Hello, I'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work with the last version of Asterisk (Asterisk 13)? I have try the following configuration without success: 2 phones and Asterisk 13 are registered in an OpenSIPS Phone1 calls Phone2 Phone1 calls Asterisk 13 Phone1 transfers call in Asterisk 13 to Phone 2 But the transfer fails with an "NOTIFY 400 Bad Request". In Asterisk log I don't see any reference to "external_replaces" extension when the REFER arrives pjsip.conf [simpletrans] type=transport protocol=udp bind=0.0.0.0 [mytrunk] type=registration transport=simpletrans outbound_auth=mytrunk server_uri=sip:111 at 89.1.23.217:5060 client_uri=sip:111 at 89.1.23.217:5060 [mytrunk] type=auth auth_type=userpass password=111 username=111 [mytrunk] type=aor contact=sip:89.1.23.217:5060 [mytrunk] type=endpoint transport=simpletrans context=bucle-weasels disallow=all allow=ulaw outbound_auth=mytrunk aors=mytrunk [mytrunk] type=identify endpoint=mytrunk match=89.1.23.217 extensions.conf [transferencia] exten => external_replaces,1,NoOp() same => n,Dial(PJSIP/${SIPREFERTOHDR}@89.1.23.217) [bucle-weasels] exten => _.,1,Answer exten => _.,n,Wait(1) exten => _.,n,Set(TRANSFER_CONTEXT=transferencia) exten => _.,n,Playback(tt-weasels) exten => _.,n,Goto(2) exten => _.,n,Hangup Thank you in advance for your help David Pinedo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150130/7d94a74b/attachment.html>