Trey Hilyard
2016-Mar-18 15:20 UTC
[asterisk-users] Incoming INVITE with Portability Info and LRN
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;npdi at 12.4.240.200:5060;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. As a PBX, Asterisk doesn't have to worry about portability, but I am using it to simulate a full-blown Class 5 switch, so I have to have an LRN assigned to it to allow users to port to that switch. -Trey -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160318/03b3cd04/attachment.html>
Administrator TOOTAI
2016-Mar-18 15:49 UTC
[asterisk-users] Incoming INVITE with Portability Info and LRN
Le 18/03/2016 16:20, Trey Hilyard a ?crit :> 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;npdi at 12.4.240.200:5060;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 -- Daniel
Trey Hilyard
2016-Mar-18 15:58 UTC
[asterisk-users] Incoming INVITE with Portability Info and LRN
On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote:> Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > 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;npdi at 12.4.240.200 > :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_headerI 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 at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160318/c03fdb13/attachment.html>