Phonesuite Gateway

To find the gateways DHCP IP address dial *11*.

Navigate to that address and log in with “installer” and the default installer passed used on Voiceware. If you don’t have this login email support@phonesuite.com.

An admin level login is available once the gateway auto provisions to Voiceware for the first time, its login is “psadmin” and the default Voiceware installer password with “!” at the end.

The PSG offers zero touch provisioning to Connectware.

1. Add the PSG to the inventory page and either assign ports or enable room walk mode.

Open

PSG Setup in Connectware

2. Plug in the gateway and connect it to the network (ETH1)

The gateway will then provision itself and be ready in about 3-5 minutes. Its normal that during this process the gateway will reboot several times as the gateway will also preform a firmware update.

Provisioning a PSG in Voiceware is very simple and works like many other phones and gateways.

1. Add the PSG to the Endpoints page in Voiceware and assign rooms to ports.

Open

PSG setup in Voiceware

2. Setup option 66 in the local router to “https://<ip>/p”

3. Plug in the gateway and connect it to the network (ETH1)

The gateway will then provision itself and be ready in about 3-5 minutes. Its normal that during this process the gateway will reboot several times as the gateway will also preform a firmware update.

Change Log

Version: 1.1 Dec 26, 2023

• Set the max ring time to 180 seconds

• Set the max wait time for RTP packets to 10 seconds

• Lowered the interdigit timeout (Dial Timeout) to 4 seconds (from 6)

• Disabled the local CDR record store

• Disabled Tr069

• Set the reboot feature code to *732668* (dial the word reboot)

These feature codes are available for Voiceware, Connectware feature codes are marked as such.

• *11*: Play back Eth1 IP

• *12*: Play back Eth2 IP

• *30*: Calls back port once phone is hung up (Connectware)

• *31*: Plays back phone number

• *61*: Set Eth1 IP

• *62*: Set Eth2 IP

• *022*: Play if port is registered

• *70*: Play back what port (80, 443) to access the web GUI on

• *732668*: Reboot gateway (dial star, reboot, star).

• *99*: Manually turn on MWI (Connectware)

• *71*: Plays back last number to call port. (Connectware Only)

Elevator Phone Disconnect

When connecting a PSG to an elevator phone you might have problems with the disconnect after the call is over. To resolve this use the below custom override.

lcfo1="800"

Open

Web GUI Enable

To enable the web GUI and set a password use the below overrides. Note: the value of the password can be set to anything.

Open

Digit Timeout

To adjust the digit timeout use the below overrides as shown above.

Set to 4 Seconds

TimeoutRules0=0 4 4 Voiceware

Set to 2 Seconds

TimeoutRules0=0 2 2 Voiceware

Set to 6 Seconds

TimeoutRules0=0 6 6 Voiceware

Version 3.0 (4-3-0-6)

This version added support for setting ring down with no delay. This version of the template only works in Voiceware version 4.3.0.6 or higher, using this template in older Voiceware versions will not correctly provision the gateway.

This version also updated the format of the config file to the latest supported by the gateway.

This page displays system info such as MAC address, firmware version, and system uptime.

Open

Here a summary of all ports is displayed including their registration status and the ports status (idle, off hook, etc).

Open

Idle

Open

Off Hook

When a call is placed information about it is displayed as below.

Open

Call from room 103 to 888

This page displays statistics on the total number of calls places to or from the gateway. This info is useful for troubleshooting.

Open

This page displays counts of SIP messages, only useful for high level troubleshooting.

Open

The Quick Config page allows network configuration for both active ETH ports. These settings are also available under System Tools → Network.

The gateways SIP server settings can be configured here. Note that only one SIP server can be configured. These settings are set with the auto provision file and don’t commonly need to be changed, do so only if you fully understand their impact.

SIP Address: Which of the two available (ETH1 or ETH2) ports are used to connect to the SIP server (i.e. the phone system). In most cases ETH1 is the primary network connection and thus the port used to connect to the phone system.

SIP Port: 5060 is always used with Phonesuite systems.

Register Status: Should always say “Unregistered”

Register Gateway: Should always be set to “No”. Setting to Yes allows one username and password to be set to register all ports to the same SIP account.

Registrar IP Address: The IP address or FQDN of the phone system. This is set via auto provisioning.

Register Port: Should always be set to 5060

IMS Network: Stands for “IP Multimedia Subsystem” and is not enabled for Phonesuite systems. Enabling this requires entry of a Externally Bound Address and Port.

Spare Register Server: If enabled the gateway will attempt to register to the backup IP or FQDN if the primary one fails. This is not enabled by default. Note that Connectware handles fail-over without using this setting. If enabled enter the IP or FQDN of the backup SIP server address.

Custom Host of From Field: Allows a custom host from name to be entered. This is not required for Phonesuite systems.

DNS-SRV Enable: This enables DNS SRV records. This is enabled when used with a Connectware Hospitality system. Note that Voiceware can not use SRV records.

Register Interval Time (ms): The time in MS in-between registration attempts. For example if this is set to 50 the gateway will register port 1, wait 50ms then register port 2 and so on. This setting is designed to help not overload the PBX with lots of registration attempts at the same time.

Registry Validity Period (s): The time between registration requests when Register Gateway is set to Yes.

Re-registration Interval(s): If registration fails how quickly will it attempt again.

Multi-Registrar Server Mode: If enabled a new option “SIP Server” will appear in the sidebar menu. Here more than one SIP server can be setup and each FXS port can register to any SIP server. Not commonly needed.

SIP Transport Protocol: Normally set to UDP. Can be set to TLS for encrypted SIP so long as the SIP server has been set with the same settings.

Switch Signal Port if SIP Registration Failed: If enabled the gateway will attempt to register to the SIP Signaling port set above. If it fails it will try to register on port+1 (i.e. 5061), this will continue until the gateway registers.

TFTP Auto Update Register Info: Enables the gateway using a local TFTP server to update its configuration information. Not used for Phonesuite systems.

These settings are again set by auto provisioning to correctly work with Phonesuite systems. Adjustment of these settings is not commonly needed and should only be done when the settings, and their impact, is understood. [SIP]

Obtain CalleeID from: By default set to “Request” Field, other options include the “UserName” and “DisPlayName” Field. Some phone systems will alter where they place the caller ID info and thus this setting can be used when incorrect, or no, CID information is displayed.

Set CallerID position: Tells the gateway where to set the Caller ID info, set to “Username of From Field” by default.

Obtain CallerID from: Similar to the setting above this can be changed should the caller ID information not be displaying correctly on inbound calls.

Use Source Address: In SIP packets there is a listed return source address, and an address where the packet was actually received from. This commonly happens in a NAT situation where the gateway and the SIP server are no on the same network. The default is to disable this setting as NAT is handled by the Phonesuite systems.

Use Contact Address: This is the address listed inside a SIP packet for where requests should be returned to. Like the setting above Phonesuite systems handle NAT and this setting is not needed.

Call Transfer Mode & Call Flash Mode: Sets if the gateway attempts to handle these requests itself or if the SIP server (i.e. the Phonesuite systems) handles the request. This should always be set to “Platform to Handel SIP Info” when using Phonesuite systems.

Hold Music Source: Sets where hold music is played from. Always set to “Remote” for Phonesuite systems.

Two Stage Dialing for SIP Incoming Call: Enables a two stage SIP dialing option. Not needed on any Phonesuite system.

Maximum Wait Answer Time (s): Sets the default amount of time an outbound call will wait to be answered by the remote PBX. Note this is not being answered by a person or voicemail, this is the SIP “answer” function.

Set SIP Identifying: Sets the default ID used in SIP packets. Note the gateway will add the last two digits to setting to “Phonesuite PSG-10” will result in “Phonesuite PSG-1024” for a 24 port gateway.

Maximum Wait RTP Time (s): The maximum number of seconds the gateway will wait for RTP packets after setting up a SIP call before automatically ending the call.

Call Abnormal Hangup Detection: Should the remote caller abandon the call (i.e. their phone was unplugged) how long should the gateway wait before ending the call. This feature is not enabled by default because Phonesuite systems detect and end abandoned calls.

Server Status Detection: Similar to the “Qualify” setting in Voiceware this feature sends out a heartbeat packet every 10 seconds by default to ensure the SIP server is online. This setting can be extended only in cases of very large installations where these heartbeat packets are causing network load.

SIP Encryption: If enabled a Encryption type, identifier, and key will need to be provided. Not currently supported on Phonesuite systems.

RTP Encryption: Can be enabled to encrypt RTP packets. Disabled by default.

INVITE 100rel: This feature enabled acknowledgments for SIP setup packets that are not normally acknowledged (like Ringing). This feature is not needed with any Phonesuite system.

Ignore ACK: Enabling this feature will allow the gateway to send a 200OK message on SIP setup without first getting an acknowledgment message. Not needed with any Phonesuite system.

User-defined SIP Code: By default the default SIP codes are used. If checked custom codes can be defined for No Idle Port, Called Party Disconnected, and Route Failed. This is not needed in a Phonesuite system.

Use Iptables: When set (the default) only calls from the register SIP address or those listed in the Access Control list will be allowed to send SIP messages to the gateway. We recommend enabling this security feature in all cases.

Session Refresh: Enabled a session refresh timer. Not used on Phonesuite systems.

Manage Refer: Always set to “Default”. Phonesuite does not use the FXO features of the gateway.

This setting only appears if Multi-Register Server Mode is checked in the SIP section above. Once enabled this page allows up to 8 SIP servers (phone systems) to be defined. These servers can then be selected when editing FXO ports.

Index: Always start at 1 and add servers sequentially.

Description: A human readable description of the server.

Registrar IP Address: The IP address of the server.

Registrar Port: Normally 5060.

Registry Validity Period (s): How long in seconds the registration to this server will be valid for before a re-registration attempt is made. Default 600 seconds (10 minutes).

IMS Network: IMS is a standardized architectural framework for delving voice traffic over different types of networks. Its beyond the scope of this manual to describe IMS in detail but more information can be found here: IP Multimedia Subsystem. If enabled the options Externally Bound Address and Externally Bound Port (5060 by default) become available. This setting is not commonly needed.

Network Address Translation (NAT) are settings designed to help the flow of SIP traffic when the gateway and SIP servers (i.e. phone system) are not on the same network. Generally Phonesuite systems will handle NAT and these settings are not required.

Auto NAT: The three settings are:

• Disable Auto NAT: Default section

• Enable PMP: Port Mapping Protocol (PMP) is a method where by the gateway will attempt to automatically set the port forward and NAT settings without user input and without a STUN server. NAT-PMP runs over the UDP and uses port number 5351.

• Enable UPNP: Universal Plug and Play (UPnP) employs the Simple Service Discovery Protocol (SSDP) for network discovery, which uses UDP port 1900.

STUN Server: A STUN server is an external host that is contacted by the client (in this case the gateway) and the STUN server then reports back to the client what public IP address it was contacted from to help enable NAT. The STUN server is contacted on UDP port 3478 but return traffic will be on a different port and / or IP address.

Mapping Contact IP: The systems Public IP can be manually entered here and thus does not need to be discovered using one of the above methods. This will be added to the Contact field in SIP messages.

Mapping SPD IP: Same as above but for the SPD field in the SIP messages.

Report: This adds a Via field to any SIP messages. Enabled by default.

Learn NAT: This will allow the gateway to use the IP address and port number returned in the Report field (enabled above) to learn NAT settings. Disabled by default.

Auto Detect NAT IP: Same as above but for RTP.

RTP Self-adaption: This feature if enabled will update the IP and/or port number in the SIP signalling message to match the actual location that RTP is being received from. Disabled by default.

These settings control how media (i.e. audio) is passed to and from the gateway. [SystemConfig]

DTMF Transmit Mode: The default setting is RFC2833 but options include SIP INFO and In-band. Normally the default setting will work but some SIP carriers have been known to use SIP INFO or In-band. Refer to the phone system documentation for what method is used. If set incorrectly then DTMF tones will either not be recognized or seen twice by the SIP server.

RFC2833 Payload: The default is 101 with possible rage of 90-127.

RTP Port Range: The port range the gateway will transmit RTP packets in. The gateway and Asterisk default is 10,000-20,000. Setting this incorrectly will lead to no audio, or intermittent no audio on calls.

Silence Suppression: During pauses in a conversation normally silent “comfort” RTP packets are sent. Setting this feature to Enabled will stop the sending of silence packets. Doing this can reduce bandwidth usage but might cause the other end to assume the call has dropped and end the call on their side. If set to enabled test long pauses in conversation (or pressing the mute button) to ensure the call does not drop.

JitterMode: Currently the only option available is Static Mode.

JitterBuffer (ms): A range of 20-280 can be set with 100 being the default. This buffer stores audio to reduce jitter (breaks in the audio of the call) but caused a slight delay from when the caller speaks to when the audio is sent to the phone system. JitterBuffer=-30

Voice Gain Output from IP (dB): A range of -24-24 can be set with 0 being the default. Can be used to increase or decrease volume of audio leaving the gateway.

G729B: Enables the codec G729B. This is enabled by default.

Codec Priority: Sets what codecs are used in what priority. These settings are set by the auto provision function of Phonesuite systems and require no adjustment.

These settings allow for fine-grain control of the FXS ports and their settings. These settings should only be changed if local FXS stations are not functioning correctly. Below is a summary of what each setting does and is not indented as a full description of the setting and all its possible use cases.

Tone Energy (dB): Possible values are between -35 and 15 with the default being -11. Describes how much energy signals are sent with.

Ringing Scheme Settings: Enables distinctive ringing (i.e. a double ring for internal calls). If enabled a field “Ringing Mode” appears where the text passed in the contact header is placed. Voiceware sends a contact header of “internal” for internal calls.

Hook Flash Detection: Allows the analog phones to use hook-flash. Disabled by default. Hook flash is where you can very quickly place the phone on hold and take it back off again to send a “flash” message, this was used in older systems to switch between an active and new inbound call (i.e. call waiting).

Minimum Time Length of On-hook Detection (ms): The minimum amount of time a phone must remain on hook to be considered on hook. Default value is 64..

Preferred 18x Response(NO valid P_Early_Media): Sets if the gateway should play a ring back tone locally. Used on systems that do not provide a ringback tone. Set to IMS Ringback by default.

Enable Press-Key Call-Forward: If enabled then calls can be forwarded by pressing either the # or * keys (as defined). If enabled the forward can be also set as a blind or negotiated forward. For Phonesuite systems this setting is disabled.

CID Transmit Mode: Sets how the caller ID information is transmitted. The default is FSK.

Occasion to Send FSK CallerID: Caller ID info can be sent to the analog phone before or after the first ring with the default being after the first ring.

Send Polarity Reversal Signal: If enabled the gateway will send a polarity reversal signal to the FSX channel when it detects the phone was answered.

Off-hook Dither Signal Duration (ms): This value is how long in milliseconds a station must be off hook in order to be considered off-hook by the gateway. Values must be an integer multiple of 16. The default value is 64 and the lower the value the more sensitive the port becomes.

Handling of Call from Internal Station: This should always be set to “Platform Handling”, thus allowing the phone system to control the call. If set to Internal Handling then station to station calls on the same gateway are not sent to the phone system and thus the phone system can’t control states like calling permission or DND and can’t record the call in the CDR logs. In a hotel environment this is a security and privacy risk.

Light Up Mode for Voice Message: Sets the voltage type to light the lamp on an analog phone for MWI. Set to H by default this setting might need to be adjusted depending on the type of analog phones in use.

Open Session In Advance: If enabled the gateway will replay with SIP message 183 (Session Progress) for inbound calls. Disabled by default.

Report FXS Status: If enabled the gateway will report to the PBX any change is port status (i.e. the port is on a call, or has ended a call). This type of reporting is required to make features like BLF work on admin phones. However its not commonly needed on analog stations. Enable only if a BLF key is setup for an analog station port. Disabled by default.

Enable Send DTMF while receiving 183: SIP status 183 (Session Progress) may be used to send extra information for a call which is still being set up. Default is enabled.

This page is used to set the gateway to detect a busy signal for example for the region where the gateway is used (i.e. the United States). These settings are set by the template file and should not be changed.

Its beyond the scope of this wiki to describe in detail how dial, busy, and fax tones are generated but there are resources to be found online about such topics such as Dial tone.

This page is used to adjust how the gateway generates Dial, Ringback, and Busy tones. These settings are set for United States standards and should not be changed.

Its beyond the scope of this wiki to describe in detail how dial, busy, and fax tones are generated but there are resources to be found online about such topics such as Dial tone.

This page is split between two sections, Tone Detection and Generation. These settings should not be changed unless there is some issue with DTMF tone detection by the phone system.

Energy Difference of High-freq minus Low-freq (dB): Sets the allowed difference of the high frequency energy in the DTMF signal over the low frequency energy. The value range is 0-24, in decibel, and the default value is 5.

Energy Difference of Low-freq minus High-freq (dB): Set the allowed difference of the low frequency energy in the DTMF signal over the high frequency energy. The value range is 0-24, in decibel, and the default value is 9.

Minimum Duration at ON (ms): Sets the minimum duration at On for the DTMF signal. The allowed range is 10-2000 with a default value of 28.

Minimum Duration at OFF (ms): Sets the minimum duration at Off for the DTMF signal. The allowed range is 10-2000 with a default value of 36.

Ratio of DT Energy(%): Set the percentage of energy in the DTMF signal. The value range is 1-100 and the default value is 83.8.

Lowest Energy Threshold (dB): Set the minimum energy threshold of the DTMF signal. The value range is -40 to 9 with a default value of -21.

DTMF Display via Channel Status: Enabling this will display the sent or received DTMF signals when the mouse cursor is hovered over the channel status. Disabled by default.

ABCD Detection: In addition to the normal 0-9 numbers there are also DTMF tones for the letters A, B, C, and D. These are normally not used on modern phones but can be enabled here should they be needed. Disabled by default.

DTMF Energy Advance Set: If enabled then each number digit along with star and pound can have a low and high Hz Energy in dB set. Disabled by default.

DTMF Energy (dB): Energy of the DTMF signal sent by the FXS gateway. Range of value: -18~11, calculated by dB, with the default value of 0.

Duration at ON (ms): Set the duration of the DTMF signal at ON state. Range of value: 0~16383,
calculated by ms, with the default value of 100.

Duration at OFF (ms): Set the duration of the DTMF signal at OFF state. Range of value: 0~16383,
calculated by ms, with the default value of 32.

The gateway can be configured to ring the FXS ports differently based on either the caller ID or Alert-Info in the SIP messages. There are up to four schemes for each. [RingPeriodGroups]

Scheme 1-4

CallerID or Alert-Info Value: An exact match for the caller ID or alert-info value. RingCallerId0=Internal

Ringing Mode: Here we have two options for ringing scheme, a one ring cycle, and a two, or double, ring cycle. RingMode0=2,400,500,400,2000 The syntax is:

• 1,A,B

• 2,A,B,X,Y

The rules for the syntax are:

1. Values are in milliseconds

2. The last value (B or Y) must be at least 1,800

3. No value can be larger than 12,000

4. The total duration for all values can’t exceed 16,000

5. Values are entered without commas

For example Voiceware sends an “Alert-Info Value” of Internal. The Ringing Mode is set to 2,400,500,400,2000 which will cause the phone to ring for 400ms, pause for 500ms, ring again for 400ms, then pause for 2000ms. This results in a "double short ring" to alert the caller that the call is an internal call, not a call from an external caller.

Fax Mode: Options here are T.30 and T.38.

T.30: This is an older protocol for use with traditional analog telephone lines.

T.38: Used when transmitting Fax over IP based connections (i.e. SIP trunks).

It it beyond the scope of this wiki to detail the T.38 protocol but more information can be found here: T.38.

Function keys, sometimes called “Feature Codes” are codes that can be dialed on an analog phone to activate some feature of the gateway or port.

Query LAN1 & 2: Reads back the IP address assigned to LAN port 1 or 2. Default *11* and *12*.

Query Phone Number: Reads back the extension number assigned to the port. Default *20* but set to *31* on Voiceware so as not to conflict with maid status codes.

Phone Test: When this code is entered and the call ended the gateway will call this port back to test two-way calling and the phones ringer. Default *30*. PrefixRules3=3 *30* 0
Set LAN1 & 2: Allows the IP address of ETH1 or 2 to be set. Default *61* and *62*.

Query WEB Port: Plays back the port number used to access the gateways web GUI (i.e. 80 or 443).
Reboot: Reboots the gateway. Set to *732668* when provisioned with Voiceware (dial the word “reboot”). This feature code is disabled when used with Connectware.

Transfer, forward, DND, and Conference codes are all disabled when the gateway is provisioned with Voiceware.

Blind Transfer: Transfers a call to an extension without talking to the receiving party first. Default *010*.
Call Forward Unconditional Activate: Enables always on call forward. Default *030*.
Call Forward Unconditional Deactivate: Disables always on call forward. Default *031*.
Call Forward Busy Activate: Enables forward when the port is already on a call. Default *040*.
Call Forward Busy Deactivate: Disables forward when the port is already on a call. Default *041*.
Call Forward No Reply Activate: Forwards the call on no-answer. *050*
Call Forward No Reply Deactivate: Disable forward on no-answer. *051*
Do Not Disturb Activate: Sets the port into do not disturb. Default *060*.
Do Not Disturb Deactivate: Disables do not disturb for the port. Default *061*.
Conference: Starts a conference call with another extension. Default *070*.

Register: Sets the port to register, this is disabled by default in Voiceware. Default *020*.
Unregister: Sets the port to unregistered, this is disabled by default in Voiceware. Default *021*.
Query Register Status: Will read back if the port is registered. Default *022*.
Query Missed Call Number: Reads back the number of the last call placed to the port. *71*
Manually turn on the warning light: Turns on the message waiting lamp. Default *99*.

Dialing rules used to know when dialing has been completed and to send the call to the PBX. This feature is known as a Digit Map in other phones. These rules can be set in Voiceware using the digit map field in the Endpoints settings. Below is the default digit map we suggest using with Voiceware.

[2-9]11|[8-9][2-9]11|[8-9]0x.|[8-9]1[2-9]xxxxxxxxx|[8-9][2-9]xxxxxxxxx|[1-7]xx.|*2|0|911|*xx|8xx|933

Valid Dialing Rule Characters

0-9: Digits 0 - 9

A-D: Letters A - D. Note that letters A, B, C, and D are supported in the DTMF standard.

x: The letter “x” (lower case) denotes signal numbers or letters A-D. For example 2xx would allow dialing to any three digit extension starting with 2.

.: Zero or more digits or letters A-D. For example 2xx. would match any number starting with 2 and including at least two additional numbers or letters A-D. The gateway would then pause to wait for any additional digits (unlimited length) before sending the call to the PBX.

[ ]: Used to denote a range of numbers or letters A-D. For example [2-4]xx would match 200 or 499, but not 500 or 199. In addition [3,7]xx would match on 300 or 799 but not 450 or 630 for example. Lastly a combination of both dashes and commas can be used, for example [2-3,5,9].

*: The star character only represents * as dialed on a phone, its not used as a wild card.

#: Pound or hash is used to represent # as dialed from a phone.

These simple settings control how long the gateway waits for a digit to be dialed.

Description: A simple human readable description, can be omitted, sets itself to “default”.

Inter Digit Timeout (s): Sets the time in seconds the gateway will wait after the last digit is dialed before sending the call to the PBX. Default is 6 seconds.

Off-hook waiting digit timeout (s): Sets how long the gateway will wait for the first digit to be dialed after the handset is picked up. Default is 6 seconds.

This is a tone or beep that is played to the caller when a call enter call waiting. This feature is not commonly needed.

Allows the upload of a custom ringback file that callers to ports on the gateway will hear when they call a port. There is a default normal ringback tone played to callers so this setting is not commonly needed.

Quality of Service (QoS) is a feature that will place a QoS value on packets. The local router can then be configured to look for and respect QoS. This will reduce the chance that voice packets are dropped and give them a higher priority for processing. This can help improve voice quality in networks with high utilization.

QoS: A check box to enable QoS. Disabled by default.

Medial Premium QoS: Sets the priority level for medial (voice / RTP) packets. Default 46, can be set between 0-63.

Control Premium QoS: Sets the priority level for SIP packets. Default 26, can be set between 0-63.

This setting is used to report off-hook and on-hook events. This feature is not used by Phonesuite.

Channel Pick up: The URL to send channel pickup events to.

Channel Hang up: The URL to send channel hangup events to.

This feature allows locally stored CDR records to be searched.

CDR records include:

• Call Time

• Caller ID

• Dialed Number

• Direction (InBound, OutBound)

• Duration (hh:mm:ss)

• Reason for Call Completion (which side ended the call)

• Port ID

• Gateway IP

• Remote IP (IP of the PBX)

Starting & Ending Date: Set the start and end date for the search.

Port: Records for all ports, or an individual port can be searched.

Call Direction: Can be set to “All”, “Inbound”, or “Outbound” to limit the report as needed.

CallerID: Searches for calls with a matching caller ID.

Call Duration(s): Sets the minimum and maximum call duration for the search.

Keyword: Searches any word (i.e. caller ID name) that is included with the CDR records.

Enable OpenVPN: If set to Yes and saved the system will then prompt for a certificates upload.

Upload VPN Certificate: Select the Open VPN certificate (with the .conf extension) from local storage and click upload.

Sets ring and dial tone, impedance, and other settings for the selected area.

Area Parameters: Select the area where the gateway will be in use from the available list. Leave as default if your area is not listed and set tone and other settings manually.

Allows new users to be setup and granted access to each page in the system as needed. Used to grant limited access to the IT staff of end users for example. This setting is optional.

Username: A username, should not include spaces or special characters.

Password: Must be 8 characters and include one lower and upper case letter, one number, and one special character.

User Authorities: Set if the user will have read only, or read/write access to the selected pages.

Page: Enables access to each page in the system individually. A check all and uncheck all button are available at the end of the page.

Check Channel Status: Check which ports the user will have access to view or edit.

This page allows the registration details (username & password) of a port to be set or modified. These settings are set via Phonesuite systems and should not modified.

Port: Sets the port number to modify.

Type: FXS, not currently changeable.

Register Port: Enables / Disables port registration. If set to No the port will not register and will not process calls.

SIP Account: This is the username of the SIP device in the PBX.

Display Name: Sets the Displayname field of the SIP message. If not set it will be set with the callerid value.

Password: The password of the SIP device in the PBX.

Display Name preferred: The entire gateway can be registered, or groups of ports can be registered. In both of these cases the Display Name set will be used unless this option is enabled, in that case the display name entered above will be used. Normally each port is registered individually and in those cases this setting has no effect.

Auto Dial Number: This settings, sometimes called Ring-Down or Hotline, will automatically dial the set extension or number when the port goes off hook.

Wait time before auto dial (s): Sets how long in seconds to delay the dialing of the number above. Only applies if a number or extension is set in the auto dial number field.

Input / Output Gain (dB): Sets the value in dB for input gain, the range is -6 to 6 with a default value of 0.

Echo Canceller: Enables the echo canceller feature. Enabled by default.

CID: This enables the caller ID feature for the ringing station. If a caller ID display phone is connected the caller ID information will be presented after the first ring (numbers only, all other characters are stripped). Enabled by default.

Call Waiting: Enables the call waiting feature, disabled by default.

DND: Enables DND for the port. If enabled a SIP message 403 (Forbidden) will be sent to any incoming call.

Call Forward: Enables call forwarding and allows the selection of Unconditional (always), Busy, and No Reply (no answer) along with what number to forward the calls to and a timeout for No Reply.

Advanced Configuration: A check box that provides additional options.

Talkback: Not used.

Port: Select the port to modify settings for.

Forbid Outgoing Call: Disables outbound calling from the selected port. If set an additional option appears to set if this applies at all times, or to allow a time of day, and day of week where outbound calls are prohibited.

Blacklist of FXS Out Calls: Here a list of numbers can be set that will be blocked from outbound dialing. To set both 888 and 887 as blocked numbers for example enter 888;887. If a blocked number is attempted the caller receives a busy signal.

Port groups can be used to tie registered ports together so they can pickup calls for another port (by dialing *8 for example). This feature is not often used and not recommended.

Index: The number of the port group.

Description: A human readable name for the group.

Register Port: If set to Yes a SIP Account, Display Name, and Password field appear. This is used register one SIP account for multiple ports.

Authentication Mode: Sets if the port group should handle the SIP registration, in most cases this is set to “Do Not Register”. If Register Port is set to Yes then an additional option of Register Port Group is available. This option is required to make more than one port use the same SIP account.

Port Select Mode: If the port group is set up a registered port group (i.e. several ports all using the same SIP account) then this setting sets the logic the gateway will use when routing inbound calls. Options are:

• Increase: Routes call to first available port starting at the lowest port (default).

• Decrease: Route calls to first available port starting at the highest port.

• Cycle Increase: AKA Round Robbin with memory, routes calls to the first available port but remembers what port took the last call and starts at the next highest port for the next call.

• Cycle Decrease: Same as above but searches for ports in decreasing sequence.

• Group Ringing: Rings all ports at the same time.

• Ringing by Turns: If ports 2,4,6,8,1,3,7 are entered the gateway will ring ports in that order. The timeout is set in seconds with a range of 15-60 with 20 being the default.

Preemptive Answer Keyboard Shortcut: AKA Pick Up Group Code, this is a code (i.e. *8) that when dialed will pickup the first inbound call ringing the group.

Port Reused by Multiple Groups: Can be set to Yes or No. If set to Yes then ports can be part of multiple groups. This is not recommended as the behavior for inbound calls becomes complex and difficult to troubleshoot.

Port: What ports are assigned to this group.

This is an advanced section that is not often needed. Its only use when a PBX will send calls to the gateway without a registration, in such cases this feature allows calls to be routed directly to port groups or routed by number. Because Phonesuite systems always register ports this setting is not used.

It first set if a call is routed before manipulation to the number or afterwords. For example if the caller dials a 10 digit number (303-555-1111) and a rule exists to add a leading 9 (9-303-555-1111) will the gateway apply the dialing rules before adding the 9 or after. This is normally set to “Route before Number Manipulate”.

Index: Rules are applied from lowest to highest index.

Description: A human readable name for the rule.

Source IP: The IP of the PBX

CallerID Prefix: By default this is set to *, which is all calls. If only calls to of from a specific number need to be routed to a specific port this setting, and the one below it can be used.

CalleeID Prefix: Same as above.

Route by Number: If this is checked then the call will be routed to the port that has been set with this number (under Port → FXS). Otherwise the destination can be manually set to any existing Port Group.

Character Mode

This page allows rules to be setup using only a text string. Useful for setting up many rules. The format is:

<ip> <callerID Prefix> <CalleeID Prefix> <Port Group (0 for none)> <Route by Number (1=yes, 0=no)> <Description>

Examples:

192.168.2.108 * * 0 0 Test2
192.168.2.108 605 * 2 1 Test

Index: Rules are applied from lowest to highest index.

Description: A human readable name for the rule.

Source IP: The IP of the PBX

Source Port Group: By default this is set to *, which is all calls. Otherwise allows the setting of an existing port group.

CallerID Prefix: By default this is set to *, which is all calls. If only calls to of from a specific number need to be routed to a specific port this setting, and the one below it can be used.

CalleeID Prefix: Same as above.

Route by Number: If this is checked then the call will be routed to the port that has been set with this number (under Port → FXS). Otherwise the destination can be manually set to any existing Port Group.

Destination Address: The IP address of the PBX.

Destination Port: The port that the SIP signaling will use, normally 5060.

Character Mode

This page allows rules to be setup using only a text string. Useful for setting up many rules. The format is:

<Port Group (0 for none)> <callerID Prefix> <CalleeID Prefix> <Destination Address> <DestinationPort> <Description>

Examples:

2 555 * 192.168.2.108 5060 AnalogtoSIP2
0 * * 192.168.2.108 5060 AnalogtoSIP1

These settings allow the caller and callee ID to be edited. For example using Tel → CalleeID a leading 9 prefix could be added to outbound calls. Number manipulation of this type should normally be handled by the PBX to ensure its standard for all ports on all gateways and as such its not recommended that this setting be used.

Open

Adds a leading 9 to all outbound calls.

WEB Port: Sets what port the web GUI will accessible on, 80 by default.

Access Setting: Allows the setting of an IP address whitelist (only IPs on the list can access the system) or blacklist (IPs on the list are excluded from accessing the system, all others are allowed). It is recommended that whitelists be setup in the router / firewall for the network.

Enable WEB Login Verification Code: Enables 4 letter captcha for logging into the Gateways web GUI.

FTP: Sets if the gateway will be allowed to download a provisioning template file via FTP. Default is set to Yes.

Interface of debug.php: Enables the logging of interface bugs. Enabled only if requested to do so by Phonesuite.

Telnet: Enables Telnet access to the gateway, disabled by default.

Rpcapd: Enables a daemon that allows remote packet capture for programs like Wireshark. Disabled by default.

Send CDR: Allows the gateway to send specified CDR records to an IP address and port. SetCDR=0

Save to Local: If set to Yes the PSG will save CDR records locally. These records are accessible at Advanced → CDR Query.

NTP: Enables NTP time updates, enabled by default.

NTP Server Address: The default NTP server to request time updates from. Set to time.nist.gov by default.

Synchronizing Cycle: How often in seconds will the gateway synchronize its time with the time server. 3600 seconds (1 hour) by default.

Daily Restart: Sets if the gateway should reboot itself daily. If set to Yes a reset time setting appears. Set to No by default.

System Time: Displays the system time if NTP is enabled (recommended and default) or allows the time to be set if NTP is disabled.

Time Zone: Sets the timezone the gateway is in. This setting must be manually set when being used with Voiceware.

Displays the status of the domain used in the VoIP → SIP Setup. This will not display info if the gateway is using an IP address (vs. a domain name) in the SIP setup.

This page displays the raw configuration of the gateway. There are two files, SMGConfig has port registration information while SHConfig contains the other gateway settings including the SIP server settings.

This page allows the creation of an SSL certificate. An externally generated certificate can also be uploaded here. The only reason to do this is to remove browser warnings about self signed certificates. In most cases this section is not used. Note that Phonesuite does not offer certificate generation.

This section allows certain settings to be reset to defaults without resetting the entire gateway to factory defaults. If settings here are reset the gateway should be provisioned to the Phonesuite systems again to ensure proper operation.

The gateway has three network modes:

• Dual: The default where Eth1 and 2 have their own IP address and network settings.

• Route: Allows Eth2 to be set to DHCP server. This then turns ports 2, 3, and 4 on the gateway into switch ports with the gateway providing the DHCP server and routing traffic out port 1.

• Switch: Bridge Eth1 and 2 using the settings for Eth1

All other settings are default network settings. IPV6 can be enabled as needed but is disabled by default.

This page sets where the gateway look for its provisioning file.

For Voiceware this can be manually set to https://<ip>/p (for example https://1.2.3.4/p).

Voiceware does not use Http usernames or passwords for provisioning and these settings are set with default values for Connectware and thus don’t need to be edited.

Url Method: Sets if the gateway will use option 66 or if the provisioning server is set manually.

URL of the Configure server: This is the URL of the provisioning server. See above for examples of a Voiceware provisioning URL.

Periodic Update Cycle (minute): This is the time in minutes that the gateway will download a provisioning file and apply any changes. Set to 10,080 minutes (one week) by default.

Http Authorization: If checked the system will use the provided username and password when requesting its provisioning file.

MQTT is a message protocol where by remote systems (like the PSG) will send messages to a Broker that acts a bit like a post office, delivering messages to clients that need them.

This setting is used for remote management of the gateway and should not be changed without the request of Phonesuite.

This page allows manual firmware upgrades. Use only with Phonesuite approval and with approved firmware.

This page has several test tools available for troubleshooting. They include:

Packet Capture: Starts a packet capture that can be limited to things like only SIP packets or be set to include all traffic to and from the gateway, once finished the capture can be downloaded for review.

Data Recording: This allows the audio of a call to be captured. The resulting tar file includes separate sound files for inbound and outbound audio as well as audio for the entire call, and only after the call has been setup and audio is transmitted to the PBX.

Test Call: Used to test a SIP call to a provided SIP server to test if a connection can be made. A PTSN test is not used with Phonesuite systems.

• Local Alias: The displayname from field in the SIP invite message.

• Local SIP Account: The username field in the SIP invite message.

• Remote Alias: The displayname in the to field of the SIP invite message.

• Remote SIP Account: The username in the to field of the SIP invite message.

• Called IP Address: The IP address to send the SIP invite to (i.e. the PBX).

• Called Port: The port the test will be made against, normally 5060.

• IP Channel: The gateway has 120 channels (0-119) that can be used for SIP calls.

• DTMF: Once the call is established these DTMF tones will be played to the remote end.

• Add or Modify Invite Header Field: These options can be used to override the SIP header fields, when used both the field name and content must be included. Advanced use only.

The Pickup, Hangup, Start, and Stop commands are like that of a physical phone. Pickup and Hangup are like picking up and placing down a physical analog handset. Start and stop are like send to start a call and an end to end a call.

Call Log

The call log page is used for troubleshooting for a specific port. To use check Enable Call Log and select a port. Once done place a test call and the gateway will display the logs of how it processes the dialed number, what dialing rule (digit map) it used and how it processed the call.

Logs are split between the IP logs and the logs for the gateways internal processing.

This tool is very useful for troubleshooting the gateway as it displays a lot of information about how the gateway is processing the call and what info the PBX sent when it started a call.

SIP Log

The SIP log shows the SIP messages that are sent between the gateway and the PBX. These higher level messages are the same that would be gathered with a packet capture. In most cases if SIP debug is needed a packet capture is a better choice because it can be opened and examined with 3rd party tools like Wireshark (thus making debug easier).

This log shows all changes made to the configuration of the gateway. This can be used to find and revert changes made to the gateway should recent changes have caused an issue.

This page allows a full backup of the gateway to be taken and / or a backup to be restored.

This page allow the factory reset of the gateway. Once clicked the gateway will ask if you are sure and if yes is selected the gateway will reboot and reset.

This page allows the setting of system maintenance features.

Watchdog: This is a timer that will reset the gateway. During normal operations the operating system resets the watchdog timer frequently but if the system locks up the watchdog timer counts to 0 and restarts the system, hopefully clearing the error.

The default settings are to enable watchdog and the timer is set to 5 seconds.

Automatically restart the service if undetected: If checked the gateway will reboot itself after the time set below if it does not detect that services are running. This setting is similar to the above setting but accounts for failures where the OS loads (and thus resets the watchdog timer) but the gateways services fail to start.

Threshold to Judge Heartbeat Loss for Service(s): The time in seconds that the gateway will wait before rebooting if services do not start. 60 seconds by default.

Technical Report 069 is a remote management protocol. It is not currently supported by Phonesuite and setting up TR069 is outside the scope for this documentation. Enable=0

By default the gateway will setup its own access control such that it will accept SIP traffic only from the SIP server IP address(es). Modification of these rules is not recommended as they can cause loss of communication to the PBX or the web GUI.

Rules are in the IPTables format, more information about IPTables can be found here: https://linux.die.net/man/8/iptables

This page allows a simple Ping test to be preformed.

Source IP Address: What Eth port will the ping be sent from.

Destination Address: What IP address to ping (i.e. 8.8.8.8)

Ping Count (1-100): How many times to ping the IP.

Package Length (56-1024 bytes): How large should the packet size be.

After the IP and ping count is filled in click the Start button to begin the pings. After the pings information will be displayed showing each ping and a summary.

This page allows the entry of a DNS name (i.e. Google.com) and tests if the DNS is reachable and what IP was used.

Fast Reverse Proxy (FRP) is a method of remote management that uses an FRP server that allows remote connection to the device without opening ports in a local firewall. Useful for situations where opening ports in a firewall is not allowed or access to the firewall is prevented.

It it outside the scope of this manual to provide information on setting up an FRP server but more information on the project can be found here GitHub - fatedier/frp: A fast reverse proxy to help you expose a local server behind a NAT or firewall to the internet.

Trace Route (TraceRT) is a tool that will send pings to a remote server and report back each hop along that path. A hop normally represents a router or managed switch between you and the destination.

Source IP Address: What Eth port will the ping be sent from.

Destination Address: What IP address to trace to (i.e. 8.8.8.8).

Maximum Jumps (1-255): Sets the total number of hops that will be reported on, default is 30.

After entering the above info press the start button to see a list of hops between the gateway and the remote service. Can be useful to troubleshoot connection issues locally and discover network equipment on the local LAN.

Allows the current login users username and password to be set. Enter the current username and password and then the new username and password as needed and then click save.

This page allows the gateway to be restated. There are two options.

Service Restart

This restarts the gateways services without powering down and back up the gateway. A boot up dump file can also be generated (if the option is clicked) and downloaded to see how the gateway booted up and if it encountered any errors.

System Restart

Restarts the gateway by powering it down and back up. Can also generate a dump file if checked.