I have it set the same for each phone within the sip.conf file if this is where you meant.
FYI, I am using Grandstream 101 phones on both ends. Should it be set to yes for these phones? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Tuesday, October 19, 2004 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to ring internal extension? at I have in my extensions.conf file > below minus my auth info :) > > What happens now is this, if I pickup the sip phone at ext. 100 and > dial extension 101 the phone at 101 rings but when 101 answers we > can't talk between the phones it's silence. Check: Canreinvite=$value Codecs are the same on both phones _______________________________________________ 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