[asterisk-users] Problem with the dial command

Mike list at virtutel.ca
Wed Aug 1 18:10:00 CDT 2007


Thanks.  Tell me, how intensive is it to use qualify?  What type of
packet/check is done with this? Is it reasonnable to use qualify for
thousands of devices?
 
Once the device is considered to be unreachable for any number of reasons,
will another poll of the device be done to check if it became available
again after the configured number of milliseconds?  Or will it be considered
unreachable until the next register attempt by the device?
 
Regards,
 
Mike

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony
Cennami
Sent: Wednesday, August 01, 2007 17:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with the dial command


qualify=yes in the sip.conf context for that device will change the device
to unreachable and should send you directly to voicemail.  There could still
be a brief period where the device is timed out and the system hasn't
qualified it yet, but outside of that, it will just continue trying to send
to the device. 



On 8/1/07, Mike <list at virtutel.ca> wrote: 

Thanks Jared. It answers most of my question.  Now, when the device doesn't
register, the behavior is as expected.  But eventually, any device that
registers successfully might be unplugged, leaving Asterisk to wonder where 
the device has gone.

So, what's the best approach to this?  Should I put a timeout=x minutes for
that SIP registration, and force the Polycom phone to reregister every y
minutes (y being smaller than x)? How do I do this? 

Is this anyway to force Asterisk to consider the peer disconnected if
Asterisk doesn't get a reply back within a second of trying a Dial command?

Is this any other obvious option that escapes me?

Mike



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:  <mailto:asterisk-users-bounces at lists.digium.com>
asterisk-users-bounces at lists.digium.com] On Behalf Of Jared Smith
Sent: Wednesday, August 01, 2007 14:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with the dial command 

On Wed, 2007-08-01 at 11:43 -0400, Mike wrote:
> Aug  1 11:47:57 NOTICE[26107]: app_dial.c:1069 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 3 - No route to destination)

This happens when Asterisk don't know where to find the peer (which is often
the case if the device has failed to register to Asterisk, for example).

> Sometimes, instead, the phone doesn't ring and I get a 15 second 
> silence on the calling end.  After the full 15 seconds, Asterisk goes
> to the next priority.

This would happen, for example, if the phone registers with Asterisk but
then gets unplugged from the network.  Asterisk has an IP address for the 
peer and is trying to call it, but the peer isn't responding.


--
Jared Smith
Community Relations Manager
Digium, Inc.


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-- 
Anthony Cennami 
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