Devices

The Devices screen is where all SIP phones, Trunks, and Spans to the Series2 cabinet are managed.

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We have several videos on our YouTube channel (Phonesuite ) that show how to setup SLA with several different types of phones as well as its usage with those phones.

Shared Line Appearance or SLA is a method whereby a user can place a caller on hold and have that call pickup at another phone without the need to transfer the call. This simulates the older key system functionality where all trunk lines were visible to every user. We can simulate this using the Parking and Directed Call Pickup features. Doing this allows phones such as Yeastar to have the SLA feature, something not available previously. Below is a list of each phone brand tested and a quick summary of the required setup. Note we have on our YouTube channel videos on the setup of the new SLA method and we encourage you to view the videos there.

General Setup

• Setup *97XXX or *97XXXX as Directed Pickup Prefix (Included in the factory default settings)

• Ensure that parking is enabled (Dial Plan, Sidebar), normally set for 9000 to park and 9001-9020 parking spots

Polycom

• In Endpoints setup the Polycom with a fast transfer key to 9000 and custom speed dial keys to 9001 and 9002 with BLF enabled. Also add at least one Speed Dial Keys to another user.

  • Note, the custom speed dial keys can be labeled whatever makes most sense to you users (i.e. Line 1, Line 2) and you can also add as many parking spots as required.

• Reboot the phone and have it auto configure from Voiceware (see the auto configuration manual for more information)

• Afterword’s the BLF key to another user will act to call that user if their phone is idle and able to pickup a call from that users phone if its ringing. The fast transfer key will park the call and the BLF keys for parking spots 1 and 2 will light and allow that call to be retrieved.

Vtech

• In Endpoints setup the Vtech with the Add Park Key checked and custom speed dial keys to 9001 and 9002 with BLF enabled. Also add at least one Speed Dial Keys to another user.

  • Note, the custom speed dial keys can be labeled whatever makes most sense to you users (i.e. Line 1, Line 2) and you can also add as many parking spots as required.

• Reboot the phone and have it auto configure from Voiceware (see the auto configuration manual for more information)

• Afterword’s the BLF key to another user will act to call that user if their phone is idle and able to pickup a call from that users phone if its ringing. The fast transfer key will park the call and the BLF keys for parking spots 1 and 2 will light and allow that call to be retrieved.

Yealink

• In Endpoints setup the Yealink with custom speed dial keys to 9001, 9002, and 9000 with BLF enabled for 9001 and 9002. Also add at least one Speed Dial Keys to another user.

  • Note, you can add as many parking spots as required.

• Reboot the phone and have it auto configure from Voiceware (see the auto configuration manual for more information)

• Afterword’s the BLF key to another user will act to call that user if their phone is idle and able to pickup a call from that users phone if its ringing. The fast transfer key will park the call and the BLF keys for parking spots 1 and 2 will light and allow that call to be retrieved.

Each device can select from many available codecs. Here is what each codec does. Not all phones support all codecs. See the phone or trunk providers documentation for further information.

• Ulaw: G7.11 as used in the USA

• Alaw: G7.11 as used in Europe

• GSM: a cellular phone system standard popular outside the U.S.

• G722: a high bit rate (48/56/64Kbps) codec

• Adpcm: Adaptive differential pulse-code modulation (audio)

• Slin: 16-bit Signed Linear PCM (audio)

• Lpc10: AKA FIPS 137 audio only codec

• Speex: audio codec

• Ilbc: Internet Low Bitrate Codec (audio)

• JPEG & png: Video codecs for video calling

• h261, h263, h263p, h264: Video codecs for video calling

• G723: Audio codec

• G726: audio codec, requires an external driver

• G729: narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds

Note the Mean Opinion Score is an average rating of users opinion of the call quality. Score is out of 5 where 5 is the best. More info found here.

This is normally the link to the Series2 cabinet or (uncommonly) a PRI connected directly to the Voiceware server. Below is a screen shot of how a normal Series2 cabinet span is setup.

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Note that each Series2 cabinet has a static MAC address. For cabinet one the MAC ends in 08, for cabinet two it ends in 09, for cabinet three it ends in 0a, etc.

• 003018a4c908

• 003018a4c909

• 003018a4c90a

Certificates are automatically generated starting in Voiceware 4.3 and do not need to be manually generated. If you need to create a certificate below is the process for doing so.

1. Click Generate CA

   1. Note the organization name, unit name, and common name can be anything, use “Phonesuite” if in doubt. The email field also does not need to be valid.

2. Click generate CA

3. Wait for 15 seconds after the next minute (system time) to allow the system to process the new CA

4. Click generate certificate

   1. Choose the CA previously created at the top, the organization name and email can be the same as for the CA. The Common Name should be the systems IP or FQDN, if needed additional FQDNs and IP can be listed together using a comma to separate (no space). Format is “IP:1.2.3.4,DNS:example.com”

5. Click Generate Certificate

6. Next click the new key, cert and click Activate SIP TLS Key

7. Wait one minute before testing the phone in the Browser Console

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Select from the drop down the type of device that is being added (SIP, IAX2, Local, Dahdi). If the Device type is unknown consult the device’s user’s manual or tech support.

The name in this field is how this device will be displayed in the system and in any dropdowns that contain devices. All devices, except specialized equipment like analog gateways, should use their extension number as the Device Name.

This allows the device to be assigned to either a User or Room that has already been added in the Users or Rooms section. If the User or Room has not yet been entered, leave this unselected and assign this device to the User or Room when that record is added. A device acting as a trunk should also be left unassigned to anyone.

This is a description of the device and should reflect whether the device is a guest room phone, front desk device, or some other location. It is recommended not only that this field is used, but that the data entered here follow a schema and be understandable. Data in this field can be helpful for troubleshooting, and a little time spent now will save lots of time later. We recommend always using “Room [#]” or “Rm [#]” for guest rooms. This is also the device name displayed on other devices when applicable (SIP to SIP calling for example) so username or department would be appropriate.

Enter the location of the room in this field. This information is passed to the 911 operator along with the address of the building. This field has a 60-character limit. Note the dispatchable location for a room overrides a dispatchable location for a device assigned to a room.

An optional notes field, has no system effect.

This field allows the setting of a password and a password is randomly generated every time a new device is added. The password created is strong and it is recommended that it be used with all new SIP and IAX2 devices. DO NOT EVER MAKE THE SECRET THE SAME AS THE EXTENSION NUMBER.

This is the number of lines the device has. Most SIP devices have two lines and most normal hotel phones have one. Consult the devices user’s manual for information about how many lines it has, however we strongly recommend using “2” for all guest rooms.

SLA stands for Shared Line Appearance. If the phone is part of an SLA group it can be selected here. Creating groups and general SLA information is found in the section on SLA and in the SLA Group sidebar description below.

In most cases this is the same as the device name above, it should always be the extension number.

This dropdown allows the selection of the type of authentication that the system will attempt to use on calls coming from other IAX2 type devices in the system. Unless there is a good reason to change this leave it as plaintext (the default).

This feature is not currently in use.

This feature is not currently in use.

This feature is not currently in use.

This feature is not currently in use.

This feature is not currently in use.

This feature is not currently in use.

This should match the Device Name and should normally be the extension number.

This field allows the setting of a specific name that the system will use when trying to call the device that’s different then the user and device name above. In most cases leave this field blank.

Used only in rare cases where the SIP provider requests that a special domain be used. In most cases this would be an IP address.

In almost all cases leave this selection on Friend. A User is configured to enter a dial plan (can only send calls) and a Peer is for calls leaving the dial plan (only receives calls), unless there is a good reason to limit the device leave the type set to Friend.

In most cases leave this drop-down set to “no”. If “port” is selected then the system will ignore the port number the request came from, i.e. allow matching of peer by IP address without matching port number. If “invite” is selected it will not require authentication of incoming invites. Modification of this setting is most commonly needed for SIP trunks and in most cases is set to Invite.

DTMF stands for Dual-tone multi-frequency signaling, this is what the system expects in terms of touch-tone format (the sounds that numbers on a phone make when pressed). By default this is set to “RFC2833” which should work fine for most SIP phones, if problems with a device are encountered change this to “Auto” which will allow the system to pick the best method.

Checking this box will compensate for dial tones coming from an older Asterisk system, in almost all cases this option should be left unchecked unless the property reports callers having problems entering key press information (for IVR systems for example).

These three boxes control the “revers” caller ID information. If the device being created is a phone these boxes should be checked. Doing so will display the called stations information (i.e. room number, guest name, call group or IVR name, etc…) when a call to that destination is made. However if the device being created is a SIP trunk this setting will often cause it to fail, thus it should be disabled.

If enabled Voiceware will attempt to deliver the call to the phone and have it answered without the user picking up the handset or pressing the speaker phone button (internal calls only).

This will allow SIP phones to pick up a call ringing at a different location by picking up the phone and pressing *8. For example if the device at the front desk, restaurant, and pool are all set up under the same pickup group (1 for example) then if the device at the front desk rings it can be answered in the restaurant by picking up the phone and dialing *8. Using this feature requires a person in the restaurant to be able to hear the front desk device ring. Assigning devices to the same pickup group WILL NOT cause them to ring together, that can be done using a Call Group (see the Call Groups chapter for more information). Directed call pickup is also an option (see the Dial Plan for more information).

Not used.

Selects if voice traffic will use UDP (default) or TCP, or both.

By default this is set to “Dynamic”, if the device has a static IP then it can be entered here. If left as “Dynamic” (recommended) then the system will keep track of the device’s IP address. If using SIP trunking this should be the address of the server designated by the Internet Telephony Service Provider (ITSP).

Use this field to set a UDP (User Datagram Protocol) port, if left blank the system will use the default port of 5060. Unless there is a very good reason to change this simply leave it as-is.

NAT Stands for Network Address Translation, this allows multiple internal devices to share an external IP address, as long as all the devices are on the same network (all on the same property). This setting is most commonly used for SIP trunks or phones that are offsite (or for all devices on a hosted system).

If checked the device will tell the carrier where it is in order to allow incoming calls from the ITSP. Leave unchecked unless you are using SIP trunking. Checking this box on will also enable 3 additional fields farther down in the page: Registration Username, Registration Password, and Registration Server, as a convenience there is a “Same as Above” link that will copy these values from the corresponding settings that were entered earlier on the page.

Checking this box will have the system check to make sure this device is online once every 60 seconds by default or whatever interval is chosen in the field below. In most cases it is best to leave this box checked. If a non-Polycom phone is able to make calls but not receive them unchecking this box may resolve the issue (Note Yealink brand phones on older firmware, and those who use their software, are known to have this issue).

This is the number of seconds between each qualify request (ping) if the qualify box above is checked.

This section defines what audio codecs will be expected and accepted. For US and Canada installs “Ulaw” (US & Japan), “Alaw” (Europe and almost everywhere else) and “gsm” (cell standard used outside the US) are all that are needed. If you are outside the US or Canada setting “Alaw” first will likely be best. For additional help with codecs consult local standards guides or visit http://www.voip-info.org/wiki/view/Asterisk+codecs .

• Ulaw: G7.11 as used in the USA

• Alaw: G7.11 as used in Europe

• GSM: a cellular phone system standard popular outside the USA

• G7.22: a high bit rate (48/56/64Kbps) codec

• Adpcm: Adaptive differential pulse-code modulation (audio)

• Slin: 16-bit Signed Linear PCM (audio)

• Lpc10: AKA FIPS 137 audio only codec

• Speex: audio codec

• Ilbc: Internet Low Bitrate Codec (audio)

• JPEG & png: Video codecs for video calling

• h261, h263, h263p, h264: Video codecs for video calling

• G723: Audio codec

• G726: audio codec, requires an external driver

• G729: narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds

This drop-down allows selection between any extensions on the Voiceware server. Leave set on the default of “Phone (all_calls)”, devices in different contexts will not be able to call each other. This setting automatically changes to “Carrier (in_pri)” when Usable as Trunk is checked (see below).

Used only when a custom context is selected from the drop-down box in the field above.

In most cases this box is left unchecked unless you are setting up a SIP trunk. Checking this box will make this device selectable in the Outbound Routes trunk drop-down.

Here an external number can be entered (i.e. 3035551234) for this trunk to test against. The number listed here must be registered to another Voiceware system and must be setup with a dial plan of “TrunkTestReceiver” (see Dial Plan on page 75 for more information). A test will then occur daily at the time specified on the settings page (see “Use Trunk Tester to ensure calls…” page 14) and will test both DTMF tones and spoken words. If this test fails it will attempt the test again after a few minutes and if a second failure is detected the system will email the notification email address listed in the settings page.

If the device is denoted as a trunk (see above) a number of available channels must then be entered. This number should always be 99 to allow for emergency calls to be processes in all cases. If this was set to 10 then Voiceware would reject the 11^th simultaneous call to be placed on a trunk regardless of number the dialed.

Checking this box will add two additional options (see below).

Here an installer can input the customer caller ID number that will be used if anyone in this room were to dial 911. NOTE: We can only pass this information to the telephone provider, we cannot guarantee they use it.

Like above this allows an installer to add a custom caller ID name to be used if anyone in this room were ever to dial 911 (i.e. “2^nd floor, room 204). NOTE: We can only pass this information to the telephone provider, we cannot guarantee they use it.

View Devices:
This link will display the list of devices currently configured in the system, this is the default view of the devices page.

Add Device:
This link is exactly like the “Add Device” button at the top of the page and will allow the addition of a new device.

SLA Groups:
Allows the setup of SLA groups. These groups are simply a name (i.e. Sales).

Hardware Spans:
This allows an installer to set up new hardware spans. Once configured a span will display a status, status elaboration: PHY = Physical Layer (i.e., can we establish a link) LOG = Logical Layer - What asterisk sees. "In Alarm" means physical layer is in alarm. Up/Down shows the D-Channel status within asterisk. Active/Inactive is the supervisory state. It will almost always be Active. Each field is described below.

This is normally left unchecked for PRI, T1/E1 setup. This option is only checked when setting up a connection to the Series 2 cabinet. When checked a new potion appears (see below).

The adaptor dropdown selects what Ethernet port is used for the connection to the Series 2 cabinet. This is normally set to Eth1 not Eth0. The remote MAC is the MAC address of the Series 2 cabinet and is normally the same, 00:30:18:a4:c9:08. In a single box solution the TDM side will pick up this MAC address automatically without the need to set it in the TDM side. Lastly this span is normally preset from the factory and does not require any changes.

Leave Tone Zone set to the default value United States unless connecting to a telco line outside the U.S.

Enter the T1/PRI clocking source, normally set to 1.

This allows selection for the cable length or distance between the PRI port on the server and where the circuit is. This is used to correct for signal loss over long distances. Unless directed otherwise by Voiceware technical support leave this value at the default “0 to 133 ft.” setting.

Used to select the framing and coding protocols. By default ESF (extended super frame) is used for framing and B8Zs is used for coding. With older telco equipment D4 and AMI are sometimes used.

Used to select between T1 (American standard) and E1 (used in Europe and Asia). By default all Digium cards come set to T1. Changing the card to E1 requires a jumper on the card itself to be adjusted, consult Digium support for proper jumper settings.

If connecting to a Phonesuite device select “Phonesuite Cabinet”. For a Telco PRI select “PRI CPE (To CO”, and select “E&M wink” for wink start CAS (aka RBS) T1 circuits. For MFC/R2 (common in Mexico) make sure to select Mexico in Tone Zone and the correct PRI switch type (as instructed by the provider).

Note: that both PRI CPE and PRI Network have “To CO” and “To PBX” options. This allows the connection of Voiceware to a PBX and for calls coming in on this PRI to be treated like internal calls not like calls coming from the CO.

Select from the dropdown the type of PRI switch type is installed, in most cases this is “National ISDN2” but this depends on the provider. For more information contact the PRI provider or see their user’s manual.

Select from the dropdown list the type of echo cancellation (if any) is enabled on this span.

In most cases this is left unchecked. Check only if the system is having problems recognizing key presses on incoming calls.

This selects what channel will be set aside for communication between Voiceware and the Telco. It is always set to 24 for T1. If using E1 the D-channel might be 16, 31, or both.

These are the voice channels that are used to make calls outside the system. For PRI, leave all 23 channels checked unless there is a good reason to have one unavailable for voice traffic (i.e. you are using a partial PRI).

For E&M Wink and other CAS lines, be sure to that all 24 channels are checked on.

When finished setting the options click “Save Span.” You will be presented with a channel configuration page.

If you are satisfied with the configuration be sure to click “RESTART NOW” for the changes to take effect. This will automatically trigger a restart of the entire Voiceware system and interrupt all calls in progress, even internal calls or calls to voicemail.

This area allows spans to be grouped together so that they act as one large span. For example two T1 PRI spans to the Telco can be linked together so that Voiceware treats them as one span for the purposes of routing outbound calls.

Use the dropdown to select the number of the span you wish to group.

Select the group that the above span will become a part of. By default Voiceware places each span into its own group. Up to 62 groups can be defined.

TLS Keys:

TLS Keys are required for the Browser Console SIP phone to function. If these certificates are not generated upon installation, follow the below steps to generate a certificate.

1. Click Generate CA
   - Note the organization name, unit name, and common name can be anything, use “Phonesuite” if in doubt. The email field also does not need to be valid.

2. Click generate CA

3. Wait for 15 seconds after the next minute (system time) to allow the system to process the new CA

4. Click generate certificate
   - Choose the CA previously created at the top, the organization name and email can be the same as for the CA. The Common Name should be the systems IP or FQDN, if needed additional FQDNs and IP can be listed together using a comma to separate (no space). Format is “IP:1.2.3.4,DNS:example.com”

5. Click Generate Certificate

6. Next click the new key, cert and click Activate SIP TLS Key

7. Wait one minute before testing the phone in the Browser Console

Logout:

Here the current user can log out to allow a different user to log in.

Shared Line Appearance or SLA is a method whereby a user can place a caller on hold and have that call pickup at another phone without the need to transfer the call. This simulates the older key system functionality where all trunk lines were visible to every user. We can simulate this using the Parking and Directed Call Pickup features. Doing this allows phones such as Yeastar to have the SLA feature, something not available previously. Below is a list of each phone brand tested and a quick summary of the required setup. Note we have on our YouTube channel videos on the setup of the new SLA method and we encourage you to view the videos there (Polycom, Vtech, Yealink).

General Setup

• Setup *97XXX or *97XXXX as Directed Pickup Prefix (Included in the factory default settings)

• Ensure that parking is enabled (Dial Plan, Sidebar), normally set for 9000 to park and 9001-9020 parking spots

Polycom

• In Endpoints setup the Polycom with a fast transfer key to 9000 and custom speed dial keys to 9001 and 9002 with BLF enabled. Also add at least one Speed Dial Keys to another user.

  • Note, the custom speed dial keys can be labeled whatever makes most sense to you users (i.e. Line 1, Line 2) and you can also add as many parking spots as required.

• Reboot the phone and have it auto configure from Voiceware (see the auto configuration manual for more information)

• Afterword’s the BLF key to another user will act to call that user if their phone is idle and able to pickup a call from that users phone if its ringing. The fast transfer key will park the call and the BLF keys for parking spots 1 and 2 will light and allow that call to be retrieved.
  
  

Vtech
In Endpoints setup the Vtech with the Add Park Key checked and custom speed dial keys to 9001 and 9002 with BLF enabled. Also add at least one Speed Dial Keys to another user.

• Note, the custom speed dial keys can be labeled whatever makes most sense to you users (i.e. Line 1, Line 2) and you can also add as many parking spots as required.

• Reboot the phone and have it auto configure from Voiceware (see the auto configuration manual for more information)

• Afterword’s the BLF key to another user will act to call that user if their phone is idle and able to pickup a call from that users phone if its ringing. The fast transfer key will park the call and the BLF keys for parking spots 1 and 2 will light and allow that call to be retrieved.

Yealink

• In Endpoints setup the Yealink with custom speed dial keys to 9001, 9002, and 9000 with BLF enabled for 9001 and 9002. Also add at least one Speed Dial Keys to another user.

  • Note, you can add as many parking spots as required.

• Reboot the phone and have it auto configure from Voiceware (see the auto configuration manual for more information)

• Afterword’s the BLF key to another user will act to call that user if their phone is idle and able to pickup a call from that users phone if its ringing. The fast transfer key will park the call and the BLF keys for parking spots 1 and 2 will light and allow that call to be retrieved.