[asterisk-users] Strange initial RING

[Alexandru Oniciuc]

Hello list!

                I'm having a strange problem with the VoIP Gateway that I'm using to go on the PSTN: if the number on the other end is busy or unavailable I hear an initial RING, generated by Asterisk from what I'm seeing and after that the line goes down with busy signal:

Here is the scenario:

 Softphone                         *ASTERISK                                                         PATTON                                                             PSTN [BUSY CALLED EXTENSION]


1.       INVITE                 >             INVITE                                                 >             INVITE

2.                                                                                                                    <             SIP/2.0 100 Trying

3.        RING                                   SIP/2.0 180 Ringing                         <             SIP/2.0 183 Session Progress

4.                                                      SIP/2.0 603 Declined                      <             SIP/2.0 406 Not Acceptable

Is this regular? Asterisk isn't supposed to generate the RING  only after the first one received from the PATTON?

Asterisk version: 1.6.0.22

Thank you in advance for the support.

Best Regards,
Alex

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