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