<br><br><div class="gmail_quote">2010/9/7 Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">On 09/06/2010 10:28 AM, Olivier wrote:<br>
> Hi,<br>
><br>
> With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :<br>
><br>
> - extension 7002 is a SIP hard phone currently configured to forward<br>
> incoming calls to extension 7003, when a call is unanswered within a 10s<br>
> time frame<br>
><br>
> - when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20)<br>
> statement and no one answers, then :<br>
> - after 10s, Asterisk receives "SIP 302 Moved temporarily" message and<br>
> enters its dialplan to call 7003, as required,<br>
> - 10s later (or 20s from the very start), call from 7001 to 7003 is cut<br>
> and the next statement after Dial(SIP/7002,20) is run.<br>
><br>
> The behaviour I would ideally implement is :<br>
> - whenever a "SIP 302 Moved temporarily" message is received, timer<br>
> associated to the original call (the one from 7001 to 7002) is reset to<br>
> another 20s period<br>
<br>
</div>That would change the logic in app_dial quite drastically, as it would<br>
have to remember separate timeouts for each of the originally-dialed<br>
destinations in case they get forwarded elsewhere.<br>
<br></blockquote><div><br>Yes.<br><br>Anyway, I think I found a workaround using a second Dial statement :<br>after the first Dial times out, I check if another call (the one launched after receiving the 302 Moved Temporarily message) have been started and interrupted by the first Dial timer.<br>
If positive, then another Dial is triggered with appropriate settings (new timeout and and extension values).<br><br>Thanks<br><br> <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
--<br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href="mailto:kfleming@digium.com">kfleming@digium.com</a><br>
Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>