[Asterisk-Users] SPA-3000 VOIP-PSTN gateway - long delay
betweenanswering and ringing
Anthony Rodgers
Anthony_Rodgers at dnv.org
Wed Feb 8 14:50:49 MST 2006
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
>
>
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