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--></style><title>Re: [asterisk-dev] SIP Not Showing
Disconnect</title></head><body>
<div>At 12:20 PM -0700 2007/7/22, Nicholas Blasgen wrote:</div>
<blockquote type="cite" cite><br></blockquote>
<blockquote type="cite" cite>I've been having some problems recently
with Asterisk thinking a SIP phone is still connected. These are
GrandStream Budge Tone 102's and even after someone hangs up the AGI
script I have running for them is still looping. The AGI script
keeps playing "beep" untill it's told to continue by the
user. But if the user has hung up the phone, the script just
keeps looping and looping, and that causes a bunch of problems for
me. Is there any way to do something on the line that would make
Asterisk realize the phone isn't connected anymore? I would have
thought something like playing a file would fix it all. Thought
Asterisk would require a response from the SIP phone saying it got the
message, and when the SIP phone didn't reply it would hang up the
line.</blockquote>
<blockquote type="cite" cite> </blockquote>
<blockquote type="cite" cite>=======================</blockquote>
<blockquote type="cite" cite> </blockquote>
<blockquote type="cite" cite> -- AGI Script
Executing Application: (PLAYBACK) Options: (beep)<br>
-- <SIP/josh-08be5690> Playing 'beep'
(language 'en')<br>
-- AGI Script Executing Application: (PLAYBACK)
Options: (beep)<br>
-- <SIP/josh-08c391b8> Playing 'beep'
(language 'en')<br>
www*CLI> sip show peers<br>
Name/username <span
></span>
Host
Dyn Nat ACL Port Status<br>
josh/josh <span
></span> <a
href="http://66.205.135.100"> 66.205.135.100</a> D
N 1024
UNREACHABLE<br>
</blockquote>
<blockquote type="cite" cite>=======================<br>
</blockquote>
<blockquote type="cite" cite>As you can see I've even unplugged the
SIP phone and Asterisk is still keeping TWO (2) lines open to user
"josh" even though Josh isn't reachable anymore. So
even removing the phone from the network doesn't make Asterisk realize
the phone isn't there anymore. Suggestions please!</blockquote>
<blockquote type="cite" cite><br>
--</blockquote>
<blockquote type="cite" cite>/Nick</blockquote>
<div><br></div>
<div>I'm uncertain if you've implemented the user-prompted method
correctly, since you do not include enough data to discern what is
happening with your system (and I would not suggest forwarding it to
this list, as it does not sound like your core question is a
development issue.)</div>
<div><br></div>
<div>There is an existing solution for this which does not require
prompts if you don't mind the media travelling through your Asterisk
system instead of directly between endpoints - look at the
descriptions for "rtptimeout" in your sip.conf file.
That can be discussed at length in the Asterisk-Users mailing list if
further explanation is required.</div>
<div><br></div>
<div>The second solution is to put a maximum call duration on calls so
that Asterisk hangs up the calls regardless of what the state of the
endpoint is - see the description of the function TIMEOUT(absolute).
That can be discussed at length in the Asterisk-Users mailing list if
further explanation is required.</div>
<div><br></div>
<div>The last method (that I can think of, at least) would require
Session-Timers, which currently do not exist in Asterisk. That
has been discussed on this list (asterisk-dev) within the past six
days:</div>
<div><br></div>
<div
>http://lists.digium.com/pipermail/asterisk-dev/2007-July/028574.html</div
>
<div><br></div>
<div>If you are interested in creating a patch to support these
features, it would be greatly appreciated. However, if your
methods or questions regard any of the first three methods (prompts,
RTPtimeout, or absolute timeout) then I would suggest that
asterisk-users is a more appropriate forum for discussion.</div>
<div><br></div>
<div>JT</div>
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