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Sometimes it is possible to have issues when trying to register your Device / Softphone with VoIP.ms, we will review some recommendations and important things to check, to start trouble shooting with this situation.
How can we notice there is an issue with registration? Some indicators that can alert us to it:
- When I try to make a call, there is no dial tone on the line.
- I used to have a working DID number and I no longer receive calls, or calls are going directly to my voicemail or to a failover.
- At the customer portal I can not see my SIP Registration (Note: This does not apply for IP Authentication).
It is possible that some specific situations will need to be addressed with VoIP.ms staff, but with this article you can start looking for potential causes and solutions for your issue.
Reboot Your Devices
Most common issues related with registration failing, can be resolved just by rebooting the Device/Softphone, the router and the modem. This suggestion can usually help in the case you had a power outage or if your Internet connection stopped working for a while.
Note: Considering the use of failover options can help to prevent losing calls during the trouble shooting, you can use a call forward or a voicemail along with your number.
Some common suggestions that can be followed if the issue is related with an ATA device or a softphone:
- Verify if your device has any field to set the "Register expires", if you find something like this, you usually will see a default of 3600 (seconds), lower this value to 300 (5 mins). This tells our side that you are no longer registered and then failovers will kick in immediately.
- Register Expires is the parameter that controls how often your client contacts the SIP server to remind it that the client is alive and confirming its current location (public IP address and listening SIP port). The SIP server is supposed to set this timer as part of the reply to each Register command. If the automated setting doesn't work you need to set this parameter manually according to the provider's instructions.
- Use the IP address from the server instead of the domain name, example: Use 220.127.116.11 instead of losangeles.voip.ms. This item is recommended as diagnostic only, if this works for you, then it is probably a DNS issue affecting in your network, test using it for a couple of weeks and then change again to the domain name, it is not recommended to have the registration with the Ip all the time, VoIP.ms will redirect the domain to another IP in the case of a server issue, but this will not work for you if you are not using the domain.
- If your device has any options similar to "Nat Keep Alive" or "Nat Mapping" please enable them. This will make sure the connection does not go idle.
- Check if you have the Latest Firmware updated on your device, some minor bugs get resolved with the new firmwares, so it is important to have them up to date.
By default is not necessary to open ports in the router and this is only required in the case of some specific firewall rules are blocking traffic on those ports, however if issues are present, we can forward ports UDP 5060 and also 10000 to 20000 UDP, in the router. Another suggestion is when using different devices under the same network, you can designate an internal port for each one (for instance using port 5061 in one device, 5062 in next one, and so on). Communication between the VoIP.ms and the router will still go through port 5060, even though different internal ports are used for each device.
Do you suspect your ISP could be blocking these ports ? It is recommended to contact them just to be certain.
If you have a device capable of supporting IAX2 protocol, you can use this as an alternative, just make sure port UDP 4569 is not blocked in your network.
Can`t Connect or Register to Login Server
Check if your device shows any information related with the error when it can not connect, for example on some ATA devices, you can see the "Registration Status", if it shows something like "Cant connect to login server" or Failed, you can follow these suggestions:
- This error is usually caused by the router that somehow is blocking the registration attempts, try rebooting the router then the ATA device, if that does not work, re-enter the password and UserID on the device settings (Please check this Page for Main Login Credentials), or try using the IP address of the server instead of the hostname. If you do not have the IP address, you may see all of our IPs here.
- If this is a new installation and you are just trying to get the service working, then if at all possible please remove your router from the loop. Connect your ATA Device / Computer directly to the Internet without the router in between. If you can then register and receive / make calls you will know there is a router configuration problem preventing proper connectivity.
- You can attempt to use a different SIP port, such as 5080 and 42872
Tomato Firmware (PAP2TNA).
This part is for PAP2T devices, when using the Tomato Firmware:
- Some of the default settings with Linksys devices conflict with the default Tomato timeout settings. These are the UDP Timeout settings and if they are left at their defaults, VoIP devices will sometimes fail to register after an IP change. These may be found on the Conntrack/Netfilter page in the Tomato config. Set "unreplied timeout" to 10 seconds.
Asterisk /PBX system.
Some common suggestions that can be followed if the issue is related with an Asterisk system or a PBX:
- Add to your trunk nat=yes and qualify=yes, these 2 values can help with your registration issues.
- Use the IP address from the server instead of the domain name, example: Use 18.104.22.168 instead of losangeles.voip.ms. This item is recommended as diagnostic only, if this works for you, then it is probably a DNS issue affecting in your network, test using it for a couple of weeks and then change again to the domain name, it is not recommended to have the registration with the IP all the time, VoIP providers tend to redirect the domain to another IP in the case of a server issue, but this will not work if you are not using the domain.'
The NAT option determines the type of setting for users trying to connect to an asterisk server. With VoIP.ms it is recommended to have the NAT option set on Yes, which is the option that will work best.
Portal settings influencing NAT with Asterisk:
- yes = Always ignore info and assume NAT
- no = Use NAT mode only according to RFC3581 (;rport)
- never = Never attempt NAT mode or RFC3581 support
- route = Assume NAT, don't send rport (work around more UNIDEN bugs)
This feature may also be used to keep a UDP session open to a device that is located behind a network address translator (NAT). By sending the OPTIONS request, the UDP port binding in the NAT (on the outside address of the NAT/firewall device) is maintained by sending traffic through it. If the binding was to expire, there would be no way for Asterisk to initiate a call to the SIP device. This can be used in conjunction with the nat=yes setting.
Note that it is the qualify=xxx(in miliseconds) or qualify=yes or qualify=no in sip.conf file that determines the registration with the server. If you have "yes" for qualify it will check the registration with the server every 2 seconds.