[asterisk-users] Asterisk not working with Festival

[Mark G. Thomas]

Hi,

I'm experiencing the same problem, with identical symptoms.

I also noticed that after making a call attempt, I see this stuck TCP
connection pair until I stop and restart the asterisk server process.

# netstat -an | grep 1314
tcp        0      0 0.0.0.0:1314                0.0.0.0:*                   LISTEN      
tcp       46      0 127.0.0.1:52206             127.0.0.1:1314              CLOSE_WAIT  
tcp        0      0 127.0.0.1:1314              127.0.0.1:52206             FIN_WAIT2   

Mark

On Thu, Aug 12, 2010 at 02:41:50PM +0530, Davinder Kumar Meen wrote:
>    I tried it but I still cannot hear any sound created from Festival()
>    function. I can hear only a voice saying one which was working earlier
>    as well. Here is log of asterisk console:
>       -- Attempting call on SIP/011xxxxxxxxxxxxxxxxx at gafachi1a for
>    s at connect-to-me:1 (Retry 1)
>        -- Executing [s at connect-to-me:1] Answer("SIP/gafachi1a-00000000",
>    "") in new stack
>        -- Executing [s at connect-to-me:2] Wait("SIP/gafachi1a-00000000",
>    "7") in new stack
>        -- Executing [s at connect-to-me:3]
>    SayDigits("SIP/gafachi1a-00000000", "'1'") in new stack
>        -- <SIP/gafachi1a-00000000> Playing 'digits/1.slin' (language 'en')
>        -- Executing [s at connect-to-me:4] Festival("SIP/gafachi1a-00000000",
>    "hello john") in new stack
>      == Parsing '/usr/local/etc/asterisk/festival.conf':   == Found
>    On 11/08/10 11:22 PM, "Danny Nicholas" <danny at debsinc.com> wrote:
>      ____________________________________________________________________
> 
>      From: asterisk-users-bounces at lists.digium.com
>      [[1]mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>      Davinder Kumar Meen
>      Subject: Re: [asterisk-users] Asterisk not working with Festival
>      Can anyone help please on this?
>      <snip>
>      >[connect-to-me]
>      >exten => s,1,Answer
>      >Exten => s,n,SayDigits(`1')
>      >exten => s,n,Festival(hello john)
>      >exten => s,n,Hangup
>      <snip>
>      When you call in from your mobile, you are using a DAHDI channel
>      which introduces a 3-7 second delay into the process, unless you
>      have one of the "blessed" phone companies that offers call
>      supervision.  If you put a wait(7) in front of SayDigits, you should
>      hear the call "normally".
>      This is what I would suggest
>      [connect-to-me]
>      exten => s,1,Answer
>      Exten => s,n,Gotoif($["${EXTEN}:0:3)" = "SIP"]?4:3
>      Exten => s,n,wait(7)
>      Exten => s,n,SayDigits(`1')
>      exten => s,n,Festival(hello john)
>      exten => s,n,Hangup
> 
>    Thanks,
>    Davinder Kumar Meen
>    Partner & Project Manager
>    Impinge Solutions, F-250, Phase 8B, Mohali (India)
>    www.impingesolutions.com
>    We also provide server hosting services. Please checkout our website
>    www.goforspace.com
> 
> References
> 
>    1. mailto:asterisk-users-bounces at lists.digium.com]

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-- 
Mark G. Thomas (Mark at Misty.com)
Web: http://mgtinternet.com/
Tel: +1-215-512-0112 US: 877-512-0112