thr3ads.net - asterisk users - [asterisk-users] call is not going to Voicemail with "1,n" [Dec 2010]

If this information is useful, please help other people find it:
Share via:

Joseph

2010-Dec-30 04:55 UTC

[asterisk-users] call is not going to Voicemail with "1,n"

I've tried to simplified the dial plan and use "n" instead of
numbers but I've noticed it is not executing my voicemail if I substitute
number with "n"

In the example below when the call is not answered, it does not go to voicemail;
call just hangup.

exten => 1,1,Playback(transfer)
exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,n,Voicemail(11,b)             ; Right to voicemail
exten => 1,n,Hangup()

Here is the transcript:

-- Executing [1 at office-open:1] Playback("SIP/pstn-5665-000000be",
"transfer") in new stack
-- <SIP/pstn-5665-000000be> Playing 'transfer' (language
'en')
-- Executing [1 at office-open:2] Dial("SIP/pstn-5665-000000be",
"SIP/11&IAX2/iaxy-322|20|jrw") in new stack
-- Called 11
-- Called iaxy-322
-- Call accepted by 10.0.0.108 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxy-322-8406 is busy
-- Hungup 'IAX2/iaxy-322-8406'
-- SIP/11-000000bf is ringing
-- Nobody picked up in 20000 ms
   == Auto fallthrough, channel 'SIP/pstn-5665-000000be' status is
'NOANSWER'


However, if I number the dial plan in the old fashion way and don't answer
the phone it goes to voicemail just fine:

exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,3,Voicemail(11,b)             ; Right to voicemail
exten => 1,4,Hangup()

-- 
Joseph

Chad Wallace

2010-Dec-30 05:12 UTC

head link

[asterisk-users] call is not going to Voicemail with "1,n"

On Wed, 29 Dec 2010 21:55:58 -0700
Joseph <syscon780 at gmail.com> wrote:
> I've tried to simplified the dial plan and use "n" instead of
numbers
> but I've noticed it is not executing my voicemail if I substitute
> number with "n"
> 
> In the example below when the call is not answered, it does not go to
> voicemail; call just hangup.
> 
> exten => 1,1,Playback(transfer)
> exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
> exten => 1,103,Voicemail(11,b)
> exten => 1,104,Hangup()
> exten => 1,n,Voicemail(11,b)             ; Right to voicemail
> exten => 1,n,Hangup()
I have a feeling you should put the 103 and 104 at the end, after all
your "n" lines for that extension. The "n"s are probably
coming up as
105 and 106, because they come after 104 in the file.

You could check the output of "dialplan show <context>" on the
Asterisk
console to verify this.


> Here is the transcript:
> 
> -- Executing [1 at office-open:1]
Playback("SIP/pstn-5665-000000be",
> "transfer") in new stack -- <SIP/pstn-5665-000000be>
Playing
> 'transfer' (language 'en') -- Executing [1 at
office-open:2]
> Dial("SIP/pstn-5665-000000be",
"SIP/11&IAX2/iaxy-322|20|jrw") in new
> stack -- Called 11 -- Called iaxy-322
> -- Call accepted by 10.0.0.108 (format ulaw)
> -- Format for call is ulaw
> -- IAX2/iaxy-322-8406 is busy
> -- Hungup 'IAX2/iaxy-322-8406'
> -- SIP/11-000000bf is ringing
> -- Nobody picked up in 20000 ms
>    == Auto fallthrough, channel 'SIP/pstn-5665-000000be' status is
> 'NOANSWER'
> 
> 
> However, if I number the dial plan in the old fashion way and don't
> answer the phone it goes to voicemail just fine:
> 
> exten => 1,1,Playback(transfer)
> exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
> exten => 1,103,Voicemail(11,b)
> exten => 1,104,Hangup()
> exten => 1,3,Voicemail(11,b)             ; Right to voicemail
> exten => 1,4,Hangup()
>

William Stillwell

2010-Dec-30 06:38 UTC

head link

[asterisk-users] call is not going to Voicemail with "1,n"

The n = prev + 1

So you dialplan technically looks like this:

exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,105,Voicemail(11,b)             ; Right to voicemail
exten => 1,106,Hangup()

which in the result, there is no 1,3 which goes auto fallthru.


Try this instead:

exten => 1,1,Playback(transfer)
exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,n,Voicemail(11,b)
exten => 1,n,Hangup()
exten => 1,n+101,Voicemail(11,b)             
exten => 1,n,Hangup()

which will result in a DP looking like this:

exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,3,Voicemail(11,b)
exten => 1,4,Hangup()
exten => 1,105,Voicemail(11,b)             
exten => 1,106,Hangup()


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joseph
Sent: Wednesday, December 29, 2010 11:56 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] call is not going to Voicemail with "1,n"

I've tried to simplified the dial plan and use "n" instead of
numbers but
I've noticed it is not executing my voicemail if I substitute number with
"n"

In the example below when the call is not answered, it does not go to
voicemail; call just hangup.

exten => 1,1,Playback(transfer)
exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,n,Voicemail(11,b)             ; Right to voicemail
exten => 1,n,Hangup()

Here is the transcript:

-- Executing [1 at office-open:1] Playback("SIP/pstn-5665-000000be",
"transfer") in new stack
-- <SIP/pstn-5665-000000be> Playing 'transfer' (language
'en')
-- Executing [1 at office-open:2] Dial("SIP/pstn-5665-000000be",
"SIP/11&IAX2/iaxy-322|20|jrw") in new stack
-- Called 11
-- Called iaxy-322
-- Call accepted by 10.0.0.108 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxy-322-8406 is busy
-- Hungup 'IAX2/iaxy-322-8406'
-- SIP/11-000000bf is ringing
-- Nobody picked up in 20000 ms
   == Auto fallthrough, channel 'SIP/pstn-5665-000000be' status is
'NOANSWER'


However, if I number the dial plan in the old fashion way and don't answer
the phone it goes to voicemail just fine:

exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,3,Voicemail(11,b)             ; Right to voicemail
exten => 1,4,Hangup()

-- 
Joseph

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

William Stillwell

2010-Dec-30 06:40 UTC

head link

[asterisk-users] call is not going to Voicemail with "1,n"

Also, a more "fancy" approach 

[macro-dialvm]

exten => s,1,NoOp("${ExTEN}|${MACRO_EXTEN}|${ARG1}")
exten => s,n,Dial(SIP/${ARG1},25,t)
exten => s,n,NoOp(${ARG1})
exten => s,n,NoOp(${DIALSTATUS})
exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?BUSY)
exten => s,n,GotoIf($["${DIALSTATUS}" =
"NOANSWER"]?NOANSWER)
exten => s,n,GotoIf($["${DIALSTATUS}" =
"CHANUNAVAIL"]?CHANUNAVAIL)
exten => s,n,VoiceMail(${ARG1},a)
exten => s,n,MacroExit()

exten => s,n(BUSY),Set(CDR(userfield)=DIAL-BUSY)
exten => s,n,NoO${MACRO_EXTEN})
exten => s,n,NoOp(${ARG1})
exten => s,n,ResetCDR(w)
exten => s,n,VoiceMail(${ARG1},b)
exten => s,n,MacroExit()

exten => s,n(NOANSWER),Set(CDR(userfield)=DIAL-NOANSWER)
exten => s,n,NoOp(${MACRO_EXTEN})
exten => s,n,NoOp(${ARG1})
exten => s,n,ResetCDR(w)
exten => s,n,VoiceMail(${ARG1},u)
exten => s,n,MacroExit()

exten => s,n(CHANUNAVAIL),Set(CDR(userfield)=DIAL-UNAVIL)
exten => s,n,NoOp(${MACRO_EXTEN})
exten => s,n,NoOp(${ARG1})
exten => s,n,ResetCDR(w)
exten => s,n,VoiceMail(${ARG1},uj)
exten => s,n,MacroExit()

exten => s,BUSY+101,Set(CDR(userfield)=DIAL-BSY-NOMBX)
exten => s,n,NoOp(${MACRO_EXTEN})
exten => s,n,NoOp(${ARG1})
exten => s,n,ResetCDR(w)
exten => s,n,NoOp("Mailbox Not found")
exten => s,n,Goto(CHANUNAVAIL+101)

exten => s,NOANSWER+101,Set(CDR(userfield)=DIAL-NA-NOMBX)
exten => s,n,NoOp(${MACRO_EXTEN})
exten => s,n,NoOp(${ARG1})
exten => s,n,NoOp("MailBox Not found")
exten => s,n,Goto(CHANUNAVAIL+101)

exten => s,CHANUNAVAIL+101,Playback(/home/asterisk/gen/themailbox)
exten => s,n,NoOp(${MACRO_EXTEN})
exten => s,n,NoOp(${ARG1})
exten => s,n,SayDigits(${MACRO_EXTEN})
exten => s,n,Playback(/home/asterisk/gen/doesnotexist)
exten => s,n,MacroExit()



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joseph
Sent: Wednesday, December 29, 2010 11:56 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] call is not going to Voicemail with "1,n"

I've tried to simplified the dial plan and use "n" instead of
numbers but
I've noticed it is not executing my voicemail if I substitute number with
"n"

In the example below when the call is not answered, it does not go to
voicemail; call just hangup.

exten => 1,1,Playback(transfer)
exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,n,Voicemail(11,b)             ; Right to voicemail
exten => 1,n,Hangup()

Here is the transcript:

-- Executing [1 at office-open:1] Playback("SIP/pstn-5665-000000be",
"transfer") in new stack
-- <SIP/pstn-5665-000000be> Playing 'transfer' (language
'en')
-- Executing [1 at office-open:2] Dial("SIP/pstn-5665-000000be",
"SIP/11&IAX2/iaxy-322|20|jrw") in new stack
-- Called 11
-- Called iaxy-322
-- Call accepted by 10.0.0.108 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxy-322-8406 is busy
-- Hungup 'IAX2/iaxy-322-8406'
-- SIP/11-000000bf is ringing
-- Nobody picked up in 20000 ms
   == Auto fallthrough, channel 'SIP/pstn-5665-000000be' status is
'NOANSWER'


However, if I number the dial plan in the old fashion way and don't answer
the phone it goes to voicemail just fine:

exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,3,Voicemail(11,b)             ; Right to voicemail
exten => 1,4,Hangup()

-- 
Joseph

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Joseph

2010-Dec-30 19:04 UTC

head link

[asterisk-users] call is not going to Voicemail with "1,n"

On 12/30/10 01:38, William Stillwell wrote:
>Try this instead:
>
>exten => 1,1,Playback(transfer)
>exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
>exten => 1,n,Voicemail(11,b)
>exten => 1,n,Hangup()
>exten => 1,n+101,Voicemail(11,b)
>exten => 1,n,Hangup()
>
>which will result in a DP looking like this:
>
>exten => 1,1,Playback(transfer)
>exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
>exten => 1,3,Voicemail(11,b)
>exten => 1,4,Hangup()
>exten => 1,105,Voicemail(11,b)
>exten => 1,106,Hangup()
I was analyzing it and I think in the above:
exten => 1,n+101,Voicemail(11,b)
is not needed at all as it does not take any effect. The (j) in the dial tell is
to jump but there is no 103 so it falls through to "3" which is
voicemail.
so actually "j" not needed nor are the:

exten => 1,n+101,Voicemail(11,b)
exten => 1,n,Hangup()

-- 
Joseph

Apparently Analagous Threads

Search for more possibly parallel threads

asterisk users - Dec 2010 - call is not going to Voicemail with "1,n"

[asterisk-users] call is not going to Voicemail with "1,n"

[asterisk-users] call is not going to Voicemail with "1,n"

[asterisk-users] call is not going to Voicemail with "1,n"

[asterisk-users] call is not going to Voicemail with "1,n"

[asterisk-users] call is not going to Voicemail with "1,n"

Apparently Analagous Threads