On Wed, 2004-04-14 at 07:55, I wrote:
> Did any of you ever experience missing "180 ringing" messages when
> dialing from a sip agent through an asterisk box? 

Reading up some more on this it now appears to me this is RFC compliant
behaviour as a 183 Session Progress is being sent instead. So, my
question really is: Is there a way to force asterisk to answer with 180
ringing?

Thilo

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