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asteriskusers at dovid...
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 Posted: Wed Jan 02, 2008 9:20 pm Post subject: [asterisk-users] How to automaticaly close calls whenAsteris |
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----- Original Message -----
From: "Jared Smith" <jsmith at digium.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Tuesday, December 18, 2007 4:48 PM
Subject: Re: [asterisk-users] How to automaticaly close calls whenAsterisk
didn't receive the bye request ?
| Quote: | On Tue, 2007-12-18 at 15:20 +0100, Anthony Chapellier wrote:
| Quote: | I'd like to know if it's possible to configure Asterisk to automaticaly
close calls when the BYE request hasn't been sent by any clients and the
call still exists for Asterisk ?
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There is a SIP timers patch in the bug tracker (see
http://bugs.digium.com/view.php?id=10665) that currently implements
this, and it's being tested in the team/group/sip_session_timers/ branch
in SVN. Please test this out and help provide feedback, so that we can
get this put into Asterisk in time for the next major release.
I'd also like to take this opportunity to thank John Todd and Raj Jain
for their hard work on this feature -- this is a great example of
patches being submitted to Asterisk with great documentation, a detailed
explanation of the current limitations, an explanation of the standard,
implementation details, and a test plan. Good job guys!
---
Jared Smith
Community Relations Manager
Digium, Inc.
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Jared,
I would think of using rtptimeout. There is a reason why you did not mention
it and I am curious as to why. |
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rj2807 at gmail.com
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| Posted: Thu Jan 03, 2008 5:51 am Post subject: [asterisk-users] How to automaticaly close calls whenAsteris |
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The rtptimeout feature has a few limitations:
. It is ineffective when the RTP is not terminated on Asterisk.
. It can cause false session hangups if the remote SIP UA does not support
silence suppression
. The companion rtpholdtimeout can cause false hangups if you make incorrect
judgment on how long a call hold can last.
. The rtptimeout period is not negotiated throughout the SIP signaling path
i.e. between the UAC, UAS, and intermediary proxies. So it does not help
clear the session state throughout the network (when your BYE doesn't make
it to all the entities in the SIP signaling path).
The SIP session-timers feature addresses all of the above limitations.
--
Raj
Jared,
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