Tony, Have a look here http://www.codepipe.com/id25.htm these are my working examples.
I have 6 GS phones. The GS set-up's are from extersion 8002 onwards in sip.conf. Regards Dave -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: 06 March 2004 21:04 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Grandstream Budgetone SIP registration fails In article <[EMAIL PROTECTED]>, Jean-Marc V. Liotier <[EMAIL PROTECTED]> wrote: > Someone on the list certainly has a working setup with Asterisk and > Grandstream Budgetone phones, I would be grateful if their SIP > configuration was posted to the list. Quite unexpectedly I found no > complete example of such working setup on the Web, maybe because it was > so simple that no one thought that posting it would be useful to anyone. > One I get mine working I shall post the parameters ! Well my sip.conf looks like this: ---------------------------------------------------------------------------- -- ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=from-sip-external ; send unknown SIP callers to this context allow=ulaw allow=ilbc ; ; Tony's phone ; [2000] type=friend username=2000 secret=password host=dynamic context=from-sip-internal mailbox=2000 callerid=2000 dtmfmode=info ; ; Rachel's phone ; [2001] type=friend username=2001 secret=password host=dynamic context=from-sip-internal mailbox=2001 callerid=2001 dtmfmode=info ---------------------------------------------------------------------------- -- Then in the admin interface for Tony's phone I have the following: IP address: dynamic from DHCP SIP server: IP of Asterisk server Outbound proxy: empty SIP User ID: 2000 Authenticate ID: 2000 Auth password: password Vocoder choices (in order): PCMU, PCMA, then others .... SIP user ID is phone number: Yes SIP Registration: Yes Clear reg on reboot: No Reg expiration: 3 Early dial: No .... Local SIP port: 5060 Local RTP port: 5004 Use random port: No NAT Traversal: No .... Send DTMF: Via SIP INFO .... I think that's all the likely relevant ones. Hope this helps Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users