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<DIV dir=ltr align=left><SPAN class=921330723-01082007><FONT face=Arial
color=#0000ff size=2>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?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=921330723-01082007><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=921330723-01082007><FONT face=Arial
color=#0000ff size=2>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?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=921330723-01082007><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=921330723-01082007><FONT face=Arial
color=#0000ff size=2>Regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=921330723-01082007><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=921330723-01082007><FONT face=Arial
color=#0000ff size=2>Mike</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Anthony
Cennami<BR><B>Sent:</B> Wednesday, August 01, 2007 17:56<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[asterisk-users] Problem with the dial command<BR></FONT><BR></DIV>
<DIV></DIV>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. <BR><BR><BR>
<DIV><SPAN class=gmail_quote>On 8/1/07, <B class=gmail_sendername>Mike</B>
<<A href="mailto:list@virtutel.ca">list@virtutel.ca</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Thanks
Jared. It answers most of my question. Now, when the device
doesn't<BR>register, the behavior is as expected. But eventually,
any device that<BR>registers successfully might be unplugged, leaving Asterisk
to wonder where <BR>the device has gone.<BR><BR>So, what's the best approach
to this? Should I put a timeout=x minutes for<BR>that SIP
registration, and force the Polycom phone to reregister every y<BR>minutes (y
being smaller than x)? How do I do this? <BR><BR>Is this anyway to force
Asterisk to consider the peer disconnected if<BR>Asterisk doesn't get a reply
back within a second of trying a Dial command?<BR><BR>Is this any other
obvious option that escapes me?<BR><BR>Mike<BR><BR><BR><BR>-----Original
Message-----<BR>From: <A
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A><BR>[mailto:<A
href="mailto:asterisk-users-bounces@lists.digium.com">
asterisk-users-bounces@lists.digium.com</A>] On Behalf Of Jared Smith<BR>Sent:
Wednesday, August 01, 2007 14:54<BR>To: Asterisk Users Mailing List -
Non-Commercial Discussion<BR>Subject: Re: [asterisk-users] Problem with the
dial command <BR><BR>On Wed, 2007-08-01 at 11:43 -0400, Mike wrote:<BR>>
Aug 1 11:47:57 NOTICE[26107]: app_dial.c:1069 dial_exec_full:
Unable<BR>> to create channel of type 'SIP' (cause 3 - No route to
destination)<BR><BR>This happens when Asterisk don't know where to find the
peer (which is often<BR>the case if the device has failed to register to
Asterisk, for example).<BR><BR>> Sometimes, instead, the phone doesn't ring
and I get a 15 second <BR>> silence on the calling end. After
the full 15 seconds, Asterisk goes<BR>> to the next priority.<BR><BR>This
would happen, for example, if the phone registers with Asterisk but<BR>then
gets unplugged from the network. Asterisk has an IP address for the
<BR>peer and is trying to call it, but the peer isn't
responding.<BR><BR><BR>--<BR>Jared Smith<BR>Community Relations
Manager<BR>Digium,
Inc.<BR><BR><BR>_______________________________________________<BR>--Bandwidth
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clear=all><BR>-- <BR>Anthony Cennami </BODY></HTML>