This is my first time configuring sipx with an ITSP. I am sorry it is a newbie question, but I don't see Speakeasy in the tested ITSP list. I'd like to ensure that I configure and test it right (along the lines of a test case spreadsheet I found on the sipx wiki) before I send my config and results to the wiki.
All I did was add a gateway and change just three items in the sipx GUI page for SIP trunk ITSP Account page: 1. User name: <username> 2. Password: <secret> 3. User public address for call setup: Off [** Our netscreen SSG's SIP ALG is supposed to rewrite addresses, do firewall pinholes, etc.**] Did a quick test of outgoing and incoming calls. Seems to work fine. But Speakeasy gave me a whole bunch of parameters when they provisioned the trunks. Here is what they gave (which I think is for asterisk): canreinvite=yes context=from-pstn dtmfmode=RFC2833 insecure=very qualify=no host=<XXX>.speakeasy.net username=<XXX> secret=<XXX> type=peer Registration String: <username>:<secret>@<host>/<username> What I am supposed to do with all the other stuff Speakeasy gave me? Just ignore it? Thanks _______________________________________________ sipx-users mailing list [email protected] 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/
