[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:

> Gene,
> 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  
> architecture?
>      | 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.


More information about the asterisk-dev mailing list