Here is a scrubbed working configuration for an Adtran TA924 SIP connection to an Asterisk server with a couple of noteworthy points:

• The internal feature codes of the Adtran have been disabled with the “voice feature-mode network” command. As long as your Adtran’s internal dialplan supports it, feature codes can be passed through to Asterisk.
• With the “accept $ cost 0” statement on the “NETWORK” trunk group, the Adtran dialplan simply passes off all traffic to the network.
• Three way calling has been disabled with the “voice call-appearance-mode single” command.
• Call waiting has been disabled with the “no call-waiting” command per SIP registration.
• G711u is the only codec enabled by choice.
• There is an example of how to connect an FXS port to a SIP user.
• There is an example of how to set transmit/receive gains on an FXS port.
• There is an example of how to register an extension range of 7000-7023 to an Asterisk server.

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