[asterisk-dev] SIP ghost calls

Johansson Olle E olle at voop.com
Thu Nov 30 12:36:00 MST 2006


30 nov 2006 kl. 20.15 skrev 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
>
Thanks for the update!

>> 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?

All depends on community efforts and funding...

/O


More information about the asterisk-dev mailing list