asterisk users - Oct 2005 - toll free dialing problems using SIP

Hi all,

I have problems when a SIP terminal try to call a toll free number. This
is a call flow that explain what is going on (see comments below and inline):


     SIP terminal       Asterisk           NGW          Foo(tool free numb or
free message)
          |                |                |                |
          |  INVITE(SDP)   |                |                |
          |--------------->|  INVITE(SDP)   |                |
          |                |--------------->|                |
          |      100       |      100       |                |
          |<---------------|<---------------|                |
          |    180(why?)   |                |                |
          |<---------------|                |                |
          |                |                |     IAM        |
          |                |                |--------------->|
          |                |                |     ACM        |
          |                |    183(SDP)    |<---------------|
          |   no 183 ?!    |<---------------|                |
          |                |                |                |
          |                |                |  One Way Voice |
          |                |                |<===============|
          .
          .
          . RTP data is flowing from bob to Asterisk (checked with tcpdump).
          . RTP data is not forwarded by Asterisk to SIP terminal
          .
          . 30s timeout, SIP terminal keep ringing
          .
          .
          |                |                |                |
          |                |   CANCEL       |                |
          |                |--------------->|                |
          |                |     200        |                |
          |                |<---------------|     REL        |
          |                |                |--------------->|
          |                |                |     RLC        |
          |                |     487        |<---------------|
          |                |<---------------|                |
          |                |     ACK        |                |
          |                |--------------->|                |
          .
          .
          .

1) Why asterisk is sending 180 to SIP terminal? Did I configure * the wrong way?
2) Why 183 with SDP is not forwarded to the SIP terminal?

I have tried canreinvite=[yes|no] and progressinband=[yes|no] and
pedantic=[yes|no] in
sip.conf but still same behaviour occur. Did I missing something?

Thank you very much, I really need help

Ciao
FF