[Serusers] RE: Transfer and Conferencing - Please help!

Atle Samuelsen clona at camaro.no
Fri Nov 5 06:56:18 CET 2004


.. this is proberbly not what you want, but look into sems, sems has a
confrancing module :-) And it works Gr8

-Atle

* Bob Carlson <bob.carlson at sigpro.com> [041105 00:46]:
> Thanks a lot for the pointer to Sip Scenario.  It looks like a fantastic
> tool.  I will take a look at Asterisk.
> 
> Thanks again, Bob
> 
> -----Original Message-----
> From: Greg Fausak [mailto:greg at addabrand.com] 
> Sent: Thursday, November 04, 2004 3:23 PM
> To: Bob Carlson
> Cc: 'SerUsers'
> Subject: Re: [Serusers] RE: Transfer and Conferencing - Please help!
> 
> Bob,
> 
> I can offer some ideas that might help.
> I certainly don't intend to be condescending...
> 
> Many of my questions are answered with call traces.
> For example, I worked on a bug with REINVITES today.
> 
> 	http://www.addaline.com/traces/andy_index.html
> 
> This is created by using a :
> 
> 1) switch with port monitoring
> 2) ethereal (or tcpdump) to grab data
> 3) sipscenario to format the data into the call trace
> 
> A transfer can be done in a few different ways, especially when
> you get an IP-PBX involved.  There is a popular one called
> Asterisk that can do transfers between extensions.  If you built
> it, get phones to register with it, and connected the outside with
> a SIP provider you could do some call traces and see how Asterisk
> makes it happen.
> 
> A conference is a different animal.  I don't think there is any
> SIP call per se to build a conference.  Some UAs have the
> function built in, and they actually create more than one phone call
> and mix the sound internally.  For example, the Cisco 7960 IP
> phone does that.
> 
>   I guess the basic problem is that SIP is a protocol, transfer is
> a feature that is implemented with the SIP protocol. There are
> quite a few ways to skin that cat :-)
> 
> -g
> 
> 
> On Nov 4, 2004, at 4:46 PM, Bob Carlson wrote:
> 
> > I sent this earlier and got no responses.  Perhaps this is not the 
> > right
> > forum to ask this question.  Can any one suggest a better place to go 
> > for
> > this information?
> >
> > Thanks, Bob
> >
> > -----Original Message-----
> > From: Bob Carlson
> > Sent: Wednesday, November 03, 2004 3:22 PM
> > To: 'SerUsers'
> > Subject: Transfer and Conferencing
> >
> > Let me apologize in advance for my question, which is a little 
> > rudimentary.
> > We are just starting a project that will use SER and I am being forced 
> > to
> > document right now how transfer and conferencing will be handled.  I 
> > have
> > spent a lot of time looking for definitive information on the subject 
> > with
> > no luck.  Well, maybe too much luck.  There seem to be many proposals 
> > and
> > models and so on, but it is not clear to me what is actually being 
> > done in
> > practice.  I have downloaded all the RFCs and proposal papers on the
> > subject.  I am still reviewing them, but I think the folks on this 
> > forum can
> > help me a lot.
> >
> > I need to know the SIP message sequences for performing a call 
> > transfer and
> > a blind call transfer and for constructing a conference.  I have found
> > information in proposals, but I need to know what actual, available SIP
> > phones can do.  We have some phones that we will test, but I do not 
> > know
> > what they do when you press their transfer and conference buttons.  
> > Pardon
> > me again for my impatience in asking before I have tried this out.
> >
> > The Transfer models are straightforward, but conferencing is more
> > complicated.  We must construct a simple conferencing model where the
> > conferencing is performed by a central server, a SIP IPX.  Only 
> > conferences
> > of 3 participants need to be supported.  We want it to look exactly 
> > like
> > 3-way calling on your home phone.  During a call, put the call on hold 
> > with
> > a conference button, call another phone, hit conference button, the two
> > calls are joined in a 3-way conference.
> >
> > The document draft-ietf-sipping-service-examples-07.txt seems to be 
> > very
> > helpful on the subject, but all examples are in the form of 3 or more 
> > UAs
> > and do not address any examples from the point of view of a PBX.  I 
> > can see
> > how to extend the examples to a PBX case, except for one aspect.  If 
> > the
> > IP-PBX is to perform the action as a proxy, what does the phone send 
> > the
> > IP-PBX to indicate the steps in the process.  Put more plainly, what 
> > happens
> > when the user hits the Transfer or Conference button on the phone?  
> > What
> > message is sent to the IP-PBX?
> >
> > Can anyone tell me where else I should be looking?  Is the service 
> > examples
> > draft the best base document to work from?
> >
> > Thanks in advance, Bob Carlson
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> Greg Fausak
> www.AddaBrand.com
> (US) 469-546-1265
> 
> 
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