I use GS 101 & 102, have a look at my configs at http://www.codepipe.com/id25.htm .
Hope they help. Dave -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen R. Besch Sent: 23 March 2004 20:22 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP --snip-- > I am having trouble setting the /etc/asterisk/sip.conf file. > This is my file: > 1) Add in the [general] section: disallow=all allow=ulaw allow=alaw allow=any other codec that you want to (or can) support. While some have found that this must be specified for each and every phone, I have found that it works fine specified just once in the general section. > [243075] > type = friend > context = default > secret = gol > host = dynamic > callerid = fono75 <243075> > 2) Include dtfmmode=info or inband and match to phone's setting 3) I may have been too tired at the time, but once I tried using long extensions (more than 5 digits) and could not make them work either - same error you are getting. I would limit your extensions to 4 digits and see if it helps. 4) You may also need to add canreinvite=no to each phone definition. > > and our SIP phones configuration are the following: > > SIP Server: 192.168.0.102 > > Outbound Proxy: <Empty> > 5) I would set this to be the same as the server if you want to make outbound calls. Hope this helps Stephen R. Besch _______________________________________________ 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