On Thu, Jun 24, 2010 at 8:26 AM, Scott Lawrence <[email protected]> wrote: > On 2010-06-24 7:59, Tony Graziano wrote: >> I am wondering if there is a way for sipx to "sense" a sip uri call >> and use a real sip uri callerid format. >> >> Right now if someone sends me a call to my sip address all I might get is >> >> Firstname Lastname >> 200 >> >> In this example I cannot call 200 back because it is not "reachable". >> If sip URI dialing is not available on my phone I can't redial it. >> > First of all, you need to be more specific about what you mean.... I > suspect that you're referring to what comes in the From header. That is > originally sent by the phone, and that's the best place to control it. > It's mostly a bad idea for a proxy to mess with To or From headers > (though the sipXecs caller-id feature can do so for calls routed out of > the system). > Yes, FROM.
BUT I was wondering if there was a way to tell sipx conditionally what type of callerid to send depending on destination. > >> I think it would be helpful is sipx could send a real callerID out, >> configurable by a superadmin for sipuri dialing, choosing the "method" >> of callerid to send (1,2,or3, see below). >> > The most important thing to understand is that what 'real' (by which I > assume you mean 'useful as a callback') depends on the system that the > recipient is using. That's why the caller-id feature can change things > when routing to gateways - it allows you to modify the From header so > that if the gateway conveys it correctly, it will have the callback > characteristics you want from the other side. No matter what you send, > though, there is the possibility that the routing configuration at the > recipient will be such that they can't use it. To maximize the chances > for SIP end-to-end calls, the most important thing you can do is to make > sure that your SIP routing in the DNS is correct (routes to you) no > matter where the recipient is calling from. >> 1. >> Firstname Lastname >> <telco_dialable_phone_number> >> >> or >> 2. >> Firstname Lastname >> <ISN> (1234*256) >> >> or >> 3. >> Firstname Lastname >> <[email protected]> >> Still leaves me wondering... I want to be able to send meaningful callerid so a "callback" feature will work on a phone. Is it possible to configure another gateway and diaplan just for sip uri dialing that can do a transform in the callerid at the gateway to send "u...@domain"? Am I on the right track here? I just don't see how to configure sip uri dialing in the dialplan in a way that makes this possible. Am I being more clear now? If not I will make another pot of coffee and consume before posting back on this. >> Does this make sense to ask for this as an improvement request? Or am >> I the only one getting calls and trying to figure out who that is this >> time? >> > > _______________________________________________ > sipx-dev mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev sipXecs IP PBX -- http://www.sipfoundry.org/
