[asterisk-users] Problems on default Attended Transfer

[derwditel derwditel]

Hi,
I'm currently using Asterisk 1.4.23.1, and I have a problem (also on
previous version).
Sometimes, when I try to do an attended transfer to another internal with
default feature *2, Asterisk doesn't make it (it doesn't play
'pbx-transfer'). Sometimes on second time, Asterisk make transfer correctly.
I have this problem on variuos type of SIP phones (GrandStream, Aastra,
OKI).

My sip.conf is like the following account:

=======================================
[intphones](!)
type=friend
qualify=yes
host=dynamic
callgroup=1
pickupgroup=1
dtmfmode=sip

[1](intphones)
context=IntPhones
username=1
secret=1234
amaflags=documentation
accountcode=11
subscribecontext=IntPhones
callerid="phone 11" <11>
limitonpeers=yes
call-limit=100

[2](intphones)
context=IntPhones
username=2
secret=1234
amaflags=documentation
accountcode=12
subscribecontext=IntPhones
callerid="phone 12" <12>
limitonpeers=yes
call-limit=100
=======================================

and on extensions.conf my dial lines are like:

=======================================
exten => _1X,1,Dial(SIP/${EXTEN:1},,tTr)
exten => _1X,n,Hangup()
=======================================



Can anyone help me? I don't underwstand where I make the mistake!

Thanks to everyone

Marco
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090316/ec68b9e6/attachment.htm