The "error" I was talking about was in your log:
"...== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-6364'..."
The call terminated here in a error which prevented the dialplan from
continuing. Something there is broken, my recommendation is to check you
registrations first inside asterisk:
> sip show peers
Something wasn't "happy" about SIP/54 in your system when Asterisk tried
talking to it.
So you tried this:
"...
Even when I put:
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
exten => 4,n(line2),Dial(${FD_L2},20,trw)
exten => 4,n(line2),Voicemail(4)
..."
What that will do is go to the first instance of "4,n(line2)", which is the
line that seems to be triggering the channel failure. If you have the
Asterisk console open, I'll bet you see it spew some errors when you try
that extension routine.
Asterisk dial plans are a serial processes, the first line that Asterisk
comes across that meets the matching for a given extension and label is
what it will run first. What you have is two lines that will match both
extension and label - that's not really good form.
My dial plan suggestion from last night would result in the functionality:
Ring extension 4/Line_1, timeout 25 seconds --> if not busy then voicemail,
else ring extension 4/Line_2, timeout 20 seconds --> voicemail.
Again, I think you have two problems, and the bigger one is causing the
annoying unexpected behavior in your dial plan
Try doing the extension 4 without the Line_1 and see what happens:
"...
exten => 4,1,Dial(${FD_L2},20,trw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Hangup()
..."
I'll bet Line_2 never rings (which is indicative of the problem).
-Tim
On Mon, May 8, 2017 at 8:21 AM, <the...@sys-concept.com> wrote:
> Thank you for the input Tim.
> Yes, that worked.
>
> exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:vmail)
> exten => 4,n(vmail),Voicemail(4)
>
> Though, I'm not sure why are you saying line 2 is FD_L2 needs to be fixed.
> Do I need to removde "t", the call can not be transferred?
>
> Even when I put:
> exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
> exten => 4,n(line2),Dial(${FD_L2},20,trw)
> exten => 4,n(line2),Voicemail(4)
>
> The call (line2) would dial "FD_L2" but would not jump to next line
> "Voicemail"
>
>
> --
> Thelma
>
> On 05/08/2017 12:19 AM, Tim S wrote:
> > The way you have the GotoIf is making it so that no matter what the busy
> > condition of the line, it will execute the next line in the dial plan.
> > What you'd need is an "if" or "then" which goes to a tagged line in the
> > dial plan. How it reads now is: "If [busy] then line2, else execute
> > next line". Also you are saying "extension 4 is not busy", but
> > extension 4 is a dialplan extension - while physical extensions "FD_L1"
> > and "FD_L2" appear to be the devices which are not busy, you need to be
> > clear and keep it straight in your head and text to get the best help...
> >
> > According to your log, nobody picked up after the 25 second timeout on
> > FD_L1, so the dial status would have been NOANSWER, which would result
> > in your gotoif test having a FALSE. Since you didn't specify what the
> > gotoif should do if the busy test failed, it just executes the next line
> > which is to call the second line (FD_L2), which it does. Then it looks
> > like you have an error with the second line which causes the call to
> > terminate, at which case it terminates the channel and never gets to
> > voicemail.
> >
> >
> > So it looks like two problems, 1) your FD_L2 physical extension is
> > buggy, and 2) you need to label the voicemail entry point and jump to it
> > if the FD_L1 was any other state but BUSY.
> >
> >
> > "...
> > exten => 4,1,Dial(${FD_L1},25,trw)
> > exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:vmail)
> > exten => 4,n(line2),Dial(${FD_L2},20,trw); <--- fix me!!
> > exten => 4,n(vmail),Voicemail(4)
> > exten => 4,n,Hangup()
> > ..."
> >
> >
> > -Tim
> >
> >
> > On Sun, May 7, 2017 at 9:21 PM, <the...@sys-concept.com
> > <mailto:the...@sys-concept.com>> wrote:
> >
> > Call is not forwarded to voicemail in below dial plan, why?
> >
> > exten => 4,1,Dial(${FD_L1},25,trw)
> > exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
> > exten => 4,n(line2),Dial(${FD_L2},20,trw)
> > exten => 4,n,Voicemail(4)
> > exten => 4,n,Hangup()
> >
> > -- Called SIP/4
> > -- SIP/4-00000288 is ringing
> > -- Nobody picked up in 25000 ms
> > -- Executing [4@extensions:2] GotoIf("IAX2/home_server-6364",
> > "0?line2") in new stack
> > -- Executing [4@extensions:3] Dial("IAX2/home_server-6364",
> > "SIP/54,20,trw") in new stack
> > == Using SIP RTP CoS mark 5
> > -- Called SIP/54
> > -- SIP/54-00000289 is ringing
> > == Spawn extension (extensions, 4, 3) exited non-zero on
> > 'IAX2/home_server-6364'
> > -- Hungup 'IAX2/home_server-6364'
> >
> > Extension 4 is not BUSY (just nobody pickup the call) so why isn't
> > call going to "Voicemail" it shouldn't ring FD_L2 (SIP/54)
> > Why isn't it going to "Voicemail"?
> >
> > --
> > Thelma
> >
>
> --
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