Dear, I have a problem with my asterisk.
I have configured a gateway Dinstar for an E1 line, it works correctly when I configured the extension derived to sip phone
The problem arises when I set the asterisk to perform a playback or say digit, the call is cut, making a packet capture see that the Dinstar send me a 200 ok bye before the asterisk.
Any suggestions?
sory for my english .
Thank you so much