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...t; ; Line 6 Display Name line6_authname: "UNPROVISIONED" ; Line 6 Registration Authentication line6_password: "UNPROVISIONED" ; Line 6 Registration Password # NAT/Firewall Traversal nat_address: "" voip_control_port: "5060" start_media_port: "16384" end_media_port: "32766" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "WaZaB-SIP" ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: EST # Phone prompt/password for telnet/console s...

...t; proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "10.6.0.223" outbound_proxy_port: "5060" proxy_register: 1 timer_register_expires : 120 # NAT/Firewall Traversal nat_enable: "1" nat_address: "" voip_control_port: "5060" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "1" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Garrett - " ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST sntp_ser...

Hi, I'm experimenting with the dev kit lite and now past the USB unpleasantness it's working great with standard phones and lines. The priority right now is getting soft phones (under Windows XP) working well. So far, I've only been able to get the XTEN Lite phone working and I really don't understand how I set it up. I used "xten" for every option everywhere (display

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

...t; # Backup Proxy info proxy_backup: "" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "" outbound_proxy_port: "" # NAT/Firewall Traversal nat_enable: "1" nat_address: "64.123.190.68" voip_control_port: "5060" start_media_port: "16000" end_media_port: "32768" nat_received_processing: "1" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Home " ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: CST # Telnet Lev...

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

...; proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "192.168.2.1" outbound_proxy_port: "5060" proxy_register: 1 timer_register_expires : 120 # NAT/Firewall Traversal nat_enable: "1" nat_address: "" voip_control_port: "5060" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "1" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Cisco - " ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: EST sntp_server...

...y info proxy_backup: "192.168.1.17" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "192.168.1.17" outbound_proxy_port: "5060" # NAT/Firewall Traversal nat_enable: "0" nat_address: "" voip_control_port: "5061" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable) proxy_register: "1" # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: "120" # Codec for m...

Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing "55 <phone icon with x>" so it looks like the phone is not registered. The phone and the asterisk are in the same local

...# Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable 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: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled outbound_proxy: "" ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060 ; default is...

...n Release 2.2 ###### # 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: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled # Outbound Proxy Support outbound_proxy: "" ; restricted to dotted IP or DNS A record o...

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47