Asterisk 12
On Thu, May 1, 2014 at 7:18 AM, Richard Kenner ken...@gnat.com wrote: It may show up in 'bridge show all' - but I'd actually expect it not to show up there either. Actually, it does. I have a screen full of bridges with 0 channels. I just tried an experiment where all I have is exten = 329,1,Answer(1000) same = n,Confbridge(1234) with absolutely nothing else going on and those leak too. I need to understand why I'm seeing this and nobody else is. Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. Thanks - Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Channel names
On Thu, May 1, 2014 at 7:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8. How are channel names constructed. I always thought they were technology/peer-hex counter but I've had a lot of instances where a channel name doesn't have the correct peer as part of it. Is it unwise to use channel names to extract the peers involved in a call? How a channel is named is a function of the channel technology. Which channel technology(ies) are you curious about? Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Channel names
On 1 May 2014 15:19, Matthew Jordan mjor...@digium.com wrote: On Thu, May 1, 2014 at 7:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8. How are channel names constructed. I always thought they were technology/peer-hex counter but I've had a lot of instances where a channel name doesn't have the correct peer as part of it. Is it unwise to use channel names to extract the peers involved in a call? How a channel is named is a function of the channel technology. Which channel technology(ies) are you curious about? Matt SIP only Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Putting a notice in the logs from the dialplan
On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance Nope, unfortunately not. It would be a relatively trivial addition to add a dialplan application that could emit an Asterisk logging message at any one of the various levels, if someone were interested. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, Apr 30, 2014 at 8:37 AM, Administrator TOOTAI ad...@tootai.netwrote: Le 30/04/2014 15:19, Matthew Jordan a écrit : On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI ad...@tootai.netmailto: ad...@tootai.net wrote: Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! I asked you not to clone and issues and to take your issue to the mailing list (which you did, thank-you). Cloning issues makes a mess of the issue tracker, and causes information to get lost. If your issue is deemed to be a bug, the original issue will get re-opened. I cloned the issue as it is a bug and I could explain how to reproduce it. If I shouldn't clone the bug, please explain me how to do to inform developpers about new informations concerning a closed bug. That say, sorry for inconvenience. 1. Bug marshals watch the asterisk-bugs mailing list. All updates to all issues in JIRA get sent to that mailing list - even comments on closed issues. 2. Bug marshals also hang out in the #asterisk-bugs IRC channel. You can talk to a bug marshal in that channel as well. 3. Finally, we all watch the mailing lists (pretty much all of the mailing lists, no less). This is all documented on the Asterisk wiki's [1] Asterisk Issue Guidelines. There's even a checkbox when you file an issue that asks if you read the guidelines... you did read them, right? :-) [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Submittingthebugreport -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Putting a notice in the logs from the dialplan
On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik i...@pack-net.co.ukmailto:i...@pack-net.co.uk wrote: Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance What about the Log application? It is available on our Asterisk 1.8.26 box. Connected to Asterisk 1.8.26.0 Verbosity is at least 3 CLI core show application Log -= Info about application 'Log' =- [Synopsis] Send arbitrary text to a selected log level. [Description] Sends an arbitrary text message to a selected log level. [Syntax] Log(level,message) [Arguments] level Level must be one of 'ERROR', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE' or 'DTMF'. message Output text message. [See Also] Not available -- _ -- 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] CBAnn channel not going away in Asterisk 12
Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. I need to wait until I'm sure this isn't something I caused somehow, so I need to first understand why I'm seeing this and nobody else is. -- _ -- 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] Putting a notice in the logs from the dialplan
That works a treat, thank you. On 1 May 2014 15:28, Steven Wheeler swhee...@usinternet.com wrote: On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance What about the Log application? It is available on our Asterisk 1.8.26 box. Connected to Asterisk 1.8.26.0 Verbosity is at least 3 CLI core show application Log -= Info about application 'Log' =- [Synopsis] Send arbitrary text to a selected log level. [Description] Sends an arbitrary text message to a selected log level. [Syntax] Log(level,message) [Arguments] level Level must be one of 'ERROR', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE' or 'DTMF'. message Output text message. [See Also] Not available -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 01/05/2014 16:28, Matthew Jordan a écrit : On Wed, Apr 30, 2014 at 8:37 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: Le 30/04/2014 15:19, Matthew Jordan a écrit : On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net mailto:ad...@tootai.net mailto:ad...@tootai.net wrote: Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! I asked you not to clone and issues and to take your issue to the mailing list (which you did, thank-you). Cloning issues makes a mess of the issue tracker, and causes information to get lost. If your issue is deemed to be a bug, the original issue will get re-opened. I cloned the issue as it is a bug and I could explain how to reproduce it. If I shouldn't clone the bug, please explain me how to do to inform developpers about new informations concerning a closed bug. That say, sorry for inconvenience. 1. Bug marshals watch the asterisk-bugs mailing list. All updates to all issues in JIRA get sent to that mailing list - even comments on closed issues. 2. Bug marshals also hang out in the #asterisk-bugs IRC channel. You can talk to a bug marshal in that channel as well. 3. Finally, we all watch the mailing lists (pretty much all of the mailing lists, no less). This is all documented on the Asterisk wiki's [1] Asterisk Issue Guidelines. There's even a checkbox when you file an issue that asks if you read the guidelines... you did read them, right? :-) [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Submittingthebugreport I did read them a long long time ago. Anyway, my bad, sorry for that. Regards -- Daniel -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
In my experience DNS issues will cause Asterisk to take a long time to reload and could stop Asterisk for working at all. List all the IPs of the box in /etc/hosts and make sure /etc/resolv.conf points to a working nameserver. See if that helps at all. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Thursday, May 01, 2014 11:04 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ? Le 01/05/2014 16:28, Matthew Jordan a écrit : On Wed, Apr 30, 2014 at 8:37 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net wrote: Le 30/04/2014 15:19, Matthew Jordan a écrit : On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI ad...@tootai.net mailto:ad...@tootai.net mailto:ad...@tootai.net mailto:ad...@tootai.net wrote: Please, people from Digium, Matt again closed the new bug ASTERISK-23689 I opened (clone from 23683) telling that it's not a bug. Did he carefully read the comments on the new bug? If not, please forward him this email, *it's* a bug or you have to explain me why it is not! -- _ -- 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] Create new channel from dialplan
Hi John,
Detailed use case:
I have a contact call connected to which played the IVR. On transfer
digit key press I need to transfer it to the predefined transfer number.
I don't want use the Dial command to do transfer because it will
automatically connect contact channel with transfer channel. Instead I
want to put transfer channel and contact channel in the MeetMe
conference.
So I need to use Originate command to originate transfer call in the
separate channel. After this command I will put current contact channel
in MeetMe conference. When transfer connects it will be put to the same
MeetMe conference as contact.
Now I'm using the System command to originate call through external
executable in dialplan, but it looks ugly, so I'm trying to find a better
way to do this.
What I'm using now:
exten = meetme,1,System(/usr/bin/php
/etc/asterisk/script/originate_call.php Channel:
Local/+${TRANSFER_NUMBER}@cf3-transfer-dial\,MaxRetries: 0\,RetryTime:
60\,WaitTime: 640\,CallerID: \${TO}\\,Context:
cf3-transfer-leg\,Extension: s\,Priority: 1\,Set:
ACTIVE_ID=${ACTIVE_ID}\,Set: TERMINATION_IP=${TERMINATION_IP}\,Set:
DIAL_TIME=${DIAL_TIME}\,Set:
TRANSFER_TEST_NUMBER=${TRANSFER_TEST_NUMBER}\,)
Igor
On Thu, May 1, 2014 at 5:23 AM, Josh Metzger joshdmetz...@gmail.com wrote:
I'm not sure exactly what your use case is, but you could execute a Dial()
and use the M option to execute a Macro (or U to execute a gosub).
From there, the call routes into the macro/subroutine, and you can process
away. After all of that is completed in the macro/subroutine, you can set
MACRO_RESULT or GOSUB_RESULT (depending on which you used) to
CONTINUE, so your dialplan continues after everything is complete (or, if
you finish everything within the routine, just let it end there).
Josh
On Wed, Apr 30, 2014 at 10:21 PM, Igor Dvorzhak idm...@gmail.com wrote:
Thanks, it almost what I need.
But I can't find a way to pass channel variables to Originate cmd in
dialplan.
Is it possible at all?
On Wed, Apr 30, 2014 at 3:13 PM, Richard Mudgett rmudg...@digium.comwrote:
On Wed, Apr 30, 2014 at 4:33 PM, Igor Dvorzhak idm...@gmail.com wrote:
Hi all,
I need a command to originate a new channel from dialplan. I should be
able to continue execution of the current context after this command.
How to do this?
Look at this application:
*CLI core show application Originate
Richard
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Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 01/05/2014 17:24, Eric Wieling a écrit : In my experience DNS issues will cause Asterisk to take a long time to reload and could stop Asterisk for working at all. List all the IPs of the box in /etc/hosts and make sure /etc/resolv.conf points to a working nameserver. See if that helps at all. As explained in one on my previous message, it's a bug, easily reproducible: take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like this (what is important is the #include): [general] persistentmembers=yes #include local/queues.d/*.conf Now modify one of the .conf file in directory local/queues.d and do a CLI module reload app_queue.so you wil get NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. despite the fact that modification was done in a .conf file. I took this example as with module reload app_queue the above message appears. For sip, iax, voicemail, aso there is no message, just SIP reload or ... To make asterisk take the modification in account, you have to open /etc/asterisk/[sip|iax|voicemail|queue|..].conf and save it without making any change. After this the command will be execute. It you run it a second time in a raw, you will see that the false behavior appears again till you again open/save the original file. -- Daniel -- _ -- 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] Need help troubleshooting Asterisk Auto dial out problem
On 4/30/2014 7:24 PM, Jesse Thompson wrote: impacted. However new files introduced into /var/spool/asterisk/outgoing/ folder get ignored. No messages spring up on asterisk -rvv console, nothing shows up in the logs, the .call files just get snubbed. We're at a loss to Are the new files being named uniquely ? there are bugs (e.g., jira# 11291) that have to do with files having the same name. my solution was to add .$$ on the end of the filename to ensure it was unique. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help troubleshooting Asterisk Auto dial out problem
Are the new files being named uniquely ? there are bugs (e.g., jira# 11291) that have to do with files having the same name. my solution was to add .$$ on the end of the filename to ensure it was unique. Yep, the files get a -MM-DD_HH:ii:ss- timestamp prefix in their names before being mv'ed into the spool directory (same filesystem) and are never realistically fired of more than once per second. Logic behind this was that after they get moved automatically into the outgoing_done/ folder by asterisk, we've got a rough log in the filenames of which alarms got tripped at what times. :) - - Jesse -- _ -- 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] CBAnn channel not going away in Asterisk 12
On 5/1/2014 10:38 AM, Richard Kenner wrote: Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. I need to wait until I'm sure this isn't something I caused somehow, so I need to first understand why I'm seeing this and nobody else is. I had seen it as well but just chalked it up to not grokking how the CBAnn channels worked. -- _ -- 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
