Can someone please send me a document showing me how to configure a ta912 for sip to analog trunk?

[donwon]

I need to configure a ta912 connecting to at&t ipflex sip and bringing out analog trunks. I haven't been successful searching for anything useful

to help me. Thanks in advance. Don Register

[Anonymous]

donwon - I moved this post to the Total Access 900/900e Series area so someone from the community can help you there.

[Anonymous]

Hello Don,

We have a document that should give you a good starting point. - TA900 Sample Config - Basic SIP Requirements

The example configures the FXS voice user as:

voice user 1001

  connect fxs 0/1

  sip-identity 1001 T01 register auth-name USER1 password PASSWORD1

Keep in mind that in order for a SIP call to hit the ADTRAN and be routed to this voice user, the number in the SIP message To: field would need to be 1001, which matches the voice user.  Another way to do this is to use the DID as the voice user:

voice user 2565551001

  connect fxs 0/1

Now any calls that hit the ADTRAN for 256-555-1001 would be routed to the FXS port.  Sometime people will want to register the number, so it could look like this:

voice user 2565551001

  connect fxs 0/1

  sip-identity 2565551001 T01 register auth-name USER1 password PASSWORD1

Other times people want the DID and the ability to dial a shorter extension when making port to port calls:

voice user 1001

  connect fxs 0/1

  sip-identity 2565551001 T01 register auth-name USER1 password PASSWORD1

The DID in the sip-identity allows for 256-555-1001 in addition to 1001 to ring this extension.  Hopefully this gets you started.  Let us know if you have any questions.

Regards,

Geoff