Building SIP Trunk to Metaswitch

We are wanting to use Asterisk as our voicemail server for our customers using POTS. I am completely new to Asterisk, so please let me know if I’m off base. I plan to deploy Asterisk in our core network, preferably on VMware.

From what I understand, I need to build a SIP trunk from Asterisk to my Meta. Then I would create user accounts in Asterisk with voicemail’s. The customer would dial into their voicemail to retrieve. We currently have voicemail servers but they are both pretty much dead and causing issues.

I’m having trouble building the SIP trunk so any guidance is appreciated

Note there are two places where you need to enter “Trunk Name” - you pick what that is, something like META_PHONENUMBER will do.

Specify Outbound Caller ID, it should be the phone number from MetaSwitch in 10-digits format.

In peer section of the trunk try using this as a template (just copy/paste and edit the host, user, and password)

host=address of the Metaswitch sip server
username=your metaswitch user
secret=your metaswitch user password

Clear the User Context and User Detail fields.

in “Register String” at the bottom put

<em>sipuser:[email protected]</em>

these are the same values as in “peer” section from above.

This should work for SIP trunk creating.

After that you need to created incoming and outgoing rules as you wish.