[Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delaybetweenanswering and ringing

[Chris Stenton]

What have you set the

PSTN Dialing Delay:

on the PSTN Line tab (logged in as admin advanced) ?

Mine is set to 1 and it works well.

Chris

----- Original Message ----- 
From: "Anthony Rodgers" <Anthony_Rodgers at dnv.org>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Wednesday, February 08, 2006 9:50 PM
Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long 
delaybetweenanswering and ringing


> Hi Jean-Michel,
>
> We did actually try the 'r' option, but it has no effect, as Asterisk will 
> only supply ringing until the dialed SIP extension answers, which it does 
> immediately. The 4 second delay occurs between when the SPA-3000 answers 
> the SIP call and then places the PSTN one. I believe that the ringing tone 
> is provided by the PSTN at that point.
>
> Regards,
> -- 
> Anthony Rodgers
> Business Systems Analyst
> District of North Vancouver
> Web: http://www.dnv.org
> RSS Feed: http://www.dnv.org/rss.asp
>
>
> On Feb 8, 2006, at 1:41 PM, Jean-Michel Hiver wrote:
>
>> Anthony Rodgers a écrit :
>>
>> > Greetings,
>> >
>> > We are currently testing a Sipura SPA-3000 as a gateway from our
>> > Asterisk system to a PSTN line for 911 access. We have a number of
>> > locations and want to place an SPA-3000 in each, connected to a PSTN
>> > line that will provide the correct ANI/ALI information to 911 for
>> each
>> > location.
>> >
>> > It all works great, except for a reasonably significant (4 seconds)
>> > delay between when the SPA-3000 answers the SIP call from the
>> Asterisk
>> > server (immediately upon dialing, according to the Asterisk CLI) and
>> > the ringing tone begins (the remote phone begins ringing at that same
>> > time).
>> >
>> > The delay is enough for users to think that the phone isn't working -
>> > not what you want for 911!
>> >
>> > Any ideas?
>>
>> You could use the 'r' flag in your Dial() command to simulate a ringing
>> tone instantly. This is less than ideal though. Have you done some SIP
>> traces (using ngrep for examples) to look when the SIP 'ringing' signal
>> is actually being sent?
>>
>> Cheers,
>> Jean-Michel.
>>
>> -- 
>> Jean-Michel Hiver - http://ykoz.net/
>> Découvrez la Réunion des Technologies IP & Telecom
>> TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>