Re: [asterisk-users] Call does not go voicemail
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, 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/4-0288 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/54-0289 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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> Check out the new Asterisk community forum at: https://community.asterisk.
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>
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[asterisk-users] Call does not go voicemail
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/4-0288 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/54-0289 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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Re: [asterisk-users] CM for menuselect choices
> Use menuselect's command line (--enable and --disable). Great idea! How would you recommend generating the set of --enable and --disable options that differ from the default from a build that was done? -- _ -- 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
Re: [asterisk-users] CM for menuselect choices
On Fri, May 05, 2017 at 11:21:20AM -0400, Richard Kenner wrote: > I'd like to be able to save the choices made in menuselect in a way > that they can be tracked in a CM system and applied to a later release > of Asterisk using an automated tool like Ansible. What's the best > way to do that? Use menuselect's command line (--enable and --disable). Note that this requires an extra build stage: $MAKE menuselect.makeopts ./menuselect/menuselect \ --enable foo \ --disable bar \ # Alternatively, patch the sources of Asterisk to have foo enabled and bar disabled. This should be simple if you maintain your own stack of patches anyway. Examples: --- a/addons/res_config_mysql.c +++ b/addons/res_config_mysql.c @@ -24,7 +24,6 @@ /*** MODULEINFO mysqlclient - no extended ***/ and: --- a/sounds/sounds.xml +++ b/sounds/sounds.xml @@ -10,7 +10,6 @@ core - yes core @@ -246,7 +245,6 @@ - yes core You can also add 'defaultenabled' to set the default, if needed. menuselect.makeopts is not a file to keep as it is a generated file that is overly verbose and breaks all too often (on a change of version. And also potentially on a change of configure options?). I tried in the past to replace menuselect. In my replacement it had a simple configuration file (build_tools/conf) that cuild be easily hand edited. See menuselect/contrib/menuselect-dummy . However, it takes effort to keep it up-to-date with menuselect, and I never bothered for quite some time. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- 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
