I'm using Asterisk 11 and have a problem with when making call transfer on remote Asterisk.
This dial plan below works when I make a call directly to remote asterisk dialing FXO on remote asterisk. exten => 4,1,Dial(${FD_L1},25,trw) exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2) exten => 4,n(line2),Dial(${FD_L2},20,rw) exten => 4,n(vmail),Voicemail(4) exten => 4,n,Voicemail(4) exten => 4,n,Hangup() Line one FD_L1 rings for 25sec. nobody pickup the line so FD_L2 rings for 20sec and if nobody pickup the line it goes to voice mail. However if I make a call over VPN to remote Asterisk, and dial exten: 4 FD_L1 rings for 25sec. and as soon as the call gets transferred to FD_L2 it gets abruptly terminated with a message: exited non-zero on 'IAX2 -- SIP/54-00000006 is ringing == Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-424' Does it have something to do with "transfer=no" in iax.conf? -- Thelma -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users