Do you need to migrate your existing SIP trunk to CloudPBX? Great!
Once this is complete, you'll experience:
- improved and automatic redundancy (multiple layers of redundancy are inherent in CloudPBX)
- no need to register more than one trunk
- improved call quality as all calls are u-law, with no transcoding at any layer
You'll be provided with the below configuration information:
- host / domain
- username
- password
- test phone number
Here is how to configure the SIP trunk in FreePBX / Elastix / Trixbox.
** Note that these instructions are for older versions of FreePBX.
On newer versions, note that ChanSIP is recommended. You also must currently disable ICE support with 'icesupport=no'.
1) Navigate to the 'Trunks' page in your UI.
2) Click 'add SIP trunk', or 'add trunk;

3) Many parts of the trunk configuration will be deployment - specific, such as the 'Dial Rules' and 'Trunk Name' - you can copy your existing configuration here, or set it as required by your deployment.
4) The Outgoing Settings should be as below:
TRUNK name: CloudPBX (or whatever you'd like)
allow=ulaw
context=from-trunk
disallow=all
username=USERNAME
secret=PASSWORD
host=account.cloudpbx.ca
insecure=very
type=peer
qualify=yes
sendrpid=no
dtmfmode=rfc2833
registertimeout=360
outboundproxy=OUTBOUNDPROXY
** "insecure=very" was removed as of Asterisk 1.6, so you probably want to use "insecure=port,invite"
** This OUTBOUNDPROXY rule is optional, we use it to ensure correct routing where Asterisk does not completely support DNS-SRV
** Please replace the USERNAME, PASSWORD, and account.cloudpbx.ca values with the corresponding values you have been provided.

5) The 'Incoming Settings' should be as below:
USER Context: USERNAME
allow=ulaw
context=from-trunk
disallow=all
host=account.cloudpbx.ca
insecure=very
type=user
dtmfmode=rfc2833
registertimeout=360
** "insecure=very" was removed as of Asterisk 1.6, so you probably want to use "insecure=port,invite". This is safe as you are establishing a trunk to a known entity.
** Please replace the USERNAME and 'account.cloudpbx.ca' values with the corresponding values you have been provided.
6) The 'Register String' should be as below:
USERNAME:PASSWORD@account.cloudpbx.ca
** Please replace the USERNAME, PASSWORD, and account.cloudpbx.ca values with the corresponding values you have been provided.

7) Click 'Submit Changes'.
8) CloudPBX sends calls down the trunk in translated E.164 format. Please route your test DID as, for example, just '6046383848'.
Please route your test DID, then apply all changes / reload, and check your sip peers - if you see that you are registered, try an inbound call!
Once you've confirmed that all is well, contact us at support@telephonic.ca to have your phone numbers re-routed.
** TWO additional steps:
1) Set outboundproxy= GATEWAY_DOMAIN on both inbound and outbound trunks.
2) Add a complete list of our SBCs to your /etc/hosts file, and restart networking.
Please contact us at support@telephonic.ca for a list of what your GATEWAY_DOMAIN should be, and what the complete list of our SBC gateways is.
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