[Asterisk-Dev] sip refer

[Richard]

I was able to get it working by creating a context for each sip domain. Then
change the sip REFER processing a little bit. Rather than sending it to the
original context, I send it to the context of the sip domain. It works very
well.

> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-
> bounces at lists.digium.com] On Behalf Of Olle E. Johansson
> Sent: Wednesday, October 20, 2004 8:20 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] sip refer
> 
> Richard wrote:
> 
> >>So my humble advice is: Fix your dialplan! If you are using SIP,
> >>the dialplan should be able to handle any sip URI.
> >
> > Not really. I don't define any domain or user in *. Everything is
> defined in
> > ser and * just uses autopeer. For example, if I have a 3 digit dialing
> for
> > company A and B. Both uses extension 200 but on different SIP domains.
> When
> > * gets a sip REFER, it will be 200 at companyA.com and 200 at companyB.com. If
> it
> > goes through the dial plan, there is no way to tell which company 200
> > belongs to. In chan_sip.c, I see that it strips the domain part and only
> > looks at the extension.
> 
> Check the SIPDOMAIN variable. If the domain doesn't show up there on a
> REFER, it's a bug.
> 
> /O
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