Sip register port - Aastra Telecom Aastra 57i CT

I'm not sure if you're aware of something called a "SIP scanner" but this could be your problem. I suggest you Google it to gain further understanding which I won't go into here. The best method is for you to go into you VoIP configuration manager and change the SIP Port to something other than 5060. This is the most common port used by SIP scanners (which are normally friendly but can be used to exploit). Don't panic, though. Just change the port and see if that helps.

Hi,
Yes it supports.

SIP Account Name: Vonage
Phone Number: xxxxxxxxxx (enter your Vonage phone number)
Authentication ID: xxxxxxxxxx (enter your Vonage phone number)
Authentication Password: xxxxxxxx (enter your Vonage Authentication Password, this can be Provided to you by Vonage. Please remember that the password is case sensitive.)
SIP Domain: leave blank
Proxy Address: sphone.vopr.vonage.net
Port: 5061
Outbound Proxy: sphone.vopr.vonage.net
Port: 5061
Local Port: 5061
Register Timer: 3600
Codec: Auto
Pkt. Time: Default
Out of Band DTMF: On

-Karthik.

SIP (Session Initiation Protocol) is a protocol used to establish, modify and terminate VOIP telephone calls. Using SIP, you can make cheap calls to a landline and regular mobile phones. With Nimbuzz SIP support, you can make Internet calls via the Nimbuzz Phonebook feature by adding any SIP provider you already use. Costs depend on your SIP provider rates.
SIP calling is mostly used to make cheap international calls.

You can go to Options > Settings > SIP Accounts to register an account.
After that you can place VoIP calls via the Phonebook.

User Fring if you want to make free calls. It works on Yahoo Messenger and Gtalk so you can call your buddies available on Yahoo! or Gtalk.

Confirmed modelLinksys IP Phone SPA941 and SPA962 with SPA932
Linksys IP PhoneThis document explains how to use Brekeke SIP Server with the Linksys IP Phone SPA hard phone .
The Standard features include two active line appearances, call forwarding, redialing, speed dialing, transferring calls, conference calling and accessing voice mail. For more information regarding this product please go to www.linksys.com
Configure Linksys PhoneSet up your Linksys phone with a fixed IP address (e.g. 192.168.0.15) and subnet mask (e.g. 255.255.255.0).

Once the IP address is configured you can access phone webpage by its IP from web browser (e.g. http://192.168.0.15). Network SetupClick on "Admin Login", then select “System" tab. Sign Static IP, Gateway and NetMask.

Use the following data as an example

• Static IP: Linksys phone IP (e.g. 192.168.0.17)
• Netmask: 255.255.255.0
• Gateway: not needed if SIP Server is in the same LAN as Linksys phone
• Click on [Submit all changes] button now to reboot you phone,
  or after whole setup is finished

Phone ConfigurationSelect tab Ext1 or Ext2 and enter SIP server information and sign extension number

Use the following as an example to complete the phone configuration.

• Proxy and Registration:
  
  • Proxy: Brekeke SIP Server’s IP address (e.g. 192.168.0.69)
    
  • Register: Yes
• Subscriber Information:
  
  • Display Name: extension number or Name (e.g. 400 or Alice)
  • User ID: extension number (e.g. 400)
  • If user authentication is set on at Brekeke SIP Server,
    type in the same information at Auth ID and Password as what you set at SIP Server side
    and Use Auth ID: Yes
• click [Submit All Changes] button to restart the phone

Brekeke SIP Server’s Registration PageClick the [Registered Clients] tab at Brekeke SIP Server admintool.
The Linksys IP Phone SPA941 Phone is registered with Brekeke SIP Server.

Lifted this straight from the manual:

Button flashes every 4 seconds
(if SIP server is configured)
HT–486 fails to register

So basicly, that means your HandyTone 486 ATA fails to register with the SIP-server. That means you either have not recieved your IP-tele yet, or that your ISP have failed to register you.

how solve problem for voice2clear unable to register to SIP server

Errors / Solutions (Error 408, 401, 503 / Not Connecting / Failed to Register / Audio Problems):

You can try using port 8891 or 80 setting if 5060 port is blocked

Simply change the sip.tpad.com settings in Xlite to

sip.tpad.com:8891
or
sip.tpad.com:80

Use our STUN Settings, simply use the info on this link (stun.fwdnet.net:3478)
http://forum.tpad.com/viewtopic.php?t=617

http://www.youtube.com/watch?v=ugHwkl--zhQ

Use these other Free Softphones with Tpad:
http://forum.tpad.com/viewtopic.php?t=623
http://forum.tpad.com/viewtopic.php?t=864
http://forum.tpad.com/viewtopic.php?t=657
http://forum.tpad.com/viewtopic.php?t=456

do you have licenses for them.

Did you figure it out?

This in it's most basic description means that the phone is unable to register with the SIP server, most likely for lack of internet connectivity or a firewall blocking the packets.

If you've verified connectivity and opened the proper ports and the phone fails to initialize and still getting “URL Call Disabled” or all of the above parameters are correct, update the firmware for the router. This can usually be done within the router configuration. Then reboot in order, the cable modem or DSL, the router or switch, then the phone. This may solve the problem.

If still having the same issue, confirm in the router configuration for settings such as SIP proxy, gateway, or application layer or any configuration setting for SIP or VOIP. If any of these do exist, make sure they are turned off as these may only confuse or adversely affect many implementations.