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@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@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@alcan.mydomain.ca [BREAK]
2013-07-04 10:01:08.913500 [DEBUG] switch_core_session.c:861 Send signal sofia/sipinterface_1/barbara@alcan.mydomain.ca [BREAK]
2013-07-04 10:01:08.913500 [NOTICE] switch_ivr.c:1831 Transfer sofia/sipinterface_1/barbara@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@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@alcan.mydomain.ca)}

Any help is appreciated.


Thanks
Lloyd