[asterisk-dev] SIP disconnect code

[Carles Pina i Estany]

Hello,

On Jul/20/2008, Johansson Olle E wrote:
> 
> 20 jul 2008 kl. 11.03 skrev Carles Pina i Estany:
> 
> > I thought that SIP in Asterisk is enough important to include some
> > special variable (specially since we needed in a real case, together
> > with some other particular Header). I understand that, for maintenance
> > and leave Asterisk not-protocol-focused you decided to not include.
> 
> SIP is very important to Asterisk. But given our multiprotocol  
> architecture, we have
> to create solutions that work well and is supportable too. In order to  
> set a
> SIPHANGUPCAUSE on the INBOUND channel, it has to go through the core
> and it will be quite some coding to make it work. To do that, and  
> still not have a
> solution that would be much more useful than what we have today is,
> well, not something I am  motivated to work with on my own free time.

Some weeks ago, I fixed my/our problems with cause_sip2hangup function.
I don't really need it :-)

But I think that I could try for this case:
     Dial(SIP/EXTEN at IP,,)

So, outbound calls and one call. This would be the beginning, maybe not
100% perfect but better than nothing.

This would be similar to this function:
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header

(SIP_HEADER, so it's only for SIP channels and I think that cannot read
from BYE package, has some limitations but I understand that it's
usefull for some people)

> Remember that this is open source. A working patch that fits the
> architecture is always welcome.

Yes, but before spend X time doing/trying to do something I wanted to
ask here if there is real interest or only me has had this necessity.

Thank you,

-- 
Carles Pina i Estany		GPG id: 0x8CBDAE64
	http://pinux.info	Manresa - Barcelona