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[ibc at aliax.net]

Hi, I read in:
http://wiki.freeswitch.org/wiki/Dialplan_Recipes
the following:

-----------------------
= Forked dial example =
Forked dial is when you want to attempt to ring 2 destinations at the
same time. Freeswitch will attempt to call both bridge options
simultaneously. The first bridge leg that replies with a 183 (session
progress) message will win the call, and the other bridge leg is
dropped. This only will work with late_negotiation enabled.
-----------------------

While the above could be valid when sending calls to varios (just
two?) PSTN gateways, I wonder why I couldn't send a call to two local
users (200 and 201) and wait until one of them *answers*. My expected
behaviour is:

- I call to FS extension 500.
- FS does a bridge to 200 and 201.
- 200 replies 180 "Ringing"
- 201 replies 183 "Session Progress" with early media (not very common
in a SIP phone but possible anyway)
- After a while 200 replies a 200 OK so FS sends a CANCEL to 201 and
replies 200 OK to the caller.

Is this common behaviour not possible in FS? is there othe way to get
it instead of using "bridge" app with two destinations in "data"?

Thanks.




--
Iñaki Baz Castillo
<ibc@aliax.net>
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[brian at freeswitch.org]

Disable 100rel on the sofia profile or update to latest SVN trunk. We have an issue on jira about this already and are working with the sofia team on this issue. The solution is to disable 100REL and it'll quit doing that.

/b

On Nov 20, 2008, at 3:13 AM, Iñaki Baz Castillo wrote:

[anthony.minessale at g...]

if you want to wait for the first one to answer instead of indicate progress you add
{ignore_early_media=true} to the beginning of the dial string
<action application="bridge" data="{ignore_early_media=true}sofa/profile/200@dom.com (200@dom.com),sofia/profile/201@dom.com (201@dom.com)"/>

On Thu, Nov 20, 2008 at 3:13 AM, Iñaki Baz Castillo <ibc@aliax.net (ibc@aliax.net)> wrote:

[ibc at aliax.net]

2008/11/20 Anthony Minessale <anthony.minessale@gmail.com>:

[anthony.minessale at g...]

We are not a proxy we are a b2bua so we are not going to send bob or carols sdp to alice
alice and FS have a private sdp between them.

We are not doing SIP forked dialing here, we are doing FS forked dialing that is designed to be
protocol agnostic. Keep that in mind because it's important to not lose track that we are not a
sip switch rather a protocol agnostic soft switch.

When either bob or carol send a 18X to mod_sofia their respective FS channel will translate it into FS specific code of either RING_READY(progress w/o media) or EARLY_MEDIA(progress w/ media) if it gets RING_READY it will in turn mark alice's channel RING_READY which will translate back into 180 to her phone. If it gets an EARLY_MEDIA it will translate back into a 183 and establish early media between FS and alice. If ignore_early_media is true the calls will continue to proceed and if alice has the variable "ringback" set to a tone description or media file that file/tone will begin to play until bob or carol answer. If it's not set then whichever one establishes 183 first will end the forked dial and alice will be bridged to that unanswerd call and hear the early media.

Again for good measure, we do not do SIP specific forked dialing/proxy fantasy that alice and bob and the white rabbit are having with the cattipilar and his hooka. I am glad you are here to provide a check and balance be be sure to respect my decisions no how the software works and you will be a welcome addition to our community. I will just warn you that too much SIP zealotry will piss me off no matter how nicely put. I am not worried about this because you have already admitted that the RFC was written by martians. =D







On Thu, Nov 20, 2008 at 9:18 AM, Iñaki Baz Castillo <ibc@aliax.net (ibc@aliax.net)> wrote:

[mike at jerris.com]

You have a catch-22 in this situation, which media path would you want
to pass along as you may have multiple? I think in this situation we
pass the media of the first media connection that we setup unless you
configure for fs to generate the ringback.

Mike


On Nov 20, 2008, at 10:18 AM, Iñaki Baz Castillo wrote:

[ibc at aliax.net]

2008/11/20 Michael Jerris <mike@jerris.com>:

[ibc at aliax.net]

2008/11/20 Anthony Minessale <anthony.minessale@gmail.com>: