Test Plan

DESCRIPTION

EXPECTED RESULT

Step 1

Go to Dashboard → Services and restart the SIP server service.

The 3CX SIP Server and any associated services will stop and then start automatically

Step 2

Registration based Trunks:

Wait for the server to start and make sure that the DNS resolution was done correctly in Wireshark.

IP based Trunks:

Force an Outbound call to force a DNS lookup.

The PBX must check for A and AAAA records as well as NAPTR and SRV records. All results must match what is configured for the registrar.

DESCRIPTION

EXPECTED RESULT

Step 1

Go to Dashboard → Services and restart the SIP Server

The 3CX SIP Server and any associated services will stop and then start automatically

Step 2

Wait for the services to start, then go to “Voice & Chat” in the Admin Console and check the registration status of the SIP Trunk

The SIP Trunk after the services have restarted should be seen as “Registered” with a green status

Step 3

Check the capture file and ensure that the registration sequence is correct

Check that 3CX sent the REGISTER message correctly and that the SIP Trunk provide responded accordingly

DESCRIPTION

EXPECTED RESULT

Step 1

Go to Dashboard → Services and restart the SIP Server

The 3CX SIP Server and any associated services will stop and then start automatically

Step 2

Wait for the services to start, then go to “Voice & Chat” in the Admin Console and check the registration status of the SIP Trunk

The SIP Trunk after the services have restarted should be seen a “Registered” with a green status

Step 3

Wait more than 60 minutes

Allow enough time for 3CX to refresh the registration multiple times

Step 4

Go to “Voice & Chat” in the Admin Console and check the registration status of the SIP Trunk.

The SIP Trunk should still be seen as “Registered” with a green status

Step 5

Check the capture and ensure that 3CX respects the ‘Expiry’ time set by the provider and that registrations occur at correct intervals

3CX should respect the ‘Expiry’ sent by the provider in the 200 OK and refresh the registration within the ‘Expiry’ interval

DESCRIPTION

EXPECTED RESULT

Step 1

Ensure that the SIP Trunk is registered

This scenario must start off with the SIP Trunk registered

Step 2

In the “Voice & Chat” section of the Admin Console, select the SIP Trunk and press “Refresh Registration”

3CX will release the registration binding and then immediately send a new registration request

Step 3

Click away from the “Voice & Chat” section and the go back, then check the SIP Trunk is registered

The SIP Trunk should be seen as “Registered” with a green status

DESCRIPTION

EXPECTED RESULT

Step 1

Ensure that the SIP Trunk is registered.

This scenario must start off with the SIP Trunk registered.

Step 2

In the “Voice & Chat” section in the Admin Console, highlight the SIP Trunk and go to the “Options” tab. Disable the options “Allow inbound calls” and “Allow outbound calls”. Press “OK” to confirm.

3CX will release the registration binding and

stay unregistered.

Step 3

Reverse the process in Step 2 by re-enabling the 2 options. Press “OK” to confirm.

3CX will send a new registration and the trunk will register.

DESCRIPTION

EXPECTED RESULT

Step 1

Go to Dashboard → Services and restart the SIP Server

The 3CX SIP Server and any associated services will stop and then start automatically

Step 2

Wait for the services to start, then go to “Voice & Chat” in the Admin Console and check the registration status of the SIP Trunk

The SIP Trunk after the services have restarted should be seen a “Registered” with a green status

Step 3

In the capture check the REGISTER message that was sent:

• If the packet was sent over TCP, the REGISTER message contains “transport=TCP”
• If the was sent over UDP, the REGISTER message does not contain “transport=TCP”

If 3CX based on the settings or DNS resolution chooses to send the REGISTER message over TCP transport, then the REGISTER message must contain “transport=TCP”.

If UDP is used, it must not contain this.

DESCRIPTION

EXPECTED RESULT

Step 1

Go to Dashboard → Services and restart the SIP Server

The 3CX SIP Server and any associated services will stop and then start automatically

Step 2

Wait for the services to start, then go to “Voice & Chat” in the Admin Console and check the registration status of the SIP Trunk

The SIP Trunk after the services have restarted should be seen as “Registered” with a green status

Step 3

Check the capture file and ensure that 3CX sends the “rinstance” parameter in the “Contact” header of the REGISTER message with a random value.

On each new registration 3CX should send the “rinstance” parameter with a new random value.

Step 4

In the 200 OK from the provider check if the “Contact” header contains the “rinstance” parameter with the same value sent in Step 3

The expected behavior is that the SIP Trunk provider will store the “rinstance” value and send it back in the 200 OK.

DESCRIPTION

EXPECTED RESULT

Step 1

Make an Inbound call to the main trunk number of the sip trunk. Check if the “rinstance” value of the register message is included in the Request Line URI on an Inbound call Invite message.

The ”rinstance”value is included in the incoming invite message; it should be part of the Request Line URI sip field.

Step 2

Make sure that the “rinstance” value matches the one sent in the last registration message from the PBX. If “rinstance” is not present in the Invite you can skip this step.

The “rinstance”value included in the incoming invite message should be the same as the one sent by the PBX in the last Register message.

Step 3

Make an Inbound call to the second DID number of the sip trunk. Check if the “rinstance” value of the register message is included in the Request Line URI on an Inbound call Invite message.

The ”rinstance”value is included in the incoming invite message; it should be part of the Request Line URI sip field.

Step 4

Make sure that the “rinstance” value matches the one sent in the last registration message from the PBX. If “rinstance” is not present in the Invite you can skip this step.

The “rinstance”value included in the incoming invite message should be the same as the one sent by the PBX in the last Register message.

DESCRIPTION

EXPECTED RESULT

Step 1

Route the main trunk number destination to Extension A and make an Inbound call.

Make sure that the call routes correctly in the PBX and that the correct extension is ringing.

Step 2

Answer the call.

Navigate to Dashboard → Active calls and make sure that the call is between the correct extension and SIP trunk.

Step 3

Route the second DID number destination to Extension B and make an Inbound call.

Make sure that the call routes correctly in the PBX and that the correct extension is ringing.

Step 4

Answer the call.

Navigate to Dashboard → Active Calls and make sure that the call is between the correct extension and SIP trunk.

DESCRIPTION

EXPECTED RESULT

Step 1

Make an Inbound call to the trunk from a national number to Extension A.

The call should be established and Extension A should have 2-way audio with the remote party

The call should run for at least 60 seconds

Step 2

Check that Extension A sees the correct CLI

Extension A should see the caller ID of the remote party.

Step 3

Extension A hangs up the call

Make sure that the call is terminated correctly on both ends

Step 4

Make an Inbound call to the trunk from an international number to Extension A.

The call should be established and Extension A should have 2-way audio with the remote party

The call should run for at least 60 seconds

Step 5

Check that Extension A sees the correct CLI

Extension A should see the caller ID of the remote party.

Step 6

Extension A hangs up the call

Make sure that the call is terminated correctly on both ends

DESCRIPTION

EXPECTED RESULT

Step 1

Establish an inbound Call with Extension A from an external number by calling the Main Trunk Number (DID A).

The inbound call should be established and Extension A should have 2-way audio with the remote party

Step 2

Check what CLI Extension A sees

Extension A should see the correct CLI

Step 3

Leave the call active for at least 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio after at least 60 seconds

Step 4

Extension A places the call on hold

The remote party should hear the MoH configured in 3CX.

Step 5

Wait 60 seconds

The remote party should still hear MoH after 60 seconds

Step 6

Extension A retrieves call from on-hold

The remote party should stop hearing the MoH and 2-way audio is possible between Extension A and the remote party

Step 7

Leave the call active for another 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio

Step 8

Extension A hangs up the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

Establish an inbound Call with Extension A from an external number  by calling a DID number Number (DID B).

The inbound call should be established and Extension A should have 2-way audio with the remote party

Step 2

Check what CLI Extension A sees

Extension A should see the correct CLI

Step 3

Leave the call active for at least 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio after at least 60 seconds

Step 4

The Remote party places the call on hold

Extension A should hear MoH (either that of the Local PBX of the Remote PBX).

Step 5

Wait 60 seconds

Extension A should still hear MoH after 60 seconds

Step 6

The Remote party retrieves call from on-hold

Extension A should stop hearing the MoH and 2-way audio is possible between Extension A and the remote party

Step 7

Leave the call active for another 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio

Step 8

The Remote party hangs up the call

3CX should receive a BYE message and an appropriate response should be sent to the Provider. Both phones should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

Establish an inbound Call with Extension A from an external number.

The inbound call should be established and Extension A should have 2-way audio with the remote party

Step 2

Edit the Trunk and add a DID number. Press Ok to save changes

The trunk should re-register but the call should continue for at least another 60 seconds.

Step 3

Extension A hangs up the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

Create a Digital receptionist and assign extension A on Option 0. Route the main trunk number to the Digital receptionist. Establish an inbound call to the Digital receptionist from an external number.

The inbound call should be established and the external party should hear the Digital receptionists Music on hold.

Step 2

On the device on the external party press 0

The Digital receptionist should recognise the DTMF tone and route the call to Extension A.

Step 3

Extension A Answers call

The inbound call should be established and Extension A should have 2-way audio with the remote party. The call should be established for at least 60 seconds

Step 4

Extension A terminates the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

Establish an inbound call to the Digital receptionist from an external number.

The inbound call should be established and the external party should hear the Digital receptionists Music on hold.

Step 2

On the device on the external party dial Extensions A number (e.g 000). 

The Digital receptionist should recognise the DTMF tones and route the call to Extension A.

Step 3

Extension A Answers call

The inbound call should be established and Extension A should have 2-way audio with the remote party. The call should be established for at least 60 seconds

Step 4

The remote party terminates the call

3CX should receive a BYE message and an appropriate response should be sent to the Provider. Both phones should hang up

Step 5

Using wireshark identify the DTMF method used.

Identify if RFC2833 or inband is used.

DESCRIPTION

EXPECTED RESULT

Step 1

Create a Queue and assign extension A a queue agent. Route the main trunk number to the Queue. Establish an inbound call to the Queue from an external number.

The inbound call should be established and the external party should hear the Queues Music on hold.

Step 2

Check what CLI Extension A sees

Extension A should see the Queue name and the correct CLI.

Step 3

Extension A Answers call

The inbound call should be established and Extension A should have 2-way audio with the remote party. The call should be established for at least 60 seconds

Step 4

Extension A terminates the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up.

DESCRIPTION

EXPECTED RESULT

Step 1

Create a Ring Group and assign extension A as a Ring Group agent. Route the main trunk number to the Ring Group. Establish an inbound call to the Ring Group from an external number.

The inbound call should arrive in the Ring Group and the Ring Group agents should be polled.

Step 2

Check what CLI Extension A sees

Extension A should see the Ring Group name and the correct CLI.

Step 3

Extension A Answers call

The inbound call should be established and Extension A should have 2-way audio with the remote party. The call should be established for at least 60 seconds

Step 4

Extension A terminates the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up.

DESCRIPTION

EXPECTED RESULT

Step 1

Route the main trunk number to voicemail of extension A. Establish an inbound call to the voicemail service from an external number.

The inbound call should be established and the external party should hear the voicemail prompt for Extension A

Step 2

Follow the prompt and leave a message

Make sure that the message was left and the call was successfully disconnected.

Step 3

Extension A retrieves the message

Make sure that Extension A can listen to the message and that the quality of the message is good.

DESCRIPTION

EXPECTED RESULT

Step 1

Route the main trunk number to Extension A.

Ensure that extensions A forwarding rules for the “Available” status are all set to “End Call”.

Establish an Inbound call to the main trunk number.

Call should ring Extension A and Extension A sees the correct CLI.

Step 2

Reject the call without answering

Make sure that the call is rejected and a “486 Busy Here” is generated from the PBX. The call should be terminated for the external party as well

DESCRIPTION

EXPECTED RESULT

Step 1

Route the main trunk number to Extension A.

Ensure that extensions A forwarding rules for the “Away” status are all set to “End Call”. Switch Extensions A status to “Away”.

Make sure that everything is setup correctly and that routing works.

Step 2

Establish an Inbound call to the main trunk number.

Make sure that the call is rejected by the PBX without polling Extension A and a “486 Busy Here” is generated from the PBX. The call should be terminated for the external party as well

DESCRIPTION

EXPECTED RESULT

Step 1

Navigate to Settings → Blacklisted Numbers and add the external party’s number into the Blacklist

Make sure that number is saved in the correct format in the Blacklisted Numbers

Establish an Inbound call to the main trunk number.

Make sure that the call is rejected by the PBX without polling Extension A and a “603 Decline” is generated from the PBX. The call should be terminated for the external party as well.

DESCRIPTION

EXPECTED RESULT

Step 1

Route the main trunk number to Extension A and ensure that:

•  extension A is set to the “Available” status.
• The callers number is not in the numbers blacklist
• Under the Sip trunk settings → General the option “Number of SIM Calls” is set to 1. 

Establish an Inbound call to the main trunk number.

The inbound call should be established and Extension A should have 2-way audio with the remote party.

Step 2

Make another call to the main trunk number

Make sure that the call is rejected and a “600 Busy Everywhere” is generated from the PBX. The call should be terminated for the external party as well

Step 3

Extension A terminates the established call.

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up.

DESCRIPTION

EXPECTED RESULT

Step 1

Make an inbound call from an external party to Extension A and make sure the call is established.

The inbound call should be established and Extension A should have 2-way audio with the remote party.

Step 2

Leave the call active for at least 1 hour

The call should be left active for at least 1 hour and should not drop. Also 2-way audio should be possible throughout the whole duration.

Step 3

Extension A hangs up the call.

The call should terminate on the remote party

Step 4

Check the capture file if any “keep alive” method was used (e.g. OPTIONS messages, session-timer, etc)

State in the note if any SIP method was used by either 3CX or the Provider. Examples of such could be either OPTIONS messages at certain intervals of a session-timer (RFC 4028)

DESCRIPTION

EXPECTED RESULT

Step 1

In 3CX under “Voice & Chat”, edit each SIP Trunk and in the “General” tab configure so that:

• Calls to the main number of SIP Trunk A is routed to Extension A
• Calls to the main number of SIP Trunk B are routed to Extension B.

Both SIP Trunks bust have different main trunk numbers configured.

Also ensure that in “Inbound Rules” report, there is no entries at all

This setup will test the “Catch All” routing ability when multiple SIP Trunks from the same provider exist

Step 2

From an external source, make a call to SIP Trunk A main trunk number and answer

Call should ring Extension A and the call should be able to be answered with 2-way audio.

Step 3

While call is active, go to Dashboard → “ActiveCalls” and check if 3CX identifies the incoming as coming from SIP Trunk A correctly

This ensures that 3CX is able to distinguish calls arriving for SIP Trunk A and associate them correctly.

Step 4

Hang up the call on SIP Trunk A

Step 5

From an external source, make a call to SIP Trunk B main trunk number and answer

Call should ring Extension B and the call should be able to be answered with 2-way audio.

Step 6

While call is active, go to Dashboard → Active Calls” and check if 3CX identifies the incoming as coming from SIP Trunk B correctly

This ensures that 3CX is able to distinguish calls arriving for SIP Trunk B and associate them correctly.

DESCRIPTION

EXPECTED RESULT

Step 1

In 3CX under “SIP Trunks”, edit each SIP Trunk and ensure that each one, in the “DIDs” tab, has at least 1 DID other than the main trunk number.

The DIDs must be different on SIP Trunk A and B.

• Assign a DID of SIP Trunk A which is not the main trunk number and route calls to Extension A
• Assign a DID of SIP Trunk B which is not the main trunk number and route calls to Extension B

This setup will test the Inbound Rule routing ability when multiple SIP Trunks from the same provider exist

Step 2

From an external source, make a call to the DID of SIP Trunk A and answer

Call should ring Extension A and the call should be able to be answered with 2-way audio.

Step 3

While call is active, go to Dashboard → “Active Calls” and check if 3CX identifies the incoming as coming from SIP Trunk A correctly

This ensures that 3CX is able to distinguish calls arriving for SIP Trunk A and associate them correctly.

Step 4

Hang up the call on SIP Trunk A

Step 5

From an external source, make a call to the DID of SIP Trunk B and answer

Call should ring Extension B and the call should be able to be answered with 2-way audio.

Step 6

While call is active, go to Dashboard → Active Calls” and check if 3CX identifies the incoming as coming from SIP Trunk B correctly

This ensures that 3CX is able to distinguish calls arriving for SIP Trunk B and associate them correctly.

DESCRIPTION

EXPECTED RESULT

Step 1

In 3CX under “Voice & Chat”, edit each SIP Trunk A and ensure that in the “DIDs” tab, it has at least 1 DID other than the main trunk number.

Edit SIP Trunk B and in the “DIDs” tab, add the DID of SIP Trunk A (do so even if it is not associated with this account).

In the “General” tab of both SIP Trunks configure so that:

• Calls to the main number of SIP Trunk A are  routed to Extension A
• Calls to the main number of SIP Trunk B are routed to Extension B.

Also ensure that in “Inbound Rules” report, there is no entries at all

This setup will test the Inbound Rule routing ability when multiple SIP Trunks from the same provider exist and there are common DIDs between multiple SIP Trunks in 3CX.

Step 2

From an external source, make a call to the DID of SIP Trunk A and answer

Call should ring Extension A and the call should be able to be answered with 2-way audio.

Step 3

While call is active, go to Dashboard → “Active Calls” and check if 3CX identifies the incoming as coming from SIP Trunk A correctly

This ensures that 3CX is able to distinguish calls arriving for SIP Trunk A and associate them correctly.

Step 4

Hang up the call

DESCRIPTION

EXPECTED RESULT

Step 1

In 3CX under“Voice & Chat”, edit each SIP Trunk B and ensure that in the “DIDs” tab, it has at least 1 DID other than the main trunk number.

Edit SIP Trunk A and in the “DIDs” tab, add the DID of SIP Trunk B (do so even if it is not associated with this account).

In the “General” tab of both SIP Trunks configure so that:

• Calls to the main number of SIP Trunk A are routed to Extension A
• Calls to the main number of SIP Trunk B are routed to Extension B.

Also ensure that in “Inbound Rules” report, there is no entries at all

This setup will test the Inbound Rule routing ability when multiple SIP Trunks from the same provider exist and there are common DIDs between multiple SIP Trunks in 3CX.

Step 2

From an external source, make a call to the DID of SIP Trunk A and answer

Call should ring Extension B and the call should be able to be answered with 2-way audio.

Step 3

While call is active, go to Dashboard → “Active Calls” and check if 3CX identifies the incoming as coming from SIP Trunk A correctly

This ensures that 3CX is able to distinguish calls arriving for SIP Trunk B and associate them correctly.

Step 4

Hang up the call

DESCRIPTION

EXPECTED RESULT

Step 1

Establish an Outbound Call from Extension A registered on 3CX to an external number.

The outbound call should be established and Extension A should have 2-way audio with the remote party

Step 2

Check what CLI the remote party sees

The remote party should see the Main Trunk number as the CLI

Step 3

Leave the call active for at least 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio after at least 60 seconds

Step 4

Extension A places the call on hold

The remote party should hear the MoH configured in 3CX.

Step 5

Wait 60 seconds

The remote party should still hear MoH after 60 seconds

Step 6

Extension A retrieves call from on-hold

The remote party should stop hearing the MoH and 2-way audio is possible between Extension A and the remote party

Step 7

Leave the call active for another 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio

Step 8

Extension A hangs up the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

Establish an Outbound Call from Extension A registered on 3CX to an external number.

The outbound call should be established and Extension A should have 2-way audio with the remote party

Step 2

Check what CLI the remote party sees

The remote party should see the Main Trunk number as the CLI

Step 3

Leave the call active for at least 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio after at least seconds

Step 4

Remote party places the call on hold

Extension A should hear the MoH

Step 5

Wait 60 seconds

Extension A should still hear MoH after 60 seconds

Step 6

Remote party retrieves call from on-hold

Extension A should stop hearing the MoH and 2-way audio is possible between Extension A and the remote party

Step 7

Leave the call active for another 60 seconds and ensure 2-way audio remains

The call should still have 2-way audio

Step 8

Remote party hangs up the call

3CX should receive a BYE message and should send an appropriate response. Also Extension A phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

In the SIP Trunk settings → “Options” tab → “Caller ID Control”, set the Caller ID to be one of the DIDs you have associated to this trunk (must not be the main trunk number)

Step 2

Make an outgoing call from Extension A to an external number

The remote party should ring and see the DID number set as CLI.

Step 3

Remote party answers the call and 2-way audio is established

The call should be established and 2-way audio possible

Step 4

Extension A hangs up the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

In the SIP Trunk settings → “Options” tab → “Caller ID Control”, set the Caller ID to be a national number that is not associated to this trunk

Enter a national number that is not associated with the SIP Trunk account being used

Step 2

Make an outgoing call from Extension A to an external number

The remote party should ring and see the number set in Step 1 as CLI.

Step 3

Remote party answers the call and 2-way audio is established

The call should be established and 2-way audio possible

Step 4

Extension A hangs up the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

Make an outgoing call from Extension A to an external number using the Anonymous dial code (default *5)

Make an anonymous outbound call using the 3CX dial code.

If using the default dial code, if the external number is 123456789, on the extension dial:

*5123456789

Step 2

Remote party phone rings

The remote party should ring and should not display any number as CLI.

Step 3

Remote party answers the call and 2-way audio is established

The call should be established and 2-way audio possible

Step 4

Extension A hangs up the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

In the SIP Trunk settings → “Options” tab → “Caller ID Control”, set the Caller ID to be one of the DIDs you have associated to this trunk (must not be the main trunk number)

Step 2

Make an outgoing call from Extension A to an external international number

The remote party should ring and see the DID number set as CLI.

Step 3

Remote party answers the call and 2-way audio is established

The call should be established and 2-way audio possible

Step 4

Extension A hangs up the call

3CX should send a BYE message and an appropriate response should be sent by the Provider. Also the remote partys phone should hang up

DESCRIPTION

EXPECTED RESULT

Step 1

Make an outgoing call from Extension A to an external number thats has an IVR that answers

Call a number that answers with an IVR.

This can be a different 3CX system registered with a different account.

Step 2

Ensure Extension A hears the IVR greeting

Extension A should be able to hear the IVR greeting

Step 3

Extension A enter a valid IVR option (single digit)

Extension A sends a DTMF tone which is also sent over the call through the SIP Trunk provider

Step 4

Validate that call is routed correctly based on option pressed

The call  must be routed to the destination based on the option set. This ensures that the remote system “understood” the DTMF tone correctly

Step 5

In the capture file, identify which DTMF method was used by 3CX to send the DTMF tones (RFC2833 or In-band)

In the notes of the test results, state which DTMF method was used to transmit the DTMF tones

DESCRIPTION

EXPECTED RESULT

Step 1

Make an outgoing call from Extension A to an external number thats has an IVR that answers

Call a number that answers with an IVR.

This can be a different 3CX system registered with a different account.

Step 2

Ensure Extension A hears the IVR greeting

Extension A should be able to hear the IVR greeting

Step 3

Extension A enter a valid mutl-digit IVR option, e.g. an extension number of the system (multiple digits)

Extension A sends a DTMF tone which is also sent over the call through the SIP Trunk provider

Step 4

Validate that call is routed correctly based on input

The call must be routed to the destination based on the option set. This ensures that the remote system “understood” the DTMF tones correctly

Step 5

In the capture file, identify which DTMF method was used by 3CX to send the DTMF tones (RFC2833 or In-band)

In the notes of the test results, state which DTMF method was used to transmit the DTMF tones

DESCRIPTION

EXPECTED RESULT

Step 1

Make an outgoing call from Extension A to an external number and make sure the call is established.

The destination number should ring and the remote part should answer the call and 2-way audio should be possible

Step 2

Leave the call active for at least 1 hour

The call should be left active for at least 1 hour and should not drop. Also 2-way audio should be possible throughout the whole duration.

Step 3

Extension A hangs up the call.

The call should terminate on the remote party

Step 4

Check the capture file if any “keep alive” method was used (e.g. OPTIONS messages, session-timer, etc)

State in the note if any SIP method was used by either 3CX or the Provider. Examples of such could be either OPTIONS messages at certain intervals of a session-timer (RFC 4028)

DESCRIPTION

EXPECTED RESULT

Step 1

Create a Fax Forward by navigating to Advanced → Fax → Add → Add Fax Forward and route the main trunk number to send FAX to the email of Extension A. Click OK to save the changes. Then send 10 pages of FAX towards the main trunk number. 

Ensure that the call is answered by the 3CX Fax server and that the 2 parties communicate using T.38. Upon completion the FAX should be sent as pdf to Extensions A email address.

DESCRIPTION

EXPECTED RESULT

Step 1

Using the configured supported ATA device send 10 pages FAX to a remote party

Ensure that the call is answered by the remote party and T.38 is negotiated correctly. All 10 pages must be sent successfully and received by the remote party.

TEXT CASE ID

TITLE

IMPORTANCE

1.1

DNS Lookup

Mandatory

2.1

Basic Registration

Mandatory

2.2

Maintaining Registration

Mandatory

2.3

Releasing the Binding and re-Registering #1

Mandatory

2.4

Releasing the Binding and re-Registering #2

Mandatory

2.5

The correct Transport is Used

Mandatory

2.6

SIP Trunk provider returns “rinstance” value

Optional

3.1

“rinstance” Support

Optional

3.2

Call Routing

Mandatory

3.3

Basic Call Functionality

Mandatory

3.4

Basic inbound Call #1

Mandatory

3.5

Basic inbound Call #2

Mandatory

3.6

Call Continuation

Mandatory

3.7

Inbound call to Digital Receptionist

Mandatory

3.8

Inbound call to Digital Receptionist #2

Mandatory

3.9

Inbound call to Queue

Mandatory

3.10

Inbound call to Ring Group

Mandatory

3.11

Inbound call to Voicemail

Mandatory

3.12

Inbound call - Rejection test - Call Reject by phone

Mandatory

3.13

Inbound call - Rejection test - Call Reject by Forwarding Rule

Mandatory

3.14

Inbound call - Rejection test - Call Reject by Number Blacklist

Mandatory

3.15

Inbound call - Rejection test - Call Reject by Maximum simultaneous calls

Mandatory

3.16

Inbound Long Duration Call

Mandatory

4.1

Basic Inbound Call #1

Optional

4.2

Basic Inbound Call #2

Optional

4.3

Basic Inbound Call #3

Optional

4.4

Basic Inbound Call #4

Optional

5.1

Basic Outbound Call #1

Mandatory

5.2

Basic Outbound Call #2

Mandatory

5.3

Presenting a DID as CLI

Mandatory

5.4

Presenting a Number as CLI that is not a DID (Clip No Screening)

Optional

5.5

Outbound Calls with Withheld CLI (Anonymous)

Optional

5.6

CLI Presented Correctly when calling International Numbers

Mandatory

5.7

DTMF Tones / Traversing IVRs #1

Mandatory

5.8

DTMF Tones / Traversing IVRs #2

Mandatory

5.9

Outbound Long Duration Call

Mandatory

6.1

Inbound FAX

Optional

6.2

Outbound FAX

Optional