search for: nat_address

...ave "NAT traversal: no" and leave "Use NAT IP" blank. All the clever stuff is done automatically by Asterisk. However, with the Cisco, that doesn't seem to be the case. I have found it necessary in the SIPDefault.cnf file to set "nat_enable: 1" and then specify as nat_address the public address of my router. Is this normal? What is different between the Grandstream and the Cisco? Is there any way to avoid having to program the external address into the Cisco when it is behind NAT? Thanks in advance for any advice. Cheers Tony -- Tony Mountifield Work: tony at softin...

...; Line 6 Extension\User ID line6_displayname: "" ; 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 resi...

..." # Backup Proxy info proxy_backup: "10.6.0.223" 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 m...

...sk server is hosted offsite and has a public IP address. I've set up port-forwarding on the firewall for both phones to tunnel the SIP messages initiated by the Asterisk box. It works like a charm with the Cisco phone by using the following config info: voip_control_port: 5077 nat_enable: 1 nat_address: "" nat_received_processing: 0 Every time the Cisco phone registers with Asterisk, it does so using port 5077 and with the corresponding port-forwarding rule added to the firewall, it works great. However, for the life of me, I can't get the Aastra to do the same thing. I thought th...

...expires: 3600 dial_template: dialplan messages_uri: 3688 telnet_level: 2 phone_label: "Nat One" line1_name: 3115552368 line1_shortname: 3115552368 line1_authname: 3115552368 line1_password: 3115552368 line1_displayname: "Nat One" logo_url: "http://192.168.1.45/pts.bmp" nat_address: 64.169.xx.xxx nat_enable: 1 nat_received_processing: 1 proxy1_address: myserver.outthere.com My NatTwo phone is similar. The only difference is the name/password using 8115552368. ----sip.conf------------- [3115552368] type=friend host=dynamic username=3115552368 secret=3115552368 nat=1 context...

...clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] Thank you. -- James (Please CC me on all replies)

...ion: P0S3-08-9-00 proxy1_address: neocipher.net ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: "100" phone_password: "cisco" ; Limited to 31 characters (Default - cisco) sntp_server: 10.10.10.1 time_zone: EST dial_template: DIALPLAN nat_enable: 1 nat_address: 172.16.2.1 nat_received_processing: 1 outbound_proxy_port: 5060 outbond_proxy: ns1.neocipher.net SIP0112B9EAFF72.cnf image_version: P0S3-08-9-00 # Line 1 Setup line1_name: 101 line1_authname: 101 line1_shortname: "Line 101" line1_password: "test" line1_displayname: "Ste...

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

...oxy info proxy_emergency: "" proxy_emergency_port: "5060" # 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...

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

...uot; # Backup Proxy info proxy_backup: "192.168.2.1" 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 mes...

...17" proxy_emergency_port: "5060" # Backup Proxy 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] (Defa...

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

...kup Proxy port (default is 5060) proxy_emergency: "" ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # 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_receiv...

...ency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable ####### New Parameters added in 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 me...

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

The 3.1 Beta is now available -- check the Shorewall home page. -- Tom Eastep \ Shorewall - iptables made easy AIM: tmeastep \ http://www.shorewall.net ICQ: #60745924 \ teastep@shorewall.net

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