[asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys
Johansson Olle E
oej at edvina.net
Fri Jan 4 02:23:23 CST 2008
2 jan 2008 kl. 15.58 skrev Raj Jain:
> I'm trying to better understand what you've developed. It seems that
> you're trying to share Asterisk SLA trunks w/ a different system
> (Broadsoft). Is the following a correct representation of your
> | SLA Trunks
> ------------ -------------
> | Asterisk |---SIP---| Broadsoft |
> ------------ peers -------------
> | |
> | |
> Asterisk Broadsoft
> SLA Stations SLA Stations
> You speak about something called as CALL_INFO event-package and then
> reference it to Call-Info: header in RFC 3261. An event-package (RFC
> 3265) and header are two separate things. There is no standard
> CALL_INFO event package in SIP. Likewise, there is no standard Line-
> Seize event-package in SIP. It seems that these are Broadsoft
> proprietary SIP event-packages.
> It seems that you've developed these proprietary SIP event-packages
> in Asterisk. Why couldn't this be done using the standard dialog-
> package (RFC 4235)?
The broadsoft system was proposed to the IETF, but Dialog-Info was
developed as a replacement as far as I understand. The Broadsoft model
seemed a bit too complicated. Even so, it's implemented in a lot of
devices out there.
It's important that all these additions are either denied if we
haven't got a zaptel timer or just not compiled with #ifdef.
I would propably go for refusing the subscriptions. Regardless, we
need to be careful of module dependencies.
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