TROUBLE SHOOTING
Why is my SIP device displaying "Login Failed- service unavailable"?
Why is my SIP device displaying "Login Timed Out"?
Why is my phone ringing every so often without anyone calling?
Why can't I access my voice mail from a PAP2 or Sipura device?
What ports do I need to open in my firewall?
When my number is called, the system immediately replies with "the number is not available", why?
Why am I being billed for two simultaneous calls?
Why doesn't my virtual number register with the sip proxy?
I can place calls but I am not able to receive them, what is wrong?
Why can't I PING the SIP proxy server?
Why can't I PING my custom SIP Proxy?
SOLUTION
Why is my SIP device displaying "Login Failed- service unavailable"?
Solution:There is no Internet connection available for the SIP device to use.
Why is my SIP device displaying "Login Timed Out"?
Solution:"Login timed out" error is released if the InPhonex number is not correctly typed, there are no dashes. The 7 digit number must be entered as XXXXXXX.
This is most often caused by an incorrect proxy address or password for your account. The proxy should be "sip.inphonex.com". This error may also be caused by poor Internet connection, unstable network, lengthy registration interval or blocked firewall.
To verify this is a blocked firwall issue, please follow the steps below:
1. Turn off the softphone and access www.testyourvoip.com
2. Run the test to Boston.
3. Please make sure the applet does not use a port other than 5060 to run the test. If the applet needs to use a different port, it will release a message.
4. If the applet allows you to complete the test, please let us know the scores.
If you can not run the test due to firewall settings interfering, you will need to set up the firewall to allow the softphone full access to the Internet or turn off the firewall so the softphone can register with the SIP proxy.
Why is my phone ringing every so often without anyone calling?
Solution:Message waiting indication- MWI service
The device will ring when new messages are waiting in the VM box of that number. The ring will not go away until the user accesses VM at 8500, retrieves and deletes the VM messages.
The device controls the MWI service.
If you do no wish to subscribe to this service, please modify the following settings:
For Sipura and LinkSys devices: access admin/advanced/all lines/Supplementary service subscription:
-MWI service: no
-VMWI service: no
For Grandstream devices: access advanced configuration settings:
SUBCRIBE for MWI: No, do not subscribe for MWI.
To adjust the existing MWI settings for Sipura/LinkSys devices: access admin/advanced/all users/ring settings:
- VMWI Ring Splash Lenght: Duration of riing splash when new messages arrive before the VMWI signal is applied (0 � 10.0s) Default is set to .5
- VMWI Ring Policy The parameter controlss when a ring splash is played when a the VM server sends a SIP NOTIFY message to the SPA indicating the status of the subscriber�s mail box.
3 settings are available:
New VM Available � ring as long as there is 1 or more unread voice mail
New VM Becomes Available � ring when the number of unread voice mail changes from 0 to non-zero
New VM Arrives � ring when the number of unread voice mail increases
- New VM Available Ring On No New VM- If enabled, the SPA will play a ring splash when the VM server sends SIP NOTIFY message to the SPA indicating that there are no more unread voice mails.
Why can't I access my voice mail from a PAP2 or Sipura device?
Solution:If you find that you cannot access the VM system even with the correct VM password, the SPA/PAP2 DTMF settings are not correct.
Follow the instructions below:
1. Access the device's configuration settings. If you do not know how to access the device's settings, please use our set up instructions here.
2. Click on the "Admin Login" button near the top right side of the screen
3. Click on the "Line 1" tab
4. Click on advanced "Line 1" tab
5. Scroll down to Audio configuration
6. Look for "DTMF Tx method" settings and change it to "AVT"
7. Update the settings
8. Apply the same procedure for line 2 DTMF setting.
9. Attempt VM access.
What ports do I need to open in my firewall?
Solution:If you have a firewall, you must give the Phone Adapter rights to access certain Internet ports, including:
69-69 UDP
5060-5069 UDP
8000-8003
10,000-20,000 UDP
Depending of the configuration/device used, different ports may be accessed. It's recommended that you put the device IP on the router DMZ.
When my number is called, the system immediately replies with "the number is not available", why?
Solution:Check the account CP to verify the device connected to the number is registering with the system. Next to the number you will see a line containing: Registration expires @ ... UA Type: type of SIP device.
-If the registration line is missing, thee device is not connecting to the system and therefore it will not receive calls. Thus the system will immediately detect the lack of registration and reply with the "person is not available to take your call at this time". In such case, the device is either offline because it is turned it off or there is a problem with it. If the device is not off, the connection to the network should be checked as well as the device configuration settings should be reviewed. A network may be in order.
-If the registration line is present, thee following needs to be verified:
a. check call forwarding-CF for that number. If CF is set to "Send immediately to my voice mail" the system will immediately send the call to VM. The send to VM call forwarding option neds to be removed in order for the device to receive calls normally.
b. the device is having connection issues where the registration interval is so long that the device is offline and the control panel registration is still present. The device registration interval needs to be changed to 300 seconds or 5 minutes.
c. the device has problems with the router, NAT issues:
If the router lost connection to the Internet, most routers will flash for an instant and have to reconnect to the attached devices. As a connection is re-established the router will block outside connections to an internal device unless the internal device makes contact first.
Please consider forwarding the device's ports on the router so it will not cut off your SIP device from outside connections.
If you reboot the SIP device, it establishes contact with the SIP server and then the router opens up that link and allows two way communication between the devices. If the router looses contact with the SIP device, then when the SIP server tries to make contact with the device, the router blocks the connection, hence busy signal.
As with all NAT sessions unless you have port forwarding setup the sessions need to be established from the inside. The problem is if the router resets or has an issue then any existing sessions normally get forgotten about and thus dropped. The SIP protocol deals with this by having a reregistration period.
At InPhonex this reregistration period is 5 minutes or 300 seconds. This normally means that if the SIP device is working correctly and the router does lose its information then the SIP device should fix that issue within 5 minutes.
Putting the SIP in the DMZ on your router should fix this problem as well.
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Note, the system will allow IP to IP calling without registration/authentication. The system will not allow incoming or outgoing PSTN calls without proper device connection to the system.
Why am I being billed for two simultaneous calls?
Solution:Most SIP devices support simultaneous calls via the same line to enable the user to utilize "hold" type functions. Such features like three way calling and call waiting require a call to be placed "on hold" while the second call is initiated. You can disable such functions by accessing the device configuration settings and turning off the service that offers the ability to "flash" and get the stuttered dial tone that allows for a "three way call" or "call waiting". Once you disable these settings, this problem should not re occur. If you are not sure how to do this, please review the device's manual for exact instructions.
Why doesn't my virtual number register with the sip proxy?
Solution:1. Check the following items.
Follow our online configuration guide at http://www.inphonex.com/support/xten-softphone-configuration.php
Make sure you have the correct VN and password
2. Disable any personal firewall.
Personal firewalls like Norton Personal Firewall could be blocking sip ports from 5060 to 5069. Blocked ports can also cause one way audio.
3. Disable Windows XP firewall
Windows XP has a built-in firewall that could also be blocking these ports.
In order to verify if your Firewall is enabled follow these steps:
1. In Windows, right click on my network places and select PROPERTIES.
2. Once in the Local Area Connection properties window, Click on the ADVANCED tab.
3. Under windows firewall click on SETTINGS. This will open de firewall settings window.
4. Set the firewall to OFF.
If you do not want to Disable your Windows XP Completely:
1. In Windows, right click on my network places and select PROPERTIES.
2. Once in the Local Area Connection properties window, Click on the ADVANCED tab.
3. Under windows firewall click on SETTINGS. This will open the firewall settings window.
4. Once in Windows firewall, click on the EXCEPTIONS tab.
5. Under PROGRAMS AND SERVICES look for the X-lite check box and make sure it is checked.
I can place calls but I am not able to receive them, what is wrong?
Solution:Make sure that every time you install any softphone in Windows 200 or Windows XP you are logged on as the computer Administrator, which will give you full control NTFS permission to the folders needed to access during the installation.
Do not install the softphone using the "Run As" option; this option is not meant for installations but to run a program.
If you use this (Run As) option you might not get an error message; but due to insufficient NTFS permissions for some folders, some components won't be installed, causing this application to fail.
Why can't I PING the SIP proxy server?
Solution:The SIP proxy server no longer accepts or returns ICMP packets. This is normal and does not indicate that the service is down.
Why can't I PING my custom SIP Proxy?
Solution:Not being able to ping your custom sip proxy can be caused by the following:
1. You pointed your sip proxy to sip.varphonex.com as recommended. If this is the case, keep in mind that for security reasons our IP address is not pingable anymore.
2. You are pointing your custom sip proxy to and IP address no longer in use.
In order to determine where your sip proxy is pointing to; you can run the nslookup command in the command prompt.
nslookup sip.youdomain.com