[asterisk-users] Call does not go voicemail

Tim S tim.strommen at gmail.com
Mon May 8 01:19:43 CDT 2017

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()


On Sun, May 7, 2017 at 9:21 PM, <thelma at 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 at extensions:2] GotoIf("IAX2/home_server-6364",
> "0?line2") in new stack
>     -- Executing [4 at 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
> --
> _____________________________________________________________________
> -- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170507/817fc3cd/attachment.html>

More information about the asterisk-users mailing list