[Freeswitch-users] How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call?

[Giovanni Maruzzelli]

Hi Jason,

I repetedly tested the scenario:

Skype->skypopen->FS->SIP->FS

and this mens, a skype call incoming to FS and then bridged via SIP to
another FS

you can test the same on your machine if you make the destination for
skypopen interface pointing to the extension 5000 of the default
dialplan (the standard IVR) and then press 1.

Can you please test the 5000 ext press 1, and confirm that it works
(or not) for you?

Can you then post (better in a Jira) the exact way to replicate your
problem? (dialplan, actions, configs, whatever)

Thanks for reporting,

-giovanni

On Tue, Aug 24, 2010 at 1:34 AM, Jason Jeffords <jason at cloudtree.net> wrote:
> Hi Giovanni,
> Our case 3 test was actually being bridged to a SIP endpoint (not in a
> conference, although
> this probably should not matter).
> We tested two type 3 cases:
> 1) Skype to FreeSWITCH Skype bridged to an extension of a registered SIP
> phone
> 2) Skype to FreeSWITCH Skype bridged to an outbound call through a PSTN
> gateway
> In both cases we are transiting FreeSWITCH, not terminating on it (could
> there be a
> race condition when doing signaling coordination to remote SIP endpoints,
> not FreeSWITCH
> itself?).
> When we terminate Skype calls on FreeSWITCH this works (case 2).  It also
> works
> for outbound Skype calls (case1 - SIP to FreeSWITCH, FreeSWITCH Skype to
> Skype).
> Also, we are running the very latest (well, as of this morning) git version,
> so that
> may introduce one more variable (if you are running an older version).
> Thanks for your help,
> Jason
>
> On Mon, Aug 23, 2010 at 6:36 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Hi Jason,
>>
>> with 3) - the problematic one - you mean an incoming call from a
>> remote skype to the fs-skype, that is then bridged to an outbound sip
>> call?
>>
>> I'm asking this because this is the most tested case, eg: in the
>> gateway for the FS-conference, where the incoming skype calls are then
>> bridged (through dialplan, like pressing "1" in the standard IVR) with
>> the sip conference. And it works with no problems...
>>
>> Let me know more, please.
>>
>> -giovanni
>>
>> On Mon, Aug 23, 2010 at 8:28 PM, Jason Jeffords <jason at cloudtree.net>
>> wrote:
>> > Hi Giovanni,
>> > We got some test cycles today and have found the following:
>> > 1) SIP to Skype calls work as desired - A-leg is SIP, B-leg is Skype
>> > 2) Skype to Internal Extensions Work as desired - A-leg is Skype, B-leg
>> > is
>> > FreeSwitch (internal extension)
>> > 3) Skype to SIP does not work - A-leg is Skype, B-leg is SIP
>> > In the last case:
>> > 1) We hear Skype ringing until the B-leg (SIP) is answered - this is
>> > correct
>> > 2) We then get dead air and the SIP B-leg hangs up on its own (i.e. the
>> > user
>> > answering does not hang up, but the connection hangs up)
>> > 3) The Skype client connection stays up with dead air until it is
>> > manually
>> > hung up
>> > We are running the latest code as of today.
>> > Any insight you could provide will be most appreciated.
>> > Thanks of all your work,
>> > Jason
>> > On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Jason,
>> >>
>> >> please test with the latest git version.
>> >>
>> >> Many messy changes to an already messy code, so maybe some bug or side
>> >> effect (ok, bug) has been introduced.
>> >>
>> >> commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090
>> >> Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
>> >> Date:   Fri Aug 13 16:19:20 2010 -0500
>> >>
>> >>    skypopen: now answer a call only when directed to do it (before
>> >> was trying to answer any incoming call). Lot of changes to a messy
>> >> part, so maybe some problem will come out...
>> >>
>> >>    Signed-off-by: Giovanni Maruzzelli <gmaruzz at gmail.com>
>> >>
>> >> -giovanni
>> >> --
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618