[Asterisk-Users] (no subject)

[Andreas Anderson]

>>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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