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