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Solution Integration > PBXs > BT Call Server > Web UI > Configuration

 

Trunks Configuration

 

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From Asterisk V3.x and greater the configuration syntax has changed due to the use of a new SIP stack module, which now makes use of newer (and different) configuration files.

Care should be taken when upgrading from Asterisk versions <= v2.0 to versions >= v3.x

It is therefore highly recommended that lab testing is done to ensure you have the correct configurations prior to any upgrade.

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From V4.x and greater, Extensions and Trunks configuration are separated into their own configuration pages.

If you are upgrading from v3.x and restoring configuration files, make sure to correctly separate the extension and trunk configurations into their respective configuration pages.

 

NOTE: Configuration syntax is the same as v3.0 and greater

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Configuration (Admin UI > Configuration > Extensions / Trunks) is handled via the PJSIP Configuration Wizard configuration file (pjsip_wizard.conf)

Refer to the official Asterisk documentation for further advanced configuration details: res_pjsip: SIP Resource using PJProject

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•Trunks Configuration provides full advanced control over settings related to trunks.

 

Configuration-Trunks_01

 

•A default template, "trunk-defaults", is available to simplify management by applying global settings to all trunks.

 

[trunk-defaults](!)

type = wizard

transport = transport-tcp                                                ; Explicit transport configuration to use

endpoint/allow = !all,ulaw,alaw,opus,ilbc,g722                ; Media Codec(s) to allow

endpoint/context = incoming                                        ; Dialplan context to use

endpoint/dtmf_mode = auto                                                ; DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.

endpoint/direct_media = yes                                        ; Enable direct media (peer to peer RTP)

;endpoint/tos_audio = ef                                                ; Sets TOS for audio streams (QoS)

;endpoint/cos_audio = 5                                                ; Sets COS for audio streams (QoS)

aor/qualify_frequency = 30                                        ; Interval at which to qualify an AoR (SIP Options Ping)

 

 

•A new Trunk can be added with the settings adapted to your needs

 

;TrunkConfiguration, please don't delete this line

; remote_hosts = Remote hosts to connect to in the form of host[:port]

; identify/match = IP addresses or networks to match against. Allows inbound from the remote hosts IP's

 

[TRUNK](trunk-defaults)

remote_hosts = IP_address_PBX:5060

identify/match = IP_address_PBX

 

NOTE: Each additional Trunk added should define which template to pull its global settings from. In this case (trunk-defaults)

 

•Once editing is complete, press Update
After updating the configuration you need to "Reload configuration"

 

•From the navigation menu, select "System > Maintenance"

•Select "Reload configuration"
(or "Reload configuration on both servers" if redundancy is set)

 

Configuration-Extensions_04

 

 

If connecting to a SIP based Carrier you may also have to set up an AudioCodes SBC configuration which can be found  at SIP PSTN Trunk SBC Configuration