Asterisk RTA sip call not connecting

We have just set up a new asterisk voicemail server. we have the odbc talking. we are able to do queries from isql and they work find. when we we place a call to the number of the sip client this is what we see in the logs. does anyone have any idea what might be going on.

[May 29 09:57:28] DEBUG[19505] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[May 29 09:57:28] DEBUG[19505] chan_sip.c: T38 state changed to 0 on channel <none>
[May 29 09:57:28] DEBUG[19505] chan_sip.c: We're settling with these formats: 0x4 (ulaw)
[May 29 09:57:28] DEBUG[19505] chan_sip.c: Checking SIP call limits for device
[May 29 09:57:28] DEBUG[19505] chan_sip.c: Updating call counter for incoming call
[May 29 09:57:28] DEBUG[19505] chan_sip.c: Updating call counter for incoming call
[May 29 09:57:28] DEBUG[19505] chan_sip.c: = Found Their Call ID: 1757472007-83946516532905758@208.x.x.x Their Tag GR52RWG346-34 Our tag: as552b2e30
[May 29 09:57:28] DEBUG[19505] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[May 29 09:57:28] DEBUG[19505] chan_sip.c: Stopping retransmission on '1757472007-83946516532905758@208.x.x.x' of Response 1: Match Not Found
[May 29 09:57:28] VERBOSE[19505] logger.c: [May 29 09:57:28] Really destroying SIP dialog '1757472007-83946516532905758@208.x.x.x' Method: ACK

[/code]