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Actually, what Hans is describing is a problem. Asterisk is setting up
the call and bridging the endpoints but it is not taking an active roll
in periodically verifying that the call is still active. Typically this
is done using a re-invite or something similar every 5-10 minutes in
order to get each endpoint to acknowledge that the call is still in
progress. Since calls may be bridged directly between endpoints once
asterisk is no longer seeing the RTP stream it will have no idea that
the call has died off - possibly because the endpoint(s) went offline
and/or failed.<br>
<br>
Periodically verifying that a call is still up between two endpoints
strikes me as very much the responsibility of a PBX and it would be
wise for this checking to be incorporated in Asterisk.<br>
<br>
Zoa wrote:
<blockquote cite="mid456F0EF9.60902@securax.org" type="cite"><br>
Google for rtptimeout
<br>
<br>
Zoa
<br>
<br>
Hans Petter Selasky wrote:
<br>
<blockquote type="cite">Hi,
<br>
<br>
I'm playing around with a new SIP phone connected to Asterisk 1.2.13. I
am performing some tests, and one of them is the "cable unplug" test.
<br>
<br>
During a call I unplugged the ethernet cable, and to my surprise, the
call was not disconnected. Even after 20 minutes. This is very bad!
<br>
<br>
Can anyone explain why Asterisk does not ping the SIP phone regularly,
and if there is no reply, disconnect all associated calls?
<br>
<br>
Just imagine what happens if two persons call eachother, using SIP
phones over the PSTN network, and both disconnect their phones. Who is
going to pay for the ghost call, which might last forever?
<br>
<br>
--HPS
<br>
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