On Mon, 20 Dec 2004 14:01:57 -0000 "Paul Brock" <[EMAIL PROTECTED]> wrote:
> > >Perhaps you could turn off the Cisco's, restart * and Turn back on the > >Cisco's, and send the CLI output. > > Hmm.. again.. nothing...however, since I'm using static Ip's for the phones, > so I guess that they would exist in * without registration. Perhaps... i'm using DHCP so I can't say. To really get some output, issue this in the CLI, and make a call between the phones. sip debug ip 192.168.1.151 The contens of your SIPDefaults.cnf looks fine, I just looked through mine and took out some parts might help. Try adding this to your SIPDefaults.cnf: # Proxy Server proxy1_address: "192.168.1.150" ; Can be dotted IP or FQDN # Proxy Server Port (default - 5060) proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # NAT/Firewall Traversal nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 10000 ; Start RTP range for media (default - 16384) end_media_port: 20000 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled -- Med venlig hilsen / Best regards Michael L�jtnant - Systems Engineer ZyXEL Communications A/S Columbusvej 5 - 2860 S�borg Tel (+45) 3955 0700 - Fax (+45) 3955 0707 _______________________________________________ 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
