[asterisk-users] Connection problem w/ Attended Transfer

[Ben Hall]

Hi all,
I'm new posting here, though not to perusing. I'm having an issue  
with attended transfer and was wondering if anyone had heard of the  
problem/had any suggestions... Apologies in advance if this post is  
excessively newb-oid.

- An incoming call C is passed to A, a POTS telephone connected via a  
Handytone 286 ATA.
- A presses atxfer key, then dials B, a Win XP laptop running x-lite.
- A and B talk and A hangs up to transfer C to B.
- Most audio between B and C is lost, for the small proportion that  
does get through, latency is very high.
- When B and C hang up, asterisk sometimes 'crashes' - incoming calls  
are rejected and the CLI becomes unresponsive to commands.

Asterisk version is 1.2.14.
An example of the cli output with max verbosity is at http:// 
nyodrinkers.com/cliout.txt

I know there have been problems with call transfers & the Handytone  
line, I recently updated the firmware which fixed blind transfer and  
attended transfer at least now works in theory... If anyone can help  
I'd be massively grateful!
Best wishes,
Ben Hall

extensions.conf:
[voiptalkincoming]
exten => 01225808102,1,Answer
exten => 01225808102,2,Dial(SIP/reception,10,t)			; at this point  
'reception' [ie A] dials 100

exten => 100,1,Dial(SIP/mrblobby,10,t)						; the quality of the  
transferred call between mrblobby and
exten => 100,2,Hangup									; voiptalk [ie B and C] is extremely poor


sip.conf
[general]
jbenable = yes
jbmaxsize = 1000
jbresyncthreshold = 1000

[reception]
type=friend
user=reception
secret=
callerid=Ben
host=dynamic
nat=no
mailbox=100 at default
allow=all
context=outgoing

[mrblobby]
type=friend
user=mrblobby
secret=
callerid=Blobby
host=dynamic
nat=no
mailbox=101 at default
allow=all
context=outgoing