[Asterisk-Users] Troubleshooting Asterisk

[Michael Løjtnant]

On Mon, 20 Dec 2004 14:01:57 -0000
"Paul Brock" <Paul at westcomuk.com> 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 Løjtnant - Systems Engineer
ZyXEL Communications A/S
Columbusvej 5 - 2860 Søborg
Tel (+45) 3955 0700 - Fax (+45) 3955 0707