[Asterisk-Users] New Asterisk bounty: SIP simultaneous

[Paul Mahler]

The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same SIP device. That's WHY
the last device that registers gets the traffic. 

This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an
Asterisk issue. You should just be happy that Asterisk will do what you
want, even if SIP won't.  

If you really, really want to do this, up the bounty to about $50,000 and
get the SIP specification changed. 


Paul Mahler 
pmahler at signate.com 	
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
 94107-1901

 Asterisk Services and Training

 

 

 

 

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Andy Powell
> Sent: Sunday, July 11, 2004 9:57 AM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
> 
> 
> On 11/07/2004 at 08:42 Paul Mahler wrote:
> 
> >You are confused about what a SIP session is and what a SIP 
> session does.
> > 
> >
> >SIP, session initiation protocol, controls an RTP, real time 
> protocol, 
> >session between two IP endpionts. The end points have to 
> have unique IP 
> >addresses for the session to run. The unique SIP 
> registration is how * 
> >finds a UNIQUE endpoint.
> 
> Sorry, but this is irrelavant... SIP allows multiple 
> endpoints to register with the same account details and will 
> all ring when called. The fact that the rtp stream goes to 
> the first endpoint to pick up (and respond) is what's 
> important ie, if multiple devices are registered with the 
> same account they will *all* be 'spoken' to...  
> 
> Asterisk currently does not support this behaviour.
> 
> 
> >You don't want SIP to solve your problem, you want * to 
> solve your problem.
> >You are asking for this SIP "feature" because you are confused as to 
> >how SIP and * work, and how they work together.
> 
> No, the idea is to get asterisk to act like a real sip proxy. 
> The dialplan solution is a poor hack.
> 
> 
> >You can easily fix your business problem with *, but not 
> with mechanism 
> >you are asking for. You should spend your money on getting a copy of 
> >each of the two books that are now available and learn *. 
> Then it will 
> >be clear to you that you don't really want what you are asking for.
> 
> again, irrelavant - the whole beauty of the way SIP works is 
> that I can add to the list of phones that get called by 
> simply registering more phones with the same details. I don't 
> need my users to mess with or make a support call to add to 
> the dial plan. They can add and remove themselves.
> 
> I'd also suggest adding something like
> 
> registrationlimit=1 
> 
> for those that do not want to support multiple client 
> registrations, I'd also like to see the implementation of the 
> q parameter...
> 
> I'm all for this modification to SIP, although I'd probably 
> want to see DTMF callerid implemented first :D
> 
> Andy
> 
> 
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