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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100319/4399a0f5/attachment.htm
On Fri, 19 Mar 2010, Alexandru Oniciuc wrote:> 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:Do you have the 'r' parameter in your Dial() instruction? Gordon