[asterisk-users] Problem with the dial command

[Anthony Francis]

Mike wrote:
> 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 <mailto: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:asterisk-users-bounces at lists.digium.com>
>     [mailto: asterisk-users-bounces at lists.digium.com
>     <mailto: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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Yes it continues to try to qualify after it is down. When it 
successfully qualifies again it triggers a peer reachable event. I 
personally have my servers qualify every peer, it does not add a 
noticeable amount of resource utilization.

Anthony