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[asterisk-users] conversation record prematurely


 
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syscon780 at gmail.com
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PostPosted: Thu Sep 18, 2014 3:16 pm    Post subject: [asterisk-users] conversation record prematurely Reply with quote

I have following line in a context:

...
exten => _587NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten => _587NXXXXXX,n,MixMonitor(${recordfilename},b)
...

It records the conversation but it ends prematurely, after 10min. Why?
Where is the setting to records until a user hangup the handset.

--
Joseph

--
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rnewton at digium.com
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PostPosted: Thu Sep 18, 2014 6:59 pm    Post subject: [asterisk-users] conversation record prematurely Reply with quote

On Thu, Sep 18, 2014 at 3:16 PM, Joseph <syscon780@gmail.com> wrote:
Quote:
I have following line in a context:

...
exten =>
_587NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten => _587NXXXXXX,n,MixMonitor(${recordfilename},b)
...

It records the conversation but it ends prematurely, after 10min. Why?
Where is the setting to records until a user hangup the handset.

Without further information the only reason I could see would be the
'b' option in use for MixMonitor. If the channels were no longer
bridged it would stop recording. That is according to the
documentation.. which every once in a while is wrong. Other than that,
it should record as long as the channel is bridged.

Can you pastebin a log showing that particular call?[1]

[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

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

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syscon780 at gmail.com
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PostPosted: Thu Sep 18, 2014 7:14 pm    Post subject: [asterisk-users] conversation record prematurely Reply with quote

On 09/18/14 18:59, Rusty Newton wrote:
Quote:
On Thu, Sep 18, 2014 at 3:16 PM, Joseph <syscon780@gmail.com> wrote:
Quote:
I have following line in a context:

...
exten =>
_587NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten => _587NXXXXXX,n,MixMonitor(${recordfilename},b)
...

It records the conversation but it ends prematurely, after 10min. Why?
Where is the setting to records until a user hangup the handset.

Without further information the only reason I could see would be the
'b' option in use for MixMonitor. If the channels were no longer
bridged it would stop recording. That is according to the
documentation.. which every once in a while is wrong. Other than that,
it should record as long as the channel is bridged.

Can you pastebin a log showing that particular call?[1]

[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

How do I find out/verify if "b" option is used with "MixMonitor"?

I'll try to set the debugger on this one.

--
Joseph

--
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rnewton at digium.com
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PostPosted: Tue Sep 23, 2014 8:58 am    Post subject: [asterisk-users] conversation record prematurely Reply with quote

On Thu, Sep 18, 2014 at 7:14 PM, Joseph <syscon780@gmail.com> wrote:
Quote:

How do I find out/verify if "b" option is used with "MixMonitor"?


In the help text for an application you'll find the available
arguments and options.

On the Asterisk CLI you can view an application's help text with "core
show application <application name>". You might be interested in
checking out a wiki page (on this same subject) I added recently.[1]

In addition, the help text for applications, functions and more gets
added onto the wiki by scripts.[2]

[1]: https://wiki.asterisk.org/wiki/display/AST/CLI+Syntax+and+Help+Commands
[2]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Command+Reference

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

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
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