SIP problem on Attended Transfer

Asterisk Asterisk Support
1

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
context=IntPhones
username=1
secret=1234
amaflags=documentation
accountcode=11
subscribecontext=IntPhones
callerid=“phone 11” <11>
limitonpeers=yes
call-limit=100

2
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

2

In features.conf try increase the value of the transferdigittimeout and featuredigittimeout, this could help you.

Cheers.

Marco Bruni
www.marcobruni.net

3

Thanks but … I just set to this values

transferdigittimeout => 180 ; seconds
featuredigittimeout = 1500 ; milliseconds
atxfernoanswertimeout = 60 ; seconds

but there isn’t the problem!