Dialing in from the pstn to sip phones (x-lite softphone on winders and a grandstream handytone-286 ata), I hear the sip phone ring a few times,
I ran into the same thing with Cisco 7960. Looks like the logic in the sip channel has changed recently.
Add a ",r" to the end of your Dial statements in extensions.conf and the issue should go away.
Does anyone know if this was done intentionally? I don't want to open a bug
for something that's really a feature, but i simply can't think of any reason
someone want's to update their whole extensions.conf.
Can someone tell me what i have to change in the source to get the old (correct :-) way ...?
Regards,
Andreas
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