Gabriel Ortiz Lour
2009-Jul-22 17:44 UTC
[asterisk-users] Attended transfer and 'pbx-invalid' - 1.4.26
Hi, I've created a tiny dialplan to test the return of a call on transfers, like this: (I had to use the DEVSTATE backport here) [phones] exten => _12XX,1,Dial(SIP/${EXTEN},6,tT) exten => _12XX,n,GotoIf($[ "x${BLINDTRANSFER}" = "x" ]?noBT) exten => _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)}); exten => _12XX,n,Goto(dRet) exten => _12XX,n(noBT),GotoIf($[ "x${TRANSFERERNAME}" = "x" ]?sai) exten => _12XX,n,Set(DIALRET=${CUT(TRANSFERERNAME,-,1)}); exten => _12XX,n,GotoIf($[ "${DEVSTATE(${DIALRET})}" = "INUSE" ]?sai); exten => _12XX,n(dRet),Set(CALLERID(all)=RET_${EXTEN} <${CALLERID(num)}>) exten => _12XX,n,Dial(${DIALRET},,mTt) exten => _12XX,n(sai),Hangup() It all works like a charm, except that when I do an atxfer and dial another SIP and it rings, but dont answer, asterisk plays the 'pbx-invalid' sound, that is a bit confusing, because the phone is there and actually rang . Here is the CLI output *CLI> -- Executing [1201 at irrestrito-user:1] Dial("SIP/1202-08330f80", "SIP/1201|6|tT") in new stack -- Called 1201 -- SIP/1201-08335530 is ringing -- SIP/1201-08335530 answered SIP/1202-08330f80 -- Started music on hold, class 'default', on SIP/1202-08330f80 -- <SIP/1201-08335530> Playing 'pbx-transfer' (language 'en') -- Executing [1203 at irrestrito-user:1] Dial("Local/1203 at irrestrito-user-70b2,2", "SIP/1203|6|tT") in new stack -- Called 1203 -- SIP/1203-08325260 is ringing -- Local/1203 at irrestrito-user-70b2,1 is ringing>>>>>>>>>> Ring and no answer...-- Nobody picked up in 6000 ms -- Executing [1203 at irrestrito-user:2] GotoIf("Local/1203 at irrestrito-user-70b2,2", "1?noBT") in new stack -- Goto (irrestrito-user,1203,5) -- Executing [1203 at irrestrito-user:5] GotoIf("Local/1203 at irrestrito-user-70b2,2", "0?sai") in new stack -- Executing [1203 at irrestrito-user:6] Set("Local/1203 at irrestrito-user-70b2,2", "DIALRET=SIP/1201") in new stack -- Executing [1203 at irrestrito-user:7] GotoIf("Local/1203 at irrestrito-user-70b2,2", "1?sai") in new stack -- Goto (irrestrito-user,1203,10) -- Executing [1203 at irrestrito-user:10] Hangup("Local/1203 at irrestrito-user-70b2,2", "") in new stack == Spawn extension (irrestrito-user, 1203, 10) exited non-zero on 'Local/1203 at irrestrito-user-70b2,2' -- Stopped music on hold on SIP/1202-08330f80>>? -- <SIP/1201-08335530> Playing 'pbx-invalid' (language 'en')am I doing something wrong? Thanks, Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090722/52672d6f/attachment.htm