Trouble Shooting
This trouble-shooter should help you to identify and fix
problems. Most problems are due to incorrect setup / configuration of the VoIP phone, ports
blocked on a Firewall or problems with the Audio. The error response codes listed below can
help to identify what the problem is.
Sound Quality decrease after installation of Service Pack 2 on Windows XP
If you have WindowsXP on your computer and have installed Service Pack 2, you may encounter sound problems that can be fixed by
clicking on this link and following the instructions.
Common error response codes
These codes will be displayed on the Pc Phone screen or in the diagnostic log, viewable by pressing F9.
-
100 codes - provisional responses
-
200 codes - request successful
-
300 codes - redirection
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400 codes - request failure
-
500 codes - server failure
-
600 codes - global failure
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SIP methods
1. VoIP Phone Configuration
The most common problem is that the VoIP Phone is not configured correctly. Please double check that you have set it up as per the configuration guides, on the left, and also ensure that you are using the VoIP password that was emailed to you and not
the web login password that you changed on first login.
2. Firewall and NAT problems
If you have a broadband router with a NAT firewall, you must open certain ports the VoIP phone to be
able work. Configure your router to open these ports:
UDP 5060 - SIP
UDP 8000 - RTP
UDP 8001 - RTP
See your router guide for specific instructions regarding your firewall and opening ports.
3. Audio problems
Identify audio problems when making calls using the ComsPhone software.
First check that your headset is working correctly. Click option and the pull down box to ensure that the correct Audio Device is selected.
| Problem |
Description |
Action |
| Popping sounds |
This is caused by 'over-modulation'. The signal volume is too
loud for the hardware to cope with. |
Ask the person who you are calling to turn their microphone
volume down and/or adjust the microphone gain. |
| Background noises |
Silence threshold set at the wrong level.
|
Ask the person who you are calling to adjust their silence threshold. |
| Echoing |
Echoes when talking |
Try using headset with your soundcard.
Use a USB-enabled headset. (Some boom headsets may create feedback between the boom microphone
and headset).
You could be too close to the person you're speaking to.
Reduce your microphone level, or turn off your amplifier in the playback section of Windows audio
controls. |
| One-way audio |
Some Router-PC-Softphone setups will override the "Reverse USP Mapping
Rules", transmitting audio in only one direction. |
Retry your call after changing to this setting:
Menu > Advanced System Settings > RTP Settings > Obey Reverse UDP Mapping Rules: No |
| Line drops or poor overall quality
|
Voice and tone quality disrupted by noise.
|
Don't transfer any large files during the call. This affects call
quality.
Select a codec with an audio compression using less bandwidth, e.g. codec G.729.
Note: Overall call quality depends on the quality of your soundcard. |
| Phone rings and call is accepted, but unable to hear.
|
Both caller and called are unable to hear each other. |
Check that the ports listed above are all open. |
| No sound or clipped sound |
Random silence breaks |
Your Transmit Silence settings may be wrong. |
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Common error response codes
These codes will be displayed on the screen or in the diagnostic log, viewable by pressing F9.
-
100 codes - provisional responses
-
200 codes - request successful
-
300 codes - redirection
-
400 codes - request failure
-
500 codes - server failure
-
600 codes - global failure
-
SIP methods
100 codes - provisional responses
'100' codes are sent when a request has been received from a caller, but the final outcome has
not yet been determined.
| Code |
Message |
Description |
Action |
|
100
|
|
An action is being taken (e.g. a database is being consulted). The called has not been located yet.
|
Manually configure the network setting dialog screen;
auto-detect has failed to detect proper settings.
|
|
180
|
|
|
Wait for call to be accepted.
|
| 181 |
'Call is being forwarded' |
Call being forwarded to a different set of destinations.
|
Wait for call to be forwarded. |
| 182 |
'Queued' |
Called temporarily unavailable, but has decided to queue the call rather
than reject it. |
Wait for call to be accepted. |
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200 codes - request successful
'200' codes are sent when a positive final response to a request from a caller has been received.
| Code |
Message |
Description |
Action |
| 200 |
|
|
Required information has been returned.
|
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300 codes - Redirection
'300' codes are sent to notify the caller that they will be re-directed to another location.
| Code |
Message |
Description |
Action |
| 300 |
|
User no longer at address.
|
Retry any new address(es) provided.
|
| 302 |
'Moved temporarily' |
Redirected call to another number. |
Retry any new address(es) provided. |
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400 codes - request failure
'400' codes are sent when a negative final response to a request from a caller has been received.
The problem is on the caller's side.
| Code |
Message |
Description |
Action |
| 400 |
|
Request could not be understood.
|
Check details and retry.
|
| 401 |
'Unauthorized' |
Request requires authorisation. |
Enter correct username and/or password. |
| 403 |
'Forbidden' |
Server has received the request but won't provide the service (i.e.
conference call feature).
|
Conference calling is not currently supported. |
| 404 |
'Not Found' |
User does not exist at domain specified. |
Check dialling instructions. |
| 405 |
'Method not allowed' |
Request contains one or more methods which are not allowed. |
Check details and retry. |
| 406 |
'Not Acceptable' |
Request cannot be completed by the server.
|
Check details and retry. |
| 407 |
'Proxy authentication required' |
Client must first authenticate itself with the proxy server. |
Re-enter correct username and password. |
| 408 |
'Request Timeout' |
The server could not respond before timing out. Possible causes:
- Service temporarily unavailable.
- Wrong SIP proxy settings.
- IP proxy not found. |
The VoIP phone is trying to create a voice-channel link via the STUN server. Some
routers and firewalls block these by closing ports.
Configure your router to open these ports:
UDP 5060 - SIP
UDP 8000 - RTP
UDP 8001 - RTP
|
| 410 |
'Gone' |
Number no longer available, no known forwarding address known. |
Unable to make call. |
| 412 |
'Extension Required' |
The server is asking you for an additional number.
|
Enter the extension number. |
| 413 |
'Request Entity Too Large' |
What you have requested is too big for the server to process.
|
Reduce size and retry. |
| 414 |
'Request-URI Too Long' |
The URI you have entered is too long for the server to understand.
|
Check details and retry. |
| 415 |
'Unsupported media type' |
Message body of your request is in a format not supported by the server. |
Retry using an acceptable format. |
| 420 |
'Bad Extension' |
The server didn't recognise the extension entered. |
Check details and retry. |
| 480 |
'Temporarily unavailable' |
The SIP user you dialled isn't currently online.
|
Test-call the number of a user who you know to be online.
|
| 481 |
'Transaction does not exist' |
Server has received a BYE or CANCEL request which it can't match to an
existing transaction. (i.e. the call has been terminated) |
Call already terminated. |
| 482 |
'Loop Detected' |
The server has detected a loop. |
Call will be disconnected automatically.
|
| 483 |
'Call failed: Too many hops' |
Server received a request that required When attempting to access
Voicemail (7001) and calling other numbers. |
Call will be disconnected automatically. |
| 484 |
'Address Incomplete' |
You made a request that doesn't match any existing transaction. |
Call will be disconnected automatically. Check details and retry.
|
| 486 |
'Busy' |
Line is engaged. |
Try calling later. |
| 487 |
'Call Terminated' |
A BYE or CANCEL request terminated the call. |
Call will be disconnected automatically.
|
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500 codes - server failure
'500' codes are sent when a valid request from a caller has been sent, but the server is unable
to complete it. The problem is on server's side.
| Code |
Message |
Description |
Action |
| 500 |
'Server Internal Error' ' |
Temporary failure. |
Call will be disconnected automatically. Try calling later.
|
| 502 |
'Bad Gateway' |
Network out of order. |
A fast 'busy' signal will be heard and the call will be
disconnected. |
| 503 |
'Service unavailable' |
Temporary failure. |
Call will be disconnected automatically. Try calling later.
|
| 504 |
'Server Time-out' |
The gateway has timed out.
|
A fast 'busy' signal is heard and the call is disconnected.
|
| 513 |
'Message too large' |
The body of your message is too large. |
Reduce length of your message. |
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600 codes - global failure
'600' codes are sent when the request cannot be completed by any server.
| Code |
Message |
Description |
Action |
| 600 |
'Busy Everywhere' |
Called doesn't wish to take the call at this particular time. |
Call will be disconnected automatically.
|
| 603 |
'Decline' |
Called doesn't want to or can't take call. |
Call will be disconnected automatically. |
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SIP methods
SIP uses a number of commands, called 'methods'. Some of the most common methods used are:
| Method |
Description |
| INVITE |
Invites a user to join a session, creating a new connection. A
description of the session may be displayed in the message body. |
| ACK |
Confirms that a user has received a final response to an INVITE request.
It may contain a description of the final session in the message body. |
| BYE |
Cancels a connection between users or to decline a call. |
| CANCEL |
Cancels a session or search which isn't yet fully established.
|
| OPTIONS |
Queries the capabilities of the server. |
| REGISTER |
Registers your current location. |
| INFO |
Used for mid-session signalling. |
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