Re: [Asterisk-Users] Can't initiate a call with X-Lite.
> to which entry have to corespond "Domain/Realm" parameter in X-lite just put the same as your SIP Proxy, that is your Asterisk box address ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't initiate a call with X-Lite.
> No. You should work on configuring xlite to "register" with asterisk. > In the xlite Sip Proxy menu, you will need a "User Name", "Password", > "Sip Proxy", and "Domain/Realm" defined to match entries in your > sip.conf definitions. to which entry have to corespond "Domain/Realm" parameter in X-lite ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't initiate a call with X-Lite.
Rich Adamson wrote: It is just sending a sip invite to [EMAIL PROTECTED] Does the X-Lite need to connect to via a proxy? No. You should work on configuring xlite to "register" with asterisk. Thanks. I can get it to work that way. What I was trying to simulate was an external user calling in. Sorry, I should have stated that. From your asterisk CLI, try "sip debug" to see the flow of packets to/from asterisk; "sip no debug" will shut it off. That is just what I needed. I found that asterisk is looking in the 'default' context for the extension, whereas our extensions are under [from-sip]. I've got a little more configuring to do. Thanks. Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't initiate a call with X-Lite.
> I'm trying to place a call to asterisk using X-Lite. Asterisk is setup > with some Grandstream phones. I can call from one grandstream extension > to another. When I try to an extension with X-Lite, it comes back with > Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk > extension. It is just sending a sip invite to [EMAIL PROTECTED] Does the > X-Lite need to connect to via a proxy? No. You should work on configuring xlite to "register" with asterisk. In the xlite Sip Proxy menu, you will need a "User Name", "Password", "Sip Proxy", and "Domain/Realm" defined to match entries in your sip.conf definitions. Your sip.conf for xlite should look something like: [3005] type=friend host=dynamic username=3005 secret=yourpassword context=from-sip canreinvite=no mailbox=3005 > After several days of reading RFCs and looking at packet traces, I know > a bit more about SIP, but not quite enough to make this work. > > Is there a way to get asterisk to say what its doing? I tried > -vv etc, but the only messages are see are when I use one of my > my Grandstream phones. On the wire, is see the same "To:" header from > both the grandstream and the X-Lite soft phone. I don't understand why > its "found" by one, and not the other. >From your asterisk CLI, try "sip debug" to see the flow of packets to/from asterisk; "sip no debug" will shut it off. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't initiate a call with X-Lite.
Hello, I'm trying to place a call to asterisk using X-Lite. Asterisk is setup with some Grandstream phones. I can call from one grandstream extension to another. When I try to an extension with X-Lite, it comes back with Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk extension. It is just sending a sip invite to [EMAIL PROTECTED] Does the X-Lite need to connect to via a proxy? After several days of reading RFCs and looking at packet traces, I know a bit more about SIP, but not quite enough to make this work. Is there a way to get asterisk to say what its doing? I tried -vv etc, but the only messages are see are when I use one of my my Grandstream phones. On the wire, is see the same "To:" header from both the grandstream and the X-Lite soft phone. I don't understand why its "found" by one, and not the other. Thanks, Andy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
