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1.12.1 Voipswitch GUI - real time monitoring

1.12.1 Voipswitch GUI - real time monitoring

Owned by Former user (Deleted)
Last updated: Oct 06, 2015Version comment

 

The voipswitch application GUI consists of the main toolbar and several windows.

Clicking on the buttons open the respective windows. Clicking on the title bar will open a menu with additional options:

  • Log configuration – allows for changing the application’s logging levels
  • Dump configuration – enable/disables the application dumps
  • Create user dump – creates a current memory dump
  • Save layout – saves the current layout of the application into an XML,

Calls window

This window contains a list of client connections processed by voipswitch at the moment. Connections which have not been authenticated are not included in the list.

There are several icons representing distinct states of the call:

  • Red crossed icon – call failed due to internal reasons, e.g. missing prefix in the tariff, no routing plan entry etc.
  • Blue stop icon – unsuccessful call, e.g. cancelled by the client or declined by the destination
  • Blue play icon – pending call, not connected yet
  • Orange play icon – connected active call
  • Orange stop icon – successful call, already ended

If you need more details on a particular call click on the plus icon next to it to expand the view. You will see detailed information for each phase of the call.

 

Clicking right mouse button on the Calls window title bar opens a menu with the following options:

  • Freeze calls list – stops the calls list, new connections are not shown
  • Follow tail – automatically scrolls the list so that the latest connections are always visible in the window
  • Maximum logs – sets the max number of connections shown in the window, when the limit is reached the oldest connections are removed
  • Filter – allows for filtering the list by a login and/or a dialed number
  • Reload settings – loads the general system settings without stopping the application; the settings are configurable in the VSM Settings/System/General

Troubleshooting calls related problems

The most common problems which can be easily traced using the voipswitch GUI are related to making calls.

If a call is not shown in the Calls window check the Logs list to see if there are any messages about unauthorized connections. The Calls window shows only authenticated connections coming from active clients.

Click on the plus icon next to the call to see the extended view. The view shows each phase of the call along with detailed information which include the errors in case of the failure.

Below is the list of the call’s related messages:

  • “Destination response:” – when call is answered, shows media codecs which were chosen – taken from the SDP carried in 200 OK response from the destination endpoint.
  • “Response sent to client:” – when the call is answered, shows the negotiated codec which is sent to client by voipswitch in its 200 OK response
  • "Cannot create RTP channel at address: ..."  - when voipswitch cannot create a media channel at given IP and port, e.g. when the port is taken already by other application
  • "Action (...) for number ... fired: after connection actions, reason (...)” – when voipswitch processes the answering rules associated with the destination client account.
  • "Action (...) for number ... fired: before connection actions, reason " – same as above, the difference is in when the actions are fired, in this case before the call is sent to the destination, e.g. Do Not Disturb
  • "Forking call ..." – when the destination client is registered from multiple endpoints or when the call is sent to a PBX hunt group
  • “Redirecting to number (302 received) …..” - when a call is being redirected by voipswitch on receiving SIP 302 Moved Temporarily response from destination
  • “SQL statement … did not return a valid result” – when a lookup SQL scenario does not return valid result
  • "Cannot resolve domain name..." – when voipswitch cannot resolve FQDN, e.g. an FQDN defined in the Destinations/Gateways menu.
  • "Client and destination ... have no common codecs” – when the client and the destination have no common codecs in the list of allowed codecs.
  • "Destination response (200 OK) has no SDP" – when 200 OK received from the destination has no SDP
  • "Leg has ... codec(s) instead of allowed ..." – when the leg presents a codec which is not allowed in the client account
  • "Media timeout: ... stopped sending media!" – when either party has not sent any media for the defined timeout
  • "Destination ... is not responding" – when there is no response from the destination endpoint for the INVITE request within 5 seconds
  • "Concurrent calls limit for ...  has been reached" – when the limit of conncurent calls set for the cleint account has been reached
  • "Call connected at ..." – when the call is connected – received 200 OK
  • "Call ended at ..."
  • "Voipswitch is not registered to the remote registrar ..."  – when the call is routed to a registrar to which voipswitch is not able to register
  • “Destination endpoint for number ….. not registered”  – when the destination endpoint is not registered to voipswitch’s SIP registrar
  • "Finishing forked calls..." – when the forked calls end or one of the calls is connected, the other will be ended
  • "Client does not have the required FAX codec allowed" -  when a client attempts to send fax from the VUP and there is no t38 codec enabled in the client account
  • “Destination limit …. has been reached” – when the limit set for the destination gateway has been reached
  • “Destination is inactive” – when a call is routed to a destination gateway which is set to inactive (in VSM)
  • “Destination does not exist in the database” – when a call is routed to a gateway which does not exist in the database
  • “Call duration limit has been reached”– the Limit call duration setting from the VSM Settings/System/General has been reached; the call is ended by voipswitch
  • "Timeout occurred : no ALERTING message" – when after sending an INVITE request to destination endpoint, voipswitch receives 100 Trying response but will not receive subsequent 180 Ringing or 183 Session progress in the time defined in the Alerting timeout setting in VSM Settings/System/General. The call will be ended by voipswitch.
  • “No matching client login:” – when the Retail client/User lookup route type is used and there is no matching login  in the client database
  • "You should start a SIP listener at address ... to reach destination ..." – when the destination is reachable only from a network for which there is no voipswitch listeners running. Happens on servers with multiple network cards.
  • "Blind transfer to number ..." – when a call is being transferred (SIP REFER)
  • “Timeout occurred: no response to re-INVITE”– when there is no response to an INVITE request sent in a session
  • "Ring timeout" – when the timeout set in the Limit ring time in Settings/System/General is reached
  • "Unknown lookup protocol" – when the protocol part of the URL defined in the Lookup type route is not recognized by voipswitch
  • “Incorrect dialed number length”– when the dialed number length is different than the one defined in the Dialing rules (either in the client or Routing plan)
  • "Attended transfer initialized"
  • "Blind transfer to number... has failed"
  • "Attended transfer has failed"
  • "Attended transfer has failed: incompatible codecs" – when the transferee and destination have no matching codecs
  • "Leg has been transferred"
  • "SDP parsing error" – when an SDP received by voipswitch is not valid and therefore cannot be processed
  • "Media has changed: ..." – occurs when the media during the call change, for example when starting video during an audio call or when call is put on HOLD.
  • "Media proposal: ..." – on receiving re-INVITE with new SDP

Billing

  • "Number ... is disabled in destination tariff ..." – when there is no matching prefix in the cost tariff assigned to the destination gateway or registrar
  • "Number ... is disabled in tariff ..." – when there is no matching prefix in the client tariff
  • "Number ... is disabled in plan" – when the matching entry in the assigned tariff bundle is disabled
  • "... has no money to make a call" – when the client has not enough balance on his account
  • "Cannot find voice rate for number ... in tariff ... day: ... hour: ..." – when there is no matching prefix in the client tariff
  • "... has run out of money" – when there is no sufficient funds to continue the call
  • "Destination ... has higher tariff's rate than ..." – when the option Refuse connection when destination rate is higher than client is on and the rate in the cost tariff is higher than the rate in the client tariff
  • "Cannot find tariff for destination ..." – when there is no matching prefix in the client tariff

Routing

  • “ Max number of failover hops has been reached”– when the value set in the option Limit number of all hops in the Settings/System/General has been reached during the failover procedure
  • " Route disabled" – when the Route disabled option in the Routing plan is set
  • “Prefix limit has been reached”– when the calls limit set in the Routing plan for the prefix has been reached
  • “Prefix disabled” – when the Prefix disabled option is set in the Routing plan for the matching prefix
  • "Cannot find destination for number ..." – when there is no matching prefix in the routing plan
  • “Disabled failover for the route” – when the option Disable failover is enabled, voipswitch ends failover procedure

Voipbox

  • “Sending media before connection disabled: incompatible DTMF method” – when the option Send media before connect in the Routing plan (only for voipbox scenarios) is on but the calling client does not support DTMF RFC2833. In such case the call gets connected immediately.
  • “Entered PIN is incorrect”– when the PIN entered in calling card scenario is wrong – failed authentication attempt
  • "Client recognized as ..." – occurs in the calling cards scenario, when the caller authenticates itself by PIN or the caller ID.
  • "Entered number : ..." – shows the number entered by the client using DTMFs in the calling cards scenarios (through the IVR)
  • “Cannot pass through NAT: no RTP packets received” – when a client call is connected to voipbox but there is no RTP packets coming from the client endpoint which is recognized as located behind NAT
  • “PIN scenario cannot be processed: PIN authorization not enabled”– when a wholesale type client sends a call to one of the voipbox PIN scenarios but the option Enable PIN authorization is off in that client account settings.
  • “Requested scenario … is not loaded by voipbox”– when a voipbox scenario is defined in the Routing plan but is not loaded in voipbox application. The warning is shown also when voipswitch is started before voipbox. Proper sequence is first voipbox and then voipswitch.
  • "Error : VoipBox at address  is not running or responding"

 

UC/PBX

  • “Cannot find a retail client with id …., company name ….” – when the company UC/PBX account is inactive or does not exist
  • “All members of the group … are busy or not reachable”
  • "Group ring timeout" – when call is routed to a PBX hunt group and ringing state has reached the timeout

Callback

  • "Callback will be triggered in ... seconds to number ... and ..."  – notifies before the callback action is executed; occurs only for callback route types
  • "Problems with tariff for source leg number, check the source leg logs" – when there is no prefix in the tariff used by the callback’s source leg
  • “CLI based authentication required: CLI …. authentication failure ”– when the callback scenario accepts only caller ID based authentication and the received caller ID cannot be authenticated (is not assigned to any of the client accounts)
  • "Voipbox cannot act as a source leg "– occurs in callback scenarios, the scenario cannot be configured as the source call leg.
  • "Destination ...  does not have codec ... chosen for the source leg "– occurs in callback scenarios, when the destination does not have the codec chosen for the source leg

 

Transcoding

  • "Transcoder from ... to ... has been started"
  • "Transcoder from ... to ... has been stopped"

Parking and PUSH

  • "Call parked, push notification sent"   – when a PUSH notification is sent to the destination
  • "Attempt to send push notification failed" – when the PUSH request sent from voipswitch to EMCI fails (e.g. when there is no connection to the EMCI API)
  • "Call pickup attempt failed: no parked call with requested call-id”  – when an INVITE with Replaces header is received but there is no parked call with matching call-id; e.g. when the call ended before the INVITE has come
  • "Received call pickup attempt" – when a parked call is picked up
  • "Attempting call pickup" – when an INVITE with Replaces header has been received, e.g. when picking up a parked call. Occurs also in PUSH scenario
  • "Park orbit timeout" – when the duration of the stay for a parked call reaches the parking timeout. The call is then sent back to the client who parked the call.
  • "Parked call sent to number: ..." – when voipswitch sends the parked call back to the client who parked the call
  • "Call parked at park orbit: ..." – when a call is being put on parking
  • "Leg moved to parked call" – when a client picks up a call from parking
  • "There are no calls parked at: ..." – when a client connects to the parking in order to pick up a call and there is no waiting calls at the moment

 

Registered clients window

The window contains the list of the SIP endpoints registered to voipswitch’s registrar.

You can select a client and double click on it, a window will appear with the basic information about the client account such as tariff, balance, codecs, tariff and dialing rules.

Clicking right mouse button on the title bar opens a filer window. You can filter the list by the client login or client’s IP address. When filtering by the login you can use a regular expression, in this case tick the Regular expression mode checkbox.

Statistics window

The statistics window is divided on the Incoming and Outgoing panels. It is because the voipswitch works as the B2BUA and therefore the inbound connections should be treated separately from the outbound calls. There are several major information shown in the panels:

  • pending – all calls which are being processed, the number includes also connected calls
  • connected – number of calls that are in connected state
  • total calls – number of all processed calls since the application was started
  • total connected calls – number of all connected calls since the application was started
  • ASR - Answer Seizure Ratio
  • ACD - Average Call Duration

Incoming statistics are calculated from the customer's perspective while outgoing are for the voipswitch owner. An example of the difference is visible when a client is calling to number which is then sent first to an off-line gateway and afterwards is rerouted to a gateway which successfully connects the call. The outgoing ASR will be 50% because the first call to the first gateway failed and the second was connected by the next gateway. The incoming ASR will be 100% because only one call from the client was received and connected successfully (regardless of the fact that two attempts were made by voipswitch). The total calls in this example for incoming statistics will be 1 and for outgoing 2.

Logs window

This window shows log messages generated by the application. When voipswitch starts you should see the following information in the log window:

  • Application version,
  • List of the active (licensed) modules
  • Status of the connection with the database server
  • Started listeners
  • Transcoding settings
  • Voipbox connection status

It is recommended to check the log window after starting voipswitch to make sure that all the components and connections has been properly initialized.

When voipswitch is running the log window displays various information depending on the events that are taking place, for example it shows authentication failures, errors in saving information to the database etc.

Registrars window

Shows the remote SIP registrars to which voipswitch should be registered as an endpoint. Registration is sometimes required by SIP trunks providers. The blue icons indicates that voipswitch is currently registered. The red icon means that an error has occurred during registration.

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