Accueil > Trunks. Cet écran répertorie tous les trunks du système avec les détails suivants : Nom Nom du Trunk SIP. […]">
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Updated on May 21, 2023

4. Trunks

Estimated reading: 33 minutes

Trunks are transmission lines between two systems. This transmission is done using a wide range of PSTN and VoIP technologies.

The trunks can be found in: Master Tenant > Home > Trunks. This screen lists all the trunks of the system with the following details:

  • Name

SIP Trunk name. It can be a name or an IP address.

  • Provider

Provider template name.

  • Slices

Number of incoming/outgoing slots available for the trunk.

  • Trunk type

Trunk type.

  • Protocol

Protocol used by the trunk.

Trunks

Add/Edit a Trunk

When adding a new trunk, the first step requires selecting "Provider" and "Device".

Search

The search bar filters trunks by name and provider.

  • Search

Provide information to search for here and press Enter to filter the records.

Edit a trunk

The second step of the trunk installation and modification command opens the following options:

General

  • Name or number

Some providers require this field to equal the DID number (for example, 55510205); but if you are connecting two systems, the IP address can also be used.

  • Country

Country where the service provider resides.

  • Emergency trunk

Indicates whether emergency services (police, ambulance, etc.) should be called via this trunk.

Authentication

  • Peer host

The IP address of a peer host system to which calls are sent.

  • Username

The username for authentication with the service provider.

  • Peer Username

The peer username for authentication with the service provider.

  • SIP password

The secret/password used to authenticate with the service provider.

  • Peer secret

The peer secret/password used to authenticate with the service provider.

Advanced options

General

These options are frequently used and are necessary for the normal operation of the trunk. Some of these fields are preconfigured with the default values. It is not recommended to change them.

  • User type

    • User

The trunk only accepts incoming calls.

    • Peer

The trunk only makes outgoing calls.

    • Friend

The trunk makes incoming and outgoing calls.

  • DTMF (Dual Tone Multi-Frequency) mode

Trunk DTMF mode. A specific frequency (made up of two distinct tones) for each key so that it can be easily identified by a microprocessor. For example:

    • Inband : In-band audio (requires a 64 kbit codec – alaw, ulaw)

    • RFC2833: default DTMF mode.

    • Info: SIP INFO messages.

    • Auto : Asterisk will use rfc2833 by default for DTMF relay but will switch to in-band DTMF tones if the remote side does not indicate support for RFC2833.

Asterisk will use rfc2833 for DTMF relay by default but will switch to in-band DTMF tones if the remote side does not indicate support for rfc2833 in SDP.

  • Context

Contexts define a scope within PBXware. The trunk context cannot be modified and is the same as the trunk name or number.

  • Status

Set the trunk status on the system. (Active/Inactive/Suspended)

(For example, rather than deleting the trunk, you can disable it at the system level by selecting "inactive" here).

  • Qualification(ms)

Time interval in milliseconds at which a "ping" is sent to a host in order to know its status. In PBXware 5.1, 'Qualify' is set to 8000 by default.

(For example, you set this field to 2000, if more time than that is needed to reach the host, the host is considered offline).

  • National dialing code

National dialing code used at the trunk destination.

(For example, for the United States it is the number 1, for the United Kingdom or Germany it is the number 0).

  • E164 accepted

Indicates whether the trunk supports calls from destinations in E164 format.

(By enabling this option, each dialed number will be transformed into this form: COUNTRY_CODE + AREA_CODE + DIALED_NUMBER.

(For example, if you dial "5551020", the system will dial "12125551020".).

  • Pass-thru mode

Pass the dialed digits without any conversion (E164, National, Area code).

  • Leave National Code

In some countries, the national code is removed automatically. If it is set to "Yes", the national code will not be removed from the dialed number.

  • Local phone code

Add the local area code to the dialed number, if required by the service provider. (By default, the local area code is removed when dialing).

(For example, the user dials 55510205, the area code is 212. If the call goes through this trunk, PBXware will dial 21210205).

  • Minimum length of the dialed number (6.6.1)

Set a minimum number of digits dialed for a call to leave the system.

  • Prefix

Value added to all dialed numbers passing through the trunk.

(For example, prefix 5 + dialed number 123 = the system dials 5123).

  • Test number

This is the number that the trunk will attempt to call when trunk monitoring is enabled. If the response is not: ANSWER, BUSY, CANCEL, or NO ANSWER, this call is considered non-functional and the monitoring system will send a notification email.

  • Check additional headers

Some trunk providers may send incoming INVITEs with additional headers (X-Mnet-InLeg) containing custom strings. When a call enters PBXware and the call flow results in some kind of call forwarding to the provider, it may request the same custom header with the same value in order to know which call is involved.

For example:

System 1 > Trunk > Custom SIP Headers > X-Mnet-Inleg: 1alpha457908

System 2 > Trunk > Check additional headers > X-Mnet-Inleg

System 1 calls system 2. System 2 must detect the X-Mnet-Inleg header and keep it.

  • SIP response for nonexistent DIDs

Select the SIP response that will be sent to a provider if a DID that was called does not exist on the system

The following options will be used to send to a provider if a DID that was called does not exist:

  • By default
  • 403 – Call rejected
  • 404 – Not Found
  • 408 – Request Timeout
  • 410 – Gone
  • 480 – Temporarily Unavailable
  • 484 – Address Incomplete
  • 486 – Busy Here
  • 488 – Not Acceptable Here
  • 500 – Internal Server Error
  • 501 – Not Implemented
  • 502 – Bad Gateway
  • 503 – Not Implemented
  • Emergency location header

Set an emergency location header.

Authentication

  • Host

The IP address to which the host trunk connects.

  • Authname

Value assigned to the Digest username = SIP header.

  • Auth

Authenticate for outbound calls to other domains.

  • Inbound IP addresses (separated by new line)

You can limit SIP traffic to and from this peer to a certain IP or a certain network.

If your trunk provider has several IP addresses from which it sends traffic, all these IP addresses must be listed here, separated by a new line.

  • Insecure

Generally used to allow incoming calls while having a “friend” type entry defined with a username and password

Example:

    • Port – Ignores the port number from which the request originates.
    • Invite – Does not require authentication of incoming INVITEs.
    • Port-Invite – Does not require an initial INVITE to authenticate and ignores the port from which the request originates.
    • Very – Ignore authentication (username/password).

  • Sign up

Registration method on the remote server.

(For example, providers may require a different form of registration on their server. You can choose between “registration not required”, “register with a phone number”, “register with a username”, “register with identification name”, “register with authname@host”, and “register with authname@trunk-name”).

  • Registry suffix

The service provider may require different registration methods for its services. Select the appropriate method as required by the provider.

  • Default expiry (s)

Registration timeout in seconds (the default value is 3600).

  • Max registration attempts

Maximum number of registration attempts (default value is 10).

  • From the user

What to display when calling this peer FROM asterisk.

  • Of the domain

Domain data is required by some providers.

(For example, if your provider needs this information, indicate the exact value here).

  • user=phone in URI

If this option is set to “Yes”, “user=phone” is added to the uri that contains a valid phone number.

  • Outbound Proxy

Send the outbound signaling to this proxy, not directly to the peer.

Related network

These options define important network-related values concerning NAT.

  • Transports

Type of transfer protocol that will be used on PBXware.

    • UDP

      (User Datagram Protocol)

With UDP, computer applications can send messages, in this case called datagrams, to other hosts on an IP (Internet Protocol) network without prior communication to set up special transmission channels or data paths.

    • TCP

      (Transmission Control Protocol)

Provides reliable, ordered and error-checked delivery of a byte stream between programs running on computers connected to an intranet or the public Internet.

    • TLS

      (Transport Layer Security)

Cryptographic protocol that provides security for communications over the Internet.[1] They use asymmetric cryptography for key exchange authentication, symmetric encryption for confidentiality, and message authentication codes for message integrity.

  • Encryption

This option enables or disables encryption in the PBXware transport.

Available options:

    • Offer if possible (TLS only)

Offer encryption only if possible and only with the TLS protocol.

    • Required

Encryption always required.

    • Offer (TLS only)

Always offer encryption with TLS.

  • Media Direct

If you allow RTP (Real-time Transport Protocol - RTP, is a network protocol for streaming audio and video over IP networks) voice traffic to bypass PBXware.

  • Direct RTP

Here, you can enable or disable the new experimental direct RTP configuration. Setting this value to “Yes” configures the call directly with peer-2-peer media without re-invites. It will not work for video and cases where the called party sends RTP payloads and fmtp headers in the 200 OK that do not match the callers' INVITE. It will also fail if Media Direct is enabled while the device is actually behind NAT.

  • Default IP

The IP address that will be used until registration.

Caller ID

  • Allow Remote-Party-ID

Define whether PBXware allows the "Remote-Party-ID" header

  • Send the Remote-Party-ID

Select whether “Remote-Party-ID” should be added to the URI.

The options are as follows:

    • Use Remote-Party-ID – Use the “Remote-Party-ID” header to send the identity of the remote party.
    • Use P-Asserted-Identity – Use the “P-Asserted-Identity” header to send the identity of the remote party.

  • Send the number to RPID for Anonymous Calls

Select whether to send the original caller ID in RPID.

  • Connected line updates

Enable or prevent sending RPID updates once the call is established.

  • RPID with SIP update

In some cases, the only method by which a connected line change can be immediately transmitted is with a SIP UPDATE request. If they are communicating with another Asterisk server and users want to pass such UPDATE messages to it, they must enable this option. Otherwise, we will have to wait until we can send a re-invite to transmit the information.

  • Send the PAI header

The P-Asserted-Identity contains the caller ID information for the call on the SIP INVITE packet. This is the acceptable way to specify privacy information for calls. This field allows users to send the preferred PAI header using one of the following variables:

    • %CALLERIDNUM%
    • %EXT%

  • Privacy for the PAI header

Set the privacy of a PAI header

Users can select one of the privacy options from the drop-down menu:

    • None – The caller ID name and number are not anonymous when calling out of the trunk.
    • Anonymous – The caller ID name and number are anonymous when calling out of the trunk.
    • Anonymous with privacy ID – The caller ID name and number are anonymous when dialing out of the trunk, but the privacy ID is added to the SIP header.
    • Anonymous with number – The caller ID name is set to anonymous, but the number is displayed.
    • Anonymous with number and privacy ID – The caller ID name is set to anonymous, but the number is displayed and the privacy ID is added to the SIP header.

  • Outbound caller ID

If the caller ID is not set by the UAD, the value provided here will be used for all outgoing calls.

  • Allow ES caller ID

This option defines whether the ES (enhanced services) caller ID will be allowed on this trunk. Any extension can set a custom caller ID for each trunk on the system. When this option is set to "Yes", this caller ID will be used instead of the trunk's outbound caller ID.

Slices

  • Inbound limit

Set the number of simultaneous inbound calls the Trunk can handle

(For example, setting this option to 4 means that 4 simultaneous incoming calls are possible. Any additional call will receive a busy tone.)

  • Outbound limit

Set the number of simultaneous outbound calls the Trunk can handle.

(For example, setting this option to 4 means that 4 simultaneous outgoing calls are possible. Any additional call attempting to use this trunk will be rejected or redirected to other trunks depending on what is defined in the system/Extensions.)

  • Ring time (sec)

Set the ring delay (in seconds) for outbound calls via the Trunk

  • Email on limit exceeded

Send an email when the outbound limit is reached

  • Outbound call options

Define advanced dialing options for all outbound calls

Codecs

Codecs are used to convert voice signals from analog to digital and vice versa. These options define the preferred codecs used by the extension.

Codecs 

    • ITU G.711 ulaw – 64 Kbps, sample-based, used in the United States.
    • ITU G.711 alaw – 64 Kbps, sample-based, used in Europe.
    • ITU G.722 – 64 Kbps.
    • Opus – 6-510 Kbps, 2.5 ms – 60 ms frame size.
    • ITU G.723.1 – 5.3/6.3 Kbps, 30 ms frame size.
    • ITU G.726 – 16/24/32/40 Kbit/s.
    • ITU G.729 – 8 Kbit/s, 10 ms frame size.
    • GSM – 13 Kbps (full rate), 20 ms frame size.
    • iLBC – 15 Kbps, 20 ms frame size: 13.3 Kbps, 30 ms frame size.
    • Speex – 2.15 to 44.2 Kbps.
    • LPC10 – 2.5 Kbit/s.
    • 261 Video – Used on ISDN lines with a resolution of 352×288.
    • 263Video – Low-bit-rate encoding solution for videoconferencing.
    • 263+ Video – Extension of H.263 that provides additional features that improve compression on packet-switched networks.
    • 264Video – Offers an excellent balance between quality and size.
  • Ptime

Time in ms – represents the amount of audio recorded in itself that each packet will have

  • Video support

Set this option to Yes to enable SIP video support.

  • Force the codec on the Trunk's outbound channel

With this option, you can force the use of the codec for outbound trunk calls.

  • Automatic framing (RTP packetization)

If auto-framing is enabled, the system will choose the packetization level based on the preferences of the remote endpoints.

(For example, if the remote end requires the RTP packets to be 30 ms, your PBXware system will automatically send packets of this size if this option is enabled. The default value is set to 20 ms and also depends on the minimum frame size of codecs like G.729 which has 10 ms at minimum).

Additional configuration

This option is used to provide additional configuration parameters for the SIP, IAX, and MGCP configuration files. The values provided here will be written to these configuration files.

Custom SIP headers (separated by a new line)

The SIP header is a function of SIP. These headers help convey message attributes to ensure that packets of information travel along the correct path when communicating between devices existing on separate networks.

In order to use custom SIP headers, here are the variables that can be used:

  • %CALLERIDNUM%
  • %EXT%

PSTN (Public Switched Telephone Network)

These options can be found when adding a trunk and selecting the device.

General DAHDI options

  • Name

Trunk name/number

  • Slice

Which card channels are used, separated by commas.

(For example, if channels 2 and 4 are used on your card, set "2, 4" here. If all four channels are used, set "1-4" here).

  • Status

Set the trunk status

  • Signaling

Signaling protocol for analog circuits that better detects remote disconnections.

  • Group

Each PSTN trunk must belong to a group. Selecting any group will enable the trunk.

Echo canceller

  • Echo cancellation

Enable the echo canceller.

  • Echo canceller

Select the type of echo canceller.

RX/TX

  • RX Gain

Receive the signal's decibel.

(For example, if the incoming sound is weak and you do not hear the other party well, set this option to 2. This should increase the incoming sound by 2 decibels).

  • TX Gain

Decibel of the transmit signal.

(For example, if the outgoing sound is weak and the other party does not hear you well, set this option to 2. This should increase the outgoing sound by 2 decibels).

Locality

  • Country

Destination of the trunk connection.

(For example, if your system is located in the United States, select USA here)

  • National dialing code

National dialing code used at the trunk destination.

(For example, for the United States it is 1, for the United Kingdom and Germany it is 0.)

  • E.164

Indicates whether the trunk supports dialing destinations in E164 format. By enabling this option, each dialed number will be formatted as follows: COUNTRY_CODE+AREA_CODE+DIALED_NUMBER.

(For example, if you dial 5551020, the system will dial 12125551020).

Advanced options

DAHDI General

  • Language

Default language.

  • Context

Contexts define a scope within PBXware.

  • Status

Trunk status

    • Active
    • Inactive
  • Signaling

Signaling method

    • E&M
    • E&M Wink
    • FXS Loopstart
    • FXS Groundstart
    • FXS Kewlstart
    • FXO Loopstart
    • FXO Groundstart
    • FXO Kewlstart
    • PRI CPE side
    • PRI Network side
    • BRI CPE side
    • BRI Network side
    • BRI CPE PTMP
    • BRI Network PTMP
  • Test number

This is the number that the trunk will attempt to call when trunk monitoring is enabled. If the response is not: ANSWER, BUSY, CANCEL, or NO ANSWER, this call is considered non-functional and the monitoring system will send a notification email.

  • Music on hold

Select the music class to be used for music on hold. If not specified, the “Default” will be used.

  • Inbox

Define a voicemail context.

  • Group method

This option specifies how to choose a channel to use in the specified group.

The four possible options are:

    • g : select the non-busy DAHDI channel with the lowest number (i.e. the ascending sequential hunt group).
    • G: select the non-busy DAHDI channel with the highest number (i.e. the descending sequential hunt group).
    • r: use a circular search, starting from the next highest channel than the last time (i.e. an ascending rotary hunt group).
    • R: use a circular search, starting from the lowest channel following the last time (i.e. a descending rotary hunt group).

RX/TX

  • RX Wink

Set the synchronization parameters

    • Pre-flash (50 ms)
    • Pre-flash (50ms)
    • Flash (150 ms)
    • Receiver flash time (250 ms)
    • Hook flash (300 ms)
    • Anti-bounce timeout (600 ms)
  • RX Gain

Receive the signal's decibel.

(For example, if the outgoing sound is weak and the other party does not hear you well, set this option to 2. This should increase the outgoing sound by 2 decibels).

  • TX Gain

Decibel of the transmit signal.

(For example, if the outgoing sound is weak and the other party does not hear you well, set this option to 2. This should increase the outgoing sound by 2 decibels).

PRI

  • Switch type

Example:

    • National ISDN 2
    • Nortel DMS100
    • AT&T 4ESS
    • Lucent 5ESS
    • EuroISDN
    • Old national ISDN

  • PRI dial plan

Set the dial plan used by certain switches

Example:

    • Unknown
    • Private ISDN
    • Local RNIS
    • National ISDN
    • International RNIS

  • PRI local dial plan

Set the dial plan for locally called destinations

Example:

    • Unknown
    • Private ISDN
    • Local RNIS
    • National ISDN
    • International RNIS

  • PRI trust CID

Trust the caller ID information provided

  • PRI indication

How to signal "busy" and "congestion" on a PRI

Example:

    • Out of band – Busy/congestion signal out of band with RELEASE/DISCONNECT
    • In-band – Busy/congestion signal using in-band tones

  • Network-specific installation

If required by the switch, select the network-specific installation

Example:

    • None
    • sdn
    • megacom
    • accunet

Caller ID

  • Outbound caller ID

Caller ID set for all outbound calls where the caller ID is not defined or not supported by a device.

  • Allow ES caller ID

This option defines whether the ES (Enhanced Services) caller ID will be allowed on this trunk. Any extension can set a custom caller ID for a trunk on the system. When this option is set to "yes", the caller ID will be used instead of the trunk's outbound caller ID.

  • Caller ID (for analog or inbound)

Caller ID can be set to "as received" or to a specific number if you want to override it.

  • Use the caller ID

Whether or not to use the caller ID.

  • Hide caller ID

Whether or not to hide the outbound caller ID.

  • Restrict the caller ID

Whether or not to use caller ID presentation for the outbound call sent by the calling switch.

  • Use caller ID presentation

Whether or not to use caller ID presentation for the outbound call sent by the calling switch.

  • Caller ID start

Which signals the start of caller ID

Example:

    • Ringing = a ring signals the departure.
    • Polarity = polarity reversal signals the start.
  • Caller ID call waiting

Whether or not to enable call waiting on FXO lines.

  • Send the caller ID after

Some countries, such as the United Kingdom, have different ring patterns (ring-ring), which means that the caller ID must be set later, and not just after the first ring, as is the case by default.

Echo canceller

  • Echo cancellation

Echo cancellation level enabled.

  • Echo training

Briefly mute the channel, for 400 ms, at the start of the conversation, cancelling the echo. (Use this only if "Echo Cancellation" does not work as expected).

  • Echo cancellation when bridged

Enable echo cancellation when bridging. Generally not necessary, and in fact undesirable, to cancel echo when the circuit path is entirely TDM.

Call functions

  • Call waiting

Whether or not to enable call waiting on FXO lines.

  • Three-way call

Supports three-way calls. If enabled, the call can be put on hold and another call can be made.

  • Transfer

Supports call transfer and also allows call parking (replaces the “canpark” setting).

  • Can call forwarding

Supports call forwarding.

  • Callback

Whether or not to support the “*69” callback. Dials the extension number of the last caller.

  • Overlap dialing

Enables overlap dial mode (sends overlapping digits).

  • Pulse dialing

Use pulse dialing instead of DTMF. Used by FXO devices (FXS signaling).

Call indications

  • Distinctive ring detection

Whether or not to perform distinctive ring detection on FXO lines.

  • Busy detection

Enable monitoring of the busy beep pattern.

  • Account busy

How many busy tones to wait for before hanging up. Larger settings reduce the likelihood of random hang-ups. "Busy detection" must be enabled.

  • 4
  • 6
  • 8
  • Call progress

Easily detect false hang-ups.

  • Immediate

If the channel is answered immediately or if the simple switch provides the tone, read the digits, etc.

Call groups

  • Call groups

Which group is allowed to pick up incoming calls by dialing *8. The default value is empty.

  • Group call pickup

Which groups are allowed to pick up calls by dialing *8. The default value is empty.

FXS channels

  • FXS Loopstart

Signals to the other end that it wants dial tone by short-circuiting the wires.

  • FXS Groundstart

Signals to the other end that it wants dial tone by grounding one of the wires.

  • FXS Kewlstart

Signaling protocol for analog circuits that better detects remote disconnections.

(For example, select the channels of the card to monitor with it. For example "1, 4" or "1-4". These numbers must match the "Slot(s)" field).

Locality

  • Country

Destination of the trunk connection.

(For example, if your system is located in the United States, select USA here)

  • National dialing code

National dialing code used at the trunk destination.

(For example, for the United States it is 1, for the United Kingdom and Germany it is 0.)

  • International dialing code

International dialing code at the trunk destination.

(For example, for the United States, the numbers are 011. For the United Kingdom and Germany, 00, etc.)

  • E.164 Accepted

Indicates whether the trunk supports dialing destinations in E164 format.

(For example, enabling this option will reformat any dialed number into the following form COUNTRY_CODE+AREA_CODE+DIALED_NUMBER. For example, if you dial 5551020, the system will dial 12125551020).

  • Pass-thru mode

If this option is enabled, the dialed number is passed through the trunk without modification.

  • Leave National Code

In some countries, the national code is removed automatically. If it is set to "Yes", the national code will not be removed from the dialed number.

  • Local phone code

Add the local area code to the dialed number, if required by the service provider. (By default, the area code is removed when dialing)

(For example, the user dials 55510205, the area code is 212. If the call goes through this trunk, PBXware will dial 21255510205).

  • Write the dial code

Should the national and international prefix be written in the configuration files.

(For example, enable this option if the provider requires it).

  • Prefix

Value added to all dialed numbers passing through the trunk.

(For example, if the prefix is 5, the dialed number is 123, the system will dial 5123).

Other PSTN options

  • ADSI (Analog Display Services Interface)

Enable remote control of the screen phone with the soft keys. (Only if you have compatible ADSI CPE equipment).

  • Jitter Buffers

Configure the Jitter Buffers. Each one lasts 20 ms.

  • Relax DTMF

If you experience problems with DTMF detection, you can relax the DTMF detection settings.

  • Fax detection

Enable fax detection

Example:

  • Both
  • Inbound
  • Outbound
  • No

Scope

  • Scope number

Range number.

  • Scope duration

How to synchronize timing devices

Example:

    • 0 – do not use this range as a synchronization source
    • 1 – use as main synchronization source
    • 2 – set as secondary, and so on
  • Line Build Out

Length of the last leg of the connection and is set to zero if the length is less than 133 feet

Example:

    • 0 db (CSU) / 0-133 feet (DSX-1)
    • 133-266 feet (DSX-1)
    • 266-399 feet (DSX-1)
    • 399-533 feet (DSX-1)
    • 533-655 feet (DSX-1)
    • -7.5db (CSU)
    • -15db (CSU)
    • -22.5db (CSU)
  • Management

How to communicate with the hardware at the other end of the line

Example:

    • For T1: the framing is one of ami d4 or esf.
    • For E1: the framing is one of ami cas or ccs.
  • Encoding

How to encode communication with the other hardware at the other end.

Example:

    • For T1: the coding is one of ami or b8zs
    • For E1: the encoding is one of ami or hdb3 (E1 may also need crc)
  • Yellow

Indicates whether the yellow alarm is transmitted when no channel is open.

Dynamic scope

  • Dynamic range driver

The name of the driver.

(Example: eth)

  • Dynamic scope address

Driver-specific address (such as a MAC for eth).

  • Dynamic range channels

Number of channels.

  • Dynamic range synchronization

Sets the synchronization priority, as for a normal range. Use ‘0’ so it is not used as a synchronization source, or give them priority as primary, secondary, etc.

FXO channels

  • FXO Loopstart

Channels are signaled using the FXO Loopstart protocol.

  • FXO Groundstart

Channels are signaled using the FXO Groundstart protocol.

  • FXO Kewlstart

Channels are signaled using the FXO Kewlstart protocol.

PRI channels

  • D channels

For example, each ISDN BRI card has 1 D- channel (control).

  • B channels

For example, each ISDN BRI card has 2 B- channels (data)

Custom trunks

The custom trunks are used in rare cases where the PBXware trunk configuration features do not offer enough flexibility for configuring certain providers. In such cases, users must create a custom trunk and then modify the dial plan in the trunks-in.conf file.

  • Search

Users can search for the existing custom trunks using the search bar.

  • Trunk

Displays the name of the custom trunk.

  • Context

Displays the context of the specific custom trunk.

  • Click this button to edit the existing custom trunk.
  • Click this button to delete the existing custom trunk.

Add a custom trunk

Once this button is clicked, users will be prompted to another window where they can create new custom trunks.

  • SIP account

Enter the name of a custom trunk that will be visible in the PBXware GUI.

Example names of custom trunks

  1. Without space (accepted) GlobalTrunk
  2. With space (not accepted) Global Trunk

  • Context

Enter the context by which the system sees this custom trunk. It will be used when editing the dialplan.

(For example, if a user defines the context as "gl_trunk", then, when sending calls to the trunk, it will use it to refer to the given trunk).

Trunks & Tenants

This page allows you to assign SIP trunks to tenants. Tenants will only be able to use the trunks assigned to them.

  • Tenant name

Provides a list of all tenants that currently exist on the system. Clicking on a tenant's name will open a page where you can assign lines to that tenant.

  • Unique tenant code

Provides a list of the tenant codes corresponding to the tenant name in the first column.

  • Trunk

displays the SIP trunks which are defined as primary/secondary/tertiary and the emergency trunks which are defined as primary/secondary/tertiary.

Edit

Click the edit button if you want to modify the trunk configuration for a specific tenant.

By clicking on this button, it is possible to assign trunks/emergency trunks to your tenant or to remove them. In addition, it is possible to set the priorities of the trunks/emergency trunks and the Caller IDs for each trunk.

Allowed trunks

Choose the trunks among all the created trunks that will be used on this tenant for incoming/outgoing calls. Only the Trunks chosen here will be available in the fields below.

Default trunk

Select the primary, secondary, or tertiary trunk by default. Only the trunks defined in the Allowed trunks can be selected.

Emergency trunks

In this section, define which trunk will be set as the primary/secondary or tertiary emergency line.

Through this trunk, emergency service numbers are dialed. Only the trunks defined in the Allowed trunks field can be selected.

Caller identification

Here, you can define a custom CallerID for each trunk. Only the trunks defined in the Allowed Trunks field are visible in this section.

Adding a trunk to the list of allowed trunks will automatically add a caller ID entry in this section. Removing a trunk from the list of allowed trunks will also remove the caller ID entry for that trunk.

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4. Trunks

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