[Asterisk-Dev] Re: multiple registrations of same credentials

[Jason Pyeron]

On Sat, 22 Oct 2005, Mikael Magnusson wrote:

> Olle E. Johansson wrote:
>> Jason Pyeron wrote:
>> 
>>>> On Sat, 22 Oct 2005, Olle E. Johansson wrote:
>>>> 
>>> 
>>>>>> Jason Pyeron wrote:
>>>>>> 
>>>> 
>>>>>> 
>>>>>> Remember that other SIP pbx's are not multiprotocol pbx's and can do
>>>>>> things differently. They usually fork in the SIP layer since it's the
>>>>>> same as the PBX layer. In Asterisk, those are two different layers, a
>>>>>> PBX and a channel driver.
>>>>>> 
>>> 
>>>> 
>>>> This does make it harder to do, but not impossible. Just think
>>>> DIAL(a&b&c&...).
>>>> 
>>>> now when DIAL(SIP/400)
>>>> 
>>>> gets back to chan_sip it gets corrected and replaced with an action for
>>>> each registered 400 extension using the pbx framework.
>>>> 
>>>> 
>> 
>> Which means that the PBX has no idea that it is in fact calling multiple
>> devices. We have to implement *two* forking schemes, which makes
>> everything much harder.
>> 
>> Believe me, I've spent a lot of time thinking about how to make this
>> work properly without breaking the Asterisk architecture.
>> 
>
> I think it can be solved by using both the peer (or AOR) and registered 
> contact address in the dial string. For example:
>
> Dial(SIP/400 at example.domain/sip:400 at a.b.c.d)
>
> I have implemented this scheme in a SIP channel based on eXosip and the 
> "forking" is done with a function (SIPPEER) which returns a dial string that 
> contains all current bindings of the peer.
>
> It can be used in the following way in the dial plan:
>
> exten 400,1,Dial(${SIPPEER(400 at example.domain})
>
> If both sip:400 at a.b.c.d and sip:400 at e.f.g.h have registered 
> sip:400 at example.domain, then the above Dial statement will expand to
>
> exten 
> 400,1,Dial(SIP/400 at example.domain/sip:400 at a.b.c.d&SIP/400 at example.domain/sip:400 at e.f.g.h)
>
> And it will use the following request URI:s and To headers in the generated 
> INVITEs.
>
>
> INVITE sip:400 at a.b.c.d SIP/2.0
> ..
> To: sip:400 at example.domain
> ..
>
>
> INVITE sip:400 at e.f.g.h SIP/2.0
> ..
> To: sip:400 at example.domain
> ..


so, have each phone have its own sub-domain and know all the phones ahead 
of time and the problem is solved?

Can I hear more about your implementation?

Thanks


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