[asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls

[Raj Jain]

On Fri, Apr 25, 2008 at 5:15 PM, Andreas Sikkema <h323 at ramdyne.nl> wrote:

> On Apr 25, 2008, at 7:16 PM, Andreas Brodmann wrote:
> > What if once asterisk receives the command to initiate a call it
> > does so,
> > and when receiving a 484 it tells the other side anything like 484
> > (you're
> > on the right way but you are missing figures) and drops its newly
> > initiated
> > call. This continues until asterisk receives a 200 from the sip trunk.
>
> Hmm, I'd not wait for a 200OK, but maybe for a 180 RINGING or 18x
> Session Progress with SDP? There might even be a standard that
> describes this kind of dialling ;-)



RFC 3578 http://www.faqs.org/rfcs/rfc3578.html has some text on using 484
for mapping ISUP overlap dialing to SIP. I think this should work if the
484s received on the trunk are propagated over to the phone. This can
probably be done by setting the hangup cause code when tearing down the
phone leg of the call.

-- 
Raj Jain

mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20080425/5e5247db/attachment.htm