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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