[Freeswitch-users] attended transfer - fails

[Lloyd Aloysius]

After looking into the log. Why the call is transferred to a
endless_play_back ? Why this is happening ? Any help is appreciated.

2013-07-04 10:01:08.913500 [DEBUG] sofia.c:6808 Process REFER to [
23 at alcan.mydomain.ca]
2013-07-04 10:01:08.913500 [DEBUG] sofia.c:6834 Replaces: [aad7cec48bc2db7b]
2013-07-04 10:01:08.913500 [NOTICE] sofia.c:6920 Attended Transfer on
originating session 2c87f080-e4b2-11e2-85de-9f583ab4e053
2013-07-04 10:01:08.913500 [DEBUG] switch_ivr.c:1824 (sofia/sipinterface_1/
barbara at alcan.mydomain.ca) State Change CS_EXECUTE -> CS_ROUTING
2013-07-04 10:01:08.913500 [DEBUG] switch_core_session.c:1341 Send signal
sofia/sipinterface_1/barbara at alcan.mydomain.ca [BREAK]
2013-07-04 10:01:08.913500 [DEBUG] switch_core_session.c:861 Send signal
sofia/sipinterface_1/barbara at alcan.mydomain.ca [BREAK]
*2013-07-04 10:01:08.913500 [NOTICE] switch_ivr.c:1831 Transfer
sofia/sipinterface_1/barbara at alcan.mydomain.ca to
inline[endless_playback:local_stream://default,park@alcan.mydomain.ca]*


Lloyd


On Thu, Jul 4, 2013 at 10:27 AM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I could not figure it out why the  attended transfer is failing.
>
> UA1 - Call - UA2
>
> UA2 Answer
>
> UA2 Transfer - Dial UA3
>
> While UA3 is Ring UA2 Press the Transfer
>
> After transfer complete.
>
> *Issues*
>
> 1. Now UA1 hear the MOH only not the UA3 Ring back
>
> 2. After timeout the call never reach the UA3 voice mail ( This is the
> timeout option ) and the call is just hangup
>
>
> UA Bridge Data - {sip_invite_domain=alcan.mydomain.ca
> ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca
> ,originate_timeout=20}[leg_timeout=20]${sofia_contact(
> nelson at alcan.mydomain.ca)}
>
> Any help is appreciated.
>
>
> Thanks
> Lloyd
>
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