Quoting myself ;) > Hi, > > I've managed to get X-Lite (v2 build 1050) working pretty well with *, but > am having problems with the DTMF signalling.
In case anyone is interested, the settings I use to get X-Lite build 1050 to talk to * are: System Settings > SIP Proxy --------------------------- User Name: jamie (my sip username in *) Password: mypassword (my sip password in *) Domain/Realm: asterisk SIP Proxy: asterisk.versado.net (hostname or IP address of asterisk server) Send Internal IP: on Anything not listed should be left as default. This works when there are no intermediate NAT devices. If the client is behind a NAT box then make these changes to the above: System Settings > Network ------------------------- NAT Firewall IP: 1.2.3.4 (public IP address of the NAT box) System Settings > SIP Proxy --------------------------- Send Internal IP: off Hope that is of some use to someone. Jamie > > I've used inband signalling with no problems on the uncompressed codecs > (G711), but obviously this doesn't work with the compressed ones (GSM). > > However when I try to use RFC 2833 it doesn't seem to pick up "0" > properly. > For example if I dial the voicemail app and then enter my > extension (600) it > says "Login incorrect" even though I haven't been prompted to enter my > password, and the console shows the dialed number as 6 or 60. > > Has anyone else got this to work? > > Jamie Neil > Versado I.T. Services Ltd. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
