[asterisk-users] Channels hanging when SIP phone gets resetduringcall

[Steve Langstaff]

22 February 2007 12:22, Olle E Johansson wrote:
>
> 22 feb 2007 kl. 12.20 skrev Steve Langstaff:
> 
> > Are the RTP timers applicable with canreinvite=yes ?
> 
> how could we possibly check RTP if the RTP doesn't touch or 
> network card at all?

You can't. I realise.

> The timers are only used when we have RTP streams going to 
> us. If the RTP stream is redirected, it's up to the end 
> points to hangup due to media failure.

The endpoints have been rebooted, so they can't detect media failure
(unless they have some persistent store of call state over a reboot!).

> The way to solve this is to implement the SIP timer extension.

I see there is a discussion of this on the bug tracker...

http://bugs.digium.com/bug_view_page.php?bug_id=0000207

Looks like I'm going to be pushing the media through the server after
all...