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
secretcallerid=Ben
host=dynamic
nat=no
mailbox=100@default
allow=all
context=outgoing
[mrblobby]
type=friend
user=mrblobby
secretcallerid=Blobby
host=dynamic
nat=no
mailbox=101@default
allow=all
context=outgoing
