[asterisk-users] Attended transfer and 'pbx-invalid' - 1.4.26

Gabriel Ortiz Lour ortiz.admin at gmail.com
Wed Jul 22 12:44:42 CDT 2009


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
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