[asterisk-users] problem transferring calls some of the times

[Raúl Gómez C.]

Ian,

I'm unable to transfer calls using *2, I'm not sure why. Here's my configs:

*sip.conf*

[User1]
type=friend
username=111
context=default
callerid=User Name <111>
host=10.10.1.111
nat=no
canreinvite=no
dtmfmode=info
call-limit=4
mailbox=111 at default
disallow=all
allow=ulaw
allow=alaw
;allow=gsm
;callgroup=1,3-4
;pickupgroup=1,3-5
;callingpres=allowed_passed_screen



*features.conf:*

[general]
transferdigittimeout => 3
xfersound = beep
xferfailsound = beeperr
pickupexten = *8
featuredigittimeout = 500
atxfernoanswertimeout = 15

[featuremap]
blindxfer => #2
disconnect => *0
;automon => *1
atxfer => *2
;parkcall => #72


In the phones the "*Send DTMF:"* is set to "in-audio" and "via SIP INFO"

What I'm missing here??

-- 
Raul
Linux Counter #156439
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