[asterisk-dev] Asterisk RTP timeout

Alexandr Olekhnovich a.olekhnovich at gmail.com
Mon Apr 23 09:02:03 MST 2007

There is an option in sip.conf named rtptimeout. That one means to terminate
the call if XX seconds of no activity, but it works only when there is no
activity from both sides. Is there any way to terminate the call if there is
no activity for XX seconds from one client?
I mean if one side of the conversation crashes, the call must be stopped.

Thanks in advance

Best Regards
Alexander Olekhnovich
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20070423/d5f64f92/attachment.htm

More information about the asterisk-dev mailing list