HOWTO SET UP TRIXBOX AS H323 GATEWAY

H323 to SIP

This is the stepwise procedure to set up a VoIP communication from a SIP client to a H323 client.

• Start your gatekeeper. For Gnugk do on a command line

gnugk -ttt -rr

• Start asterisk
• Make sure the file ooh323.conf is present in /etc/asterisk
• In ooh323.conf you have to change:

gateway = yes
gatekeeper = xxx.xxx.xxx.xxx (ip address of your gatekeeper)

In FreePBX -> Setup -> Extensions:

• Create a SIP extension (e.g. extension 501)
• Create a Custom extension:

Extension number: 601
Displayname: Anneke h323
Dial: OOH323/601@129.125.71.172 (ip-address of machine where extension can be found)

(Our convention is to have our SIP extensions in the 5xx range, and H323 extension in the 6xx range).

(Asterisk saves these extensions in the file extensions_additional.conf in the [ext-local] context).

• At asterisk prompt

amportal stop
amportal start

Your gatekeeper should produce output on the command line telling you that ObjSysAsterisk is registering.

ObjSysAsterisk is the h323id of the trixbox as specified in ooh323.conf. (If for some reason you want to register more trixboxes with the gatekeeper you should give each of them a unique h323id in ooh323.conf).

• Open your X-lite and configure it like you always would for a given TrixBox
• Open Gnomemeeting:
• Goto Edit -> Preferences -> H323 settings -> Gatekeeper settings

Registering method: Gatekeeper host
Gatekeeper host: xxx.xxx.xxx.xxx (fill in the IP address of your gatekeeper)
Gatekeeper alias: 601 (number of your h323 extension as you registered it in FreePBX)
Check: register this alias as the primary alias with the gatekeeper
Click on Apply

Your gatekeeper should now produce output on the command line telling you that extension 601 (with the personal data you filled in) is registering.

• Go to your X-lite and dial 601

At this point the H323 client Gnomemeeting should ring; after accepting the VoIP call, the conversation can start :)

SIP to H323

This is the stepwise procedure to set up a VoIP communication from a SIP client to a H323 client.

• Start your gatekeeper. For Gnugk do on a command line

gnugk -ttt -rr

• Start asterisk
• Make sure the file ooh323.conf is present in /etc/asterisk
• In ooh323.conf you have to change:

gateway = yes
gatekeeper = xxx.xxx.xxx.xxx (ip address of your gatekeeper)

In FreePBX -> Setup -> Extensions:

• Create a SIP extension (e.g. extension 501)
• Create a Custom extension:

Extension number: 601
Displayname: Anneke h323
Dial: OOH323/601@129.125.71.172 (ip-address of machine where extension can be found)

(Our convention is to have our SIP extensions in the 5xx range, and H323 extension in the 6xx range).

(Asterisk saves these extensions in the file extensions_additional.conf in the [ext-local] context).

• At asterisk prompt

amportal stop
amportal start

Your gatekeeper should produce output on the command line telling you that ObjSysAsterisk is registering.

ObjSysAsterisk is the h323id of the trixbox as specified in ooh323.conf. (If for some reason you want to register more trixboxes with the gatekeeper you should give each of them a unique h323id in ooh323.conf).

• Open your X-lite and configure it like you always would for a given TrixBox
• Open Gnomemeeting:
• Goto Edit -> Preferences -> H323 settings -> Gatekeeper settings

Registering method: Gatekeeper host
Gatekeeper host: xxx.xxx.xxx.xxx (fill in the IP address of your gatekeeper)
Gatekeeper alias: 601 (number of your h323 extension as you registered it in FreePBX)
Check: register this alias as the primary alias with the gatekeeper
Click on Apply

Your gatekeeper should now produce output on the command line telling you that extension 601 (with the personal data you filled in) is registering.

• Go to your X-lite and dial 601

At this point the H323 client Gnomemeeting should ring; after accepting the VoIP call, the conversation can start :)