[asterisk-dev] SIP ghost calls

[Hans Petter Selasky]

On Thursday 30 November 2006 18:57, Johansson Olle E wrote:
> The proper way would be to develop the SIP timer extension.
>
> That would also solve the issue when you have external RTP bridges.
>

http://www.jdrosen.net/sip_timer.html

http://www.jdrosen.net/papers/draft-ietf-sip-session-timer-14.txt

When I turn on SIP debug, I see that my phone supports the "timer" extension.

Are there any plans to get this into Asterisk?

It should not be that difficult?

--HPS