On Tue, 2010-02-16 at 09:28 -0600, Ken Fulmer wrote: > We’ve had some problems with the Polycom phones use of SIP REFER > messages with Cisco and Adtran voice gateways. Call transfers and > holds fail due to incompatibilities in the way the Cisco and Adtran > routers handle these events. My understanding is the routers use a > mid-call INVITE rather than SIP REFER messages. This forces us to send > calls through the sipXbridge, like Asterisk, and that limits > scalability.
> We’ve noticed other phones in the list like Grandstream , Aastra, > Snom, etc. Do you guys know which phones in the sipX list work well > with Cisco and Adtran voice gateways? For a phone to work with sipXecs, it must use REFER to execute transfers. Similarly, sipXecs assumes that any PSTN gateway can handle REFER. (The exception being gateways that are treated as ITSPs, that is, which are accessed through sipXbridge.) If a gateway doesn't handle REFER correctly, it won't work natively with sipXecs, regardless of the phones that you're using. Tell the manufacturer that their product is deficient. Dale _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
