if you provide a console trace of both situations with console loglevel debug and put them on pastebin i can tell you what's happening.


On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
Thanks Anthony,

however this rather deteriorated the situation.
Now it works the following
- A calls B
- B enters *4 gets an announcement and enters digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C both have no voice (and B is
dropped)
- When A hangs up, C hangs up

Before it did:
- A calls B
- B enters *4 gets an announcement and enters digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C are connected and B is dropped
- When A hangs up, C hangs up

Best regards
Peter

Anthony Minessale schrieb:
> hit send too soon
> you want to set  loopback_bowout=false
>
> This keeps loopback from trying to destroy itself when it sees a
> chance to cut out of the call path.
>
>
> On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale
> <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
>
>
>     set/export the channel variable loopback_bowout=true so it's on
>     the loopback leg
>
>
>
>
>     On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX
>     <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
>
>         Hello,
>
>         I have some problems with attended transfer and loopback
>
>         Scenario how id <does> work
>         - A calls B
>         - B enters *4 gets an announcement and enter digits for C (A
>         get MOH)
>         - C is called
>         - As soon as C picks up the call, A and C are connected and B
>         is dropped
>
>         How it <should> work until here:
>         - A calls B
>         - B enters *4 gets an announcement and enter digits for C (A
>         get MOH)
>         - C is called
>         - As soon as C picks up the call, B and C are connected (A
>         still MOH)
>
>         The dial string for C is dynamic and dependent on certain
>         parameters,
>         therefore C must be called via Loopback in our scenario.
>
>
>         Here are the configs:
>         In dialplan for calling B:
>         <anti-action application="bind_meta_app" data="4 b b
>         execute_extension::attended_xfer XML default"/>
>
>         Dialplan for executing the att_xfer:
>         <extension name="attended_xfer">
>            <condition field="destination_number"
>         expression="^attended_xfer$">
>              <action application="set" data="continue_on_fail=true"/>
>              <action application="read" data="3 4 ivr/ivr-enter_ext.wav
>         attxfer_callthis 30000 #"/>
>              <action application="set" data="origination_cancel_key=#"/>
>              <action application="att_xfer"
>         data="loopback/${attxfer_callthis}"/>
>            </condition>
>         </extension>
>
>         So this is pretty standard, except the loopback. SVN is 15322.
>
>         Anybody has a solution for this?
>
>
>         Best regards
>         Peter
>
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users@lists.freeswitch.org
>         <mailto:FreeSWITCH-users@lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     --
>     Anthony Minessale II
>
>     FreeSWITCH http://www.freeswitch.org/
>     ClueCon http://www.cluecon.com/
>     Twitter: http://twitter.com/FreeSWITCH_wire
>
>     AIM: anthm
>     MSN:anthony_minessale@hotmail.com
>     <mailto:MSN%3Aanthony_minessale@hotmail.com>
>     <mailto:PAYPAL%3Aanthony.minessale@gmail.com>
>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>     FreeSWITCH Developer Conference
>     sip:888@conference.freeswitch.org
>     <mailto:sip%3A888@conference.freeswitch.org>
>     <http://iax:guest@conference.freeswitch.org/888>
>     <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
>     pstn:213-799-1400
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale@hotmail.com
> <mailto:MSN%3Aanthony_minessale@hotmail.com>
> <mailto:PAYPAL%3Aanthony.minessale@gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888@conference.freeswitch.org
> <mailto:sip%3A888@conference.freeswitch.org>
> <http://iax:guest@conference.freeswitch.org/888>
> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400