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

[Chad Wallace]

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()
>