Difference between revisions of "IP PBX Manual Reporting Diagnostics"
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− | | <span style="color:#ff0000">WARNING[19775] app_queue.c: Unable to join queue 'rg_2'</span><br/> | + | | <span style="color:#ff0000">WARNING[19775] app_queue.c: Unable to join queue 'rg_2'</span><br/> |
− | | Due to any number of configuration reasons, the call was not able to join the group. | + | | Due to any number of configuration reasons, the call was not able to join the group. It could be the group is empty, it could be the group has a limit to how many calls it can process at one time. |
− | |||
|- | |- | ||
| <span style="color:#ff0000">ERROR[18636] utils.c: write() returned error: Broken pipe </span><br/> | | <span style="color:#ff0000">ERROR[18636] utils.c: write() returned error: Broken pipe </span><br/> | ||
− | | This means that an audio file was playing (MOH, Prompt, VM) and the call was ended (user hung up, call dropped, etc). | + | | This means that an audio file was playing (MOH, Prompt, VM) and the call was ended (user hung up, call dropped, etc). |
|- | |- | ||
| <span style="color:#ff0000">WARNING app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)</span><br/> | | <span style="color:#ff0000">WARNING app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)</span><br/> | ||
− | | This occurs when someone tries to dial a SIP device that is supposed to be there but for some reason it cannot be contacted. This could be an extension that was registered but is not responding because it has lost power, it could be a SIP provider that cannot be reached. There are a lot of issues that can cause this message.< | + | | |
+ | This occurs when someone tries to dial a SIP device that is supposed to be there but for some reason it cannot be contacted. This could be an extension that was registered but is not responding because it has lost power, it could be a SIP provider that cannot be reached. There are a lot of issues that can cause this message. | ||
+ | |||
+ | |- | ||
+ | | <span style="color: rgb(255, 0, 0); font-family: 'Lucida Sans Unicode', sans-serif; line-height: 21.333332061767578px;">[2014-08-24 21:23:24] WARNING[8134] pbx.c: Unable to register extension 'XXXXXXXXXXX', priority 1 in 'dids', already in use</span> | ||
+ | | This occurs when you have the same DID entered twice under the same or different providers. Since DIDs are global, they only need to be entered once, regardless of how many different carriers/providers are being used on the system. Delete any duplicate instances of the DID and the issue will be resolved. | ||
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Latest revision as of 12:13, 29 August 2014
Diagnostics
This section, when configured under PBX Setup->Services, displays a message log of actions in the PBX. Typically this page will be referenced by Tech Support to pinpoint where troubleshooting for a specific problem should begin.
View System Diagnostics
STEPS:
- Click the Reporting->Diagnostics linkon the Admin Page.
- The System Diagnostics page appears displaying the messages that have been processed and their status.
- The type of message that is displayed on this page is set in the Logging Level section of the Systems page.
Common Error Messages and their Meanings
Error Message |
Problem |
rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP XXX.XXX.XXX.XXX |
This may or may not be an issue. If you have a local client that has comfort noise support on, turn it off. If it is a Provider, you may want to set RTP-Keep Alive. Note if you have no music on hold and you place a client like this on hold, they will be disconnected if you have set RTP Timeout or RTP Timeout on hold, once this timer has elapsed. |
NOTICE chan_sip.c: Disconnecting call 'SIP/XXX-XXXXXXX' for lack of RTP activity in 11 seconds |
If you have RTP Timeout set, (which is a good thing). Then a call to SIP/<EXTENSION# or PROVIDER NAME> was terminated because voice traffic that was expected did not transmit for 11 seconds. |
Call from to extension 'XXXX' rejected because extension not found |
Someone tried to make a guest call to your system. If Guest calls are allowed under PBXSetup->SIP, then they can be made to 's' by or any dids only. If you notice a lot of these and that your system performance is degrading, it is recommended that you do a packet capture to determine the offending IP address and then block it with an ACL entry to prevent system performance from being affected. Note if from has an extension in it then it indicates that a number that is not supported by the dialplan was dialed. Like 1 for instance. |
WARNING[19775] app_queue.c: Unable to join queue 'rg_2' |
Due to any number of configuration reasons, the call was not able to join the group. It could be the group is empty, it could be the group has a limit to how many calls it can process at one time. |
ERROR[18636] utils.c: write() returned error: Broken pipe |
This means that an audio file was playing (MOH, Prompt, VM) and the call was ended (user hung up, call dropped, etc).
|
WARNING app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
This occurs when someone tries to dial a SIP device that is supposed to be there but for some reason it cannot be contacted. This could be an extension that was registered but is not responding because it has lost power, it could be a SIP provider that cannot be reached. There are a lot of issues that can cause this message. |
[2014-08-24 21:23:24] WARNING[8134] pbx.c: Unable to register extension 'XXXXXXXXXXX', priority 1 in 'dids', already in use | This occurs when you have the same DID entered twice under the same or different providers. Since DIDs are global, they only need to be entered once, regardless of how many different carriers/providers are being used on the system. Delete any duplicate instances of the DID and the issue will be resolved. |