search for: voip_control_port

I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both

...ser 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 reside in time_zone: EST # Phone p...

...o 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 messaging # Time Zone phone will...

...ind a NAT device on my LAN. The Asterisk 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...

...: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk - Cisco 7960 - NAT We have 10 Cisco 7960 phones at our office and a single static IP. Our asterisk server sits in the colo facility at our ISP. All phones are setup with a unique voip_control_port and they are all able to dial out. However, my phone is the only one that can receive a call. Every phone in the office can dial my extension and it will ring. I can call our main number and my phone will ring. But no other phone will ring! I get a fastbusy signal when trying to dial someone el...

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

...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 on SIP messaging # Time Zone phone will res...

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: "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 messaging # Time Zone phone will r...

...ort: "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] (Default - 3600) timer_register_expir...

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

...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_received_processing: 0 ; 0-Disabled (default), 1-Enabled outbound_proxy: "" ; restricted...

...; 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 media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled # Outbound Proxy Suppo...

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

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