As other have reported before, it's not straight forward to get kphone to properly register at an Asterisk server. http://lists.digium.com/pipermail/asterisk-users/2003-March/009441.html
I've traced this a bit and have come to the following conclusion: # ngrep REGISTER udp U 193.80.224.82:32816 -> 193.171.3.9:5060 REGISTER sip:193.171.3.9 SIP/2.0 Via: SIP/2.0/UDP 193.80.224.82 U 193.171.3.9:5060 -> 193.80.224.82:32816 SIP/2.0 407 Proxy Authentication Required.. Via: SIP/2.0/UDP 193.80.224.82;received=193.80.224.82 kphone is sending from port 32816. Looking with netstat and lsof shows that it expects its answers back at port 5060. Asterisk's interpretation of the Via header (w/o port) is to reply to the src port. kphone seems to interpret the Via header w/o port as to mean port 5060 for answers. ------------- Then I tested kphone -p 10000, thus moving kphone from the default port. Result: U 193.80.224.82:32816 -> 193.171.3.9:5060 REGISTER sip:193.171.3.9 SIP/2.0 Via: SIP/2.0/UDP 193.80.224.82:10000 U 193.171.3.9:5060 -> 193.80.224.82:32816 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 193.80.224.82:10000 So: Asterisk is still sending the answers to the src port (32816) instead of the port in the Via header. ------------- A quick read of the SIP RFC gave me no fast answer on what the correct behaviour is (only that I tend to agree with Mark that SIP is a mess). Is anybody here firm enough on SIP to give a definite answer on which software needs patching? /ol -- < Otmar Lendl ([EMAIL PROTECTED]) | nic.at Systems Engineer > _______________________________________________ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev
