Enable SIP Trunking for a Cloud PBX extension
This article describes the SIP trunking feature and how to use it to connect a PBX to an extension. Also, it includes a set of recommendations with examples for seiting up Asterisk to act as a PBX.
When enabled at extension level, this Cloud PBX feature allows you to connect a PBX to the extension (when a DID assigned to that extension is called, it is passed further to the PBX in the SIP:To header). This option was added because some customers wanted to connect different PBX systems to Cloud PBX.
In order to enable this feature, you need a SIP trunking license. If the license is valid, in the Extension → Provisioning and SIP page, under the SIP Preferences area, you'll find the PBX is connected to this extension [] Enable for SIP trunking service option. Select this checkbox, and you will be able to connect a different PBX to that extension.
How to configure Asterisk to act as a PBX
For this, you will need Asterisk box. There are plenty of installation tutorials on the web, so you shouldn't find any difficulty in installing Asterisk.
In our example, ./configure --prefix=/opt is used to set /opt/ as a destination directory for the Asterisk files.
After completing the installation process, you will need to edit the files below:
Example of /opt/etc/asterisk/modules.conf
[modules] autoload=yes noload => pbx_gtkconsole.so noload => pbx_spool.so noload => chan_skinny.so noload => res_smdi.so noload => pbx_ael.so noload => app_voicemail.so load => res_musiconhold.so noload => chan_alsa.so noload => chan_console.so
Example of /opt/etc/asterisk/sip.conf
[general] defaultexpirey=3600 maxexpirey=3600 disallow = all allow = g729 allow = ulaw allow = alaw bindaddr = asterisk_server_ip port = 5060 context = phones nat = no domain = your_server_ip register => 0003*001:password@Cloud PBX_server_ip/0003*001 [mysip] fromuser = 0003*001 fromdomain = asterisk_server_ip defaultuser = 0003*001 authuser = 0003*001 dtmfmode = rfc2833 dtmf = rfc2833 disallow = all allow = g729 allow = ulaw allow = alaw type=peer host = Cloud PBX_server_ip qualify = yes nat = no context = from-voip-provider canreinvite = yes [5000] type=friend defaultuser = 5000 secret = secret qualify=yes nat=no host=dynamic canreinvite=no disallow=all allow=alaw context=phones
The example above registers the local Asterisk PBX to the Cloud PBX system using extension 0003*001. Make sure the extension has SIP trunking enabled on the Cloud PBX server. 5000 is a local Asterisk extension that will be used for both incoming and outgoing calls.
Example of /opt/etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[phones]
exten => 5000,1,Dial(SIP/5000)
exten => _X.,1,Dial(SIP/${EXTEN}@mysip)
exten => _X.,2,Hangup
[from-voip-provider]
exten => 18002304043,1,Dial(SIP/5000)
exten => _X.,1,Congestion()
This is a basic example of Asterisk dialplan that can place outgoing calls and receive incoming calls. 5000 is the only local extension. The rest of the prefixes matching _X. get forwarded through the mysip channel. The[from-voip-provider] context is used for incoming calls.
To launch Asterisk, run the following:
/opt/sbin/asterisk -f -g -U asterisk -G asterisk
Test the configuration
For example, if you want to register the 5000 extension using a X-Lite softphone, you need to open its SIP accounts → Properties menu page and set:
- User name: 5000
- Password: secret
- Authorization user name: 5000
- Domain: asterisk_server_ip
To call a different extension (e.g. 0003*002) from the Asterisk PBX, you need to simply dial 0003*002).
To take incoming calls via Cloud PBX on extension 5000, the [from-voip-provider] context needs to be added. When configuring your /opt/etc/asterisk/extensions.conf file, you must replace the 18002304043 example with the DID number assigned to your account by your SIP provider.
Note that 18002304043 is a DID number assigned in the Cloud PBX interface for the specific extension with enabled SIP trunking.