1.0 Dashboard

In your daily operations you will be using the VSM web application which provides a graphical interface to manage the data stored in the voipswitch database to which it writes either directly or through a set of web APIs methods.

Although most of the voipswitch settings are in the database still some settings must be kept in a text file. Those include, first of all, the database connection information, listeners IP addresses and ports and logging and dump files configuration. The rest of the settings is grouped under the Settings/System and subdivides to General, WebAPIs and Users menus.

Dashboard view shows us all our platform statistics from where we can see:

• CPU usage,
• Memory usage,
• Disk usage.

We also see the number of:

From the dashboard, we can also capture SIP messages or SMPP logs ,check the live call details with the ability to Barge-in call or use whispering technology to overhear the conversation, we can also check our gateways status:

Live calls

Barge-in and call whispering:

1) Barge and call whispering works only on a live calls so first you need to enable live calls on VSM4 to see the live traffic:

2) Then from the drop down menu select Barge-in call :

3) You have to select the user that you are logged with to be able to connect to the option:

4) You have to select the option from the drop down menu to berge-in or to just Listen the call:

For ended and failed calls there are additional messages on the release reason or the error code.

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
• "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

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

Gateways

Inside the dashboard on the right side we can check gateways status displayed on a round graph

Each of the gateways is pinged with a sip options to check the condition (up/down).

Sip messages

By clicking on the capture button under the sip message we will be able to capture all live new messages 

Once we have captured the message and filtered the messages that interest us, we can select the packages we are interested in and check the message flow:

Smpp messages