On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <[email protected]> wrote:
> Le 18/03/2016 16:20, Trey Hilyard a écrit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from > > the INVITE as the extension in the dialplan. > > > > The INVITE R-URI looks like: > > INVITE > > sip:+19135041291;rn=+19136630000;[email protected] > :5060;user=phone;transport=udp > > SIP/2.0 > > > > The +1913663000 is the LRN of the Asterisk box, so I would want to have > > the dialplan validate that the "rn" is that number. The +19136631291 is > > the extension within the system that they are trying to reach, that > > extension will vary, and will have an exten defined in the dialplan. > > > > I assume that this is just going to require that I do some matching and > > substring-type variable replacement to hit a context with just the > > Called Number part of the request, but I wondered if anyone had a > > working example of this before I started putting too much effort into it. > > Use the SIP_HEADER function > > http://www.voip-info.org/wiki/view/Asterisk+func+sip_header I am not sure that this is needed here. The Request URI has all of the values that I need. I agree that I might need to CUT part of the R-URI, but I don't need access to any other header to find the info I need. When the call arrives at the Asterisk right now, this is the exten/context that it is hitting, so it already has the info I need: Executing [9135041291;rn=+19136630000;npdi@from_pstn:1] As far as I can tell, I think that I just need to figure out how to make an extension entry that matches on the "rn=+19136630000\;npdi" and then moves to another context (or same one) with ${EXTEN,0,10}. I just can't get that first extension to match on the RN value. > > > -- > Daniel > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
