.. 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(a)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@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
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Greg Fausak
www.AddaBrand.com
(US) 469-546-1265
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