[asterisk-dev] SIP ghost calls

[Klaus Darilion]

On Thu, November 30, 2006 19:57, Hans Petter Selasky said:
> 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
>

its already an RFC: 4028

klaus


> 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
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