I think the invite is what should make the difference. I don't normally prepend or strip digits to interconnect two systems for migration purposes when I can help it. Have you tried changing the sipx dial plan to not prepend digits?
i.e. - 4200-4350 are on nortel, so custom dialplan say 42 plus 2 digits and 435 plus 1 digits, send all digits to patton. in the patton say 42.. and 435. matches got to the pri where the nortel is located. I think if I recall you have a point about the refer not matching the invite. I agree that should be stripped at the proxy before sending the call, not just at the invite. In my instance, I don't think I can create a custom dialplan and do similar tests (albeit with only polycom handsets) until Monday at earliest. It sound like you are in a transitional phase, so keeping the dialing as "native as possible" is most desirable, so perhaps the number matching without adding or stripping digits might be less problematic. On Thu, Sep 1, 2011 at 7:28 PM, Steve Beaudry <[email protected]>wrote: > > Content-Type: text/plain; > charset="utf-8" > Content-Transfer-Encoding: 8bit > Organization: SipXecs Forum > In-Reply-To: < > camgknjwqzrf18a5kxnopamq_iwgxokb_t0lwr82xv+aoe6b...@mail.gmail.com> > X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <63028> > Message-ID: <[email protected]> > > > > To answer your most recent question first, All 4 digit > calls match a custom dialplan, I've named 'Nortel Locals'. > That dialplan entry is very simple. It simply prepends '99' > to the dialed number, and passes it to the gateway. The > gateway is the Patton smartnode 4960, at IP Address > 10.60.0.10. > > Also, I understand that the Patton folks could be wrong, but > the logic makes sense. It seems to me that you're > suggesting that the Patton gateway should simply ignore the > domain suffix of the URI in the 'REFER-TO' packet, and > forward the call based on the numeric portion of the URI, > compared against it's dialplan/route table. I would suggest > that even SipXecs itself wouldn't behave this way. If > SipXecs is configured for the domain 'voip.royalroads.ca', > and I sent it a request to initiate or transfer a call to > mailto:'[email protected]', it would not send the > call to the '9999' extension, even if there were a dialplan > entry matching '9999' exactly. > > The following are two examples of call flow, and where they > get transformed. The first example is a direct dialled > call, which completes successfully. The second is the same > call flow, but transferred, instead of dialled directly. > > Direct dialling: > The User dials '4495'. > > The Grandstream phone (and every other endpoint) transforms > this to an 'INVITE:mailto:[email protected]' (the > registered domain of the endpoint), and sends it to it's > proxy, the SipXecs server. > > The SipXecs server transforms the number according to it's > dialplan to mailto:'[email protected]', and sends it to the > Smartnode gateway (also according to the dialplan). > > The Smartnode 4960 has a routing table (which is ONLY > evaluated > if the domain is the IP/Hostname of the smartnode itself) > with a routing entry for any 6-digit destination that starts > with '99' (ie. 994495) should be sent out to the Nortel via > PRI. > > > Transferring: > Someone calls the user. Everything works as it should. > > The user transfers the call to '4495'. > > The Grandstream phone (and every other endpoing) transforms > this to a 'REFER-TO:mailto:[email protected]' (again, > because it's the registered domain of the endpoint), and > sends it to it's proxy, the SipXecs server. > > The SipXecs authenticates the REFER-TO request (as it > should, per past bug tickets), then forwards it, unmodified, > as 'REFER-TO:mailto:[email protected]' to the > originating endpoint (which in the cases I've been > troubleshooting are the Patton smartnode 4960 as well, but > it could be any endpoint). > > The Patton gateway, recieving a > 'REFER-TO:mailto:[email protected]', sends and > 'INVITE:mailto:[email protected]' back to the SipXecs > server (which is registered for the 'voip.royalroads.ca' > domain. > > > Cheers, > > ...Steve... > > PS - I've affectionately renamed my iPhone spell checker to > 'Auto-incorrect'. :) > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet faxservices!
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