asterisk users - Jan 2013 - DAHDI: How to know since when it is used? How to shutdown after max time?

Hi;

How can I know the duration that the DAHDI channel is still used? I need to know
its status and since when it is in this status, how?

Also, is it possible to hangup the channel if it has been openned more than 90
minute? Other than using the timeout in the Dial command (because this I know
it).

What is happening with me that from time to time, I find some DAHDI channels are
stayed connected (stuck) for long time. I know how to write the extensions.conf
in a way to handle the hangup properly, also I send the incoming calls to the
voicemail to be sure it is hanged up properly. One more thing, I set the
rtptimeout in case there is any problem in the sip phone and the network .. But,
still after sometime, I am surprised that some channels are stuck and stayed
connected and then I have to reset it manually !! This is happening only in the
analoge channels.

What other than the rtptimeout, the hangup in the extensions.conf, the
voicemail? Is there anything I have to take care for it that might cause this
stuck and keeping the channel openned?

By the way, for such cases, what should I place the value of the rtpkeepalive as
currently it is 0?

What other things I have to take care for it?

Regards
Bilal

2013/1/3 bilal ghayyad <bilmar_gh at yahoo.com>
> Hi;
>
> How can I know the duration that the DAHDI channel is still used? I need
> to know its status and since when it is in this status, how?
>
> Also, is it possible to hangup the channel if it has been openned more
> than 90 minute? Other than using the timeout in the Dial command (because
> this I know it).
>
> What is happening with me that from time to time, I find some DAHDI
> channels are stayed connected (stuck) for long time. I know how to write
> the extensions.conf in a way to handle the hangup properly, also I send the
> incoming calls to the voicemail to be sure it is hanged up properly. One
> more thing, I set the rtptimeout in case there is any problem in the sip
> phone and the network .. But, still after sometime, I am surprised that
> some channels are stuck and stayed connected and then I have to reset it
> manually !! This is happening only in the analoge channels.
>
> What other than the rtptimeout, the hangup in the extensions.conf, the
> voicemail? Is there anything I have to take care for it that might cause
> this stuck and keeping the channel openned?
>
> By the way, for such cases, what should I place the value of the
> rtpkeepalive as currently it is 0?
>
> What other things I have to take care for it?
>
> Regards
> Bilal
>
>
I checked on my PBX and I find no way to identify the duration of a call
involving a DAHDI channel like it happens on SIP channels. I think the only
way will be to assign a not so huge AbsoluteTimeout to each call.

Leandro
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130103/811b5a48/attachment.htm>

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Thursday, January 03, 2013 2:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI: How to know since when it is used? How
to shutdown after max time?

 

 

2013/1/3 bilal ghayyad <bilmar_gh at yahoo.com>

Hi;

How can I know the duration that the DAHDI channel is still used? I need to
know its status and since when it is in this status, how?

Also, is it possible to hangup the channel if it has been openned more than
90 minute? Other than using the timeout in the Dial command (because this I
know it).

What is happening with me that from time to time, I find some DAHDI channels
are stayed connected (stuck) for long time. I know how to write the
extensions.conf in a way to handle the hangup properly, also I send the
incoming calls to the voicemail to be sure it is hanged up properly. One
more thing, I set the rtptimeout in case there is any problem in the sip
phone and the network .. But, still after sometime, I am surprised that some
channels are stuck and stayed connected and then I have to reset it manually
!! This is happening only in the analoge channels.

What other than the rtptimeout, the hangup in the extensions.conf, the
voicemail? Is there anything I have to take care for it that might cause
this stuck and keeping the channel openned?

By the way, for such cases, what should I place the value of the
rtpkeepalive as currently it is 0?

What other things I have to take care for it?

Regards
Bilal

 

I checked on my PBX and I find no way to identify the duration of a call
involving a DAHDI channel like it happens on SIP channels. I think the only
way will be to assign a not so huge AbsoluteTimeout to each call. 

 

My suggestion would be to either do a cron job that executes asterisk -rx
"core show channels verbose" and kill anything with a duration over 90
minutes or do the same thing with an AMI task (cron optional here). 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130103/b23fa499/attachment.htm>

On Wed, Jan 2, 2013 at 5:39 PM, bilal ghayyad <bilmar_gh at yahoo.com>
wrote:
> Hi;
>
> How can I know the duration that the DAHDI channel is still used? I need
> to know its status and since when it is in this status, how?
>
> Also, is it possible to hangup the channel if it has been openned more
> than 90 minute? Other than using the timeout in the Dial command (because
> this I know it).
>
> What is happening with me that from time to time, I find some DAHDI
> channels are stayed connected (stuck) for long time. I know how to write
> the extensions.conf in a way to handle the hangup properly, also I send the
> incoming calls to the voicemail to be sure it is hanged up properly. One
> more thing, I set the rtptimeout in case there is any problem in the sip
> phone and the network .. But, still after sometime, I am surprised that
> some channels are stuck and stayed connected and then I have to reset it
> manually !! This is happening only in the analoge channels.
>
> What other than the rtptimeout, the hangup in the extensions.conf, the
> voicemail? Is there anything I have to take care for it that might cause
> this stuck and keeping the channel openned?
>
>
What kind of trunks you are using? Analog? Digital (E1, T1)?

Saludos/Regards
--
Ing. Gerardo Barajas Puente

Proyectos Especiales/Preventa | www.neocenter.com
T:+52 (55)  8590-9000 x 7003
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130103/f4dd3dcd/attachment.htm>