[Freeswitch-users] Cannot transfer incoming call

[Tomasz Hyziak]

Hi

Transfers from local to local user work without problem, but when a
person call in via sip provider (selects call type via ivr and he is
put into fifo) and somebody answers this call, he is unable to
transfer this call to another person (via *1/*3/*4). When he presses
*, he is disconnected. Any ideas why ?

2011-08-19 14:55:17.475054 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF *:800
2011-08-19 14:55:17.475054 [DEBUG] switch_channel.c:2775
(sofia/external/693xxxxxx at trunk.dialog.pl) Callstate Change ACTIVE ->
HANGUP


--
pozdrawiam - Tomasz Hyziak