[asterisk-dev] SIP session-timers: concept, discussion

[Grey Man]

> ----- Original Message ----
> From: John Todd <jtodd at loligo.com>
> To: asterisk-dev at lists.digium.com
> Sent: Wednesday, 18 July, 2007 12:20:27 AM
> Subject: [asterisk-dev] SIP session-timers: concept, discussion
>
> The issue of SIP session-timers has been raised before, and I'd like 
> to start a discussion here if there is any interest in implementing 
> this in Asterisk, and to solicit anyone who might think that they 
> would be up to the task of coding such a useful extension to the code.
> ...

This is a great idea! I wasn't aware of that SIP RFC but now having read it and checked that a lot of our customer ua's are indicating support for the timer (Netgear ATAs for one) it would solve a lot of call disputes we have as well!

I'm about to go off and explain to a customer why he was charged twice for a call that he claims wasn't even picked up. I end up doing this two or three times a month and invariably it's with respect to international GSM networks with nice hefty call charges. I've never been able to work out exactly why these calls don't hangup ( we use rtptimeout and rtpholdtimeout) and at the very least it's nice to know I'm not the only one :).

My company would also be happy yo conrtibute to any bounty put on this functionality. From my very limited knowledge of the sip channel code it may not be a huge job since there is already a scheduling mechanism that gets used for retransmits (ast_sched_add_variable) and the same approach could be incorporated for session timers.

Regards,

Greyman.






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