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Hello,
I have a SIP gateway connected to my asterisk but
the thing is that when I try to use my analog phone line
and I hung up the gateway doesn´t hungs up the
line.
I think that when I call my gateway from a
softphone to get access to the analog line and I close the
communication
the asterisk server sends a tone to the gateway and
it doesn´t understand it. I want to know wich tone is it because
I can configure it from the gateway.
If anyone has an idea... I´d love to hear it...
thanks for your time,
Danko
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