Good morning, i’ve created an IVR with a distro pbx12 and asterisk v11. It is made with an 8 choice ivr, each choice leading to 3 more choices. I’ve set a trunk sip, whit qualify=no because the calls were interrupted after few seconds now the calls are interruppted after 32 seconds. I’ve Read that could be a time out issue. How can solve this?
Get sufficient logging to actually know what the problem is/ Only then will it be possible to be specific about what needs fixing and how.
However the best guess is that Asterisk either hasn’t been told its external address for NAT, and is dropping the call when it fails to receive an ACK because the ACK is being sent to an address that is unreachable from the client.
Fixed issue, when setting the NAT I was wrong the last digit, instead of a 1 I put a 2. Then it all started to perfection
Thank you all for the support to solve the problem
Good afternoon, the reason it dropped the call after 30 seconds is that your timeout for RTP activity is at 30. This means it was not receiving any VOICE packets from the person you called. Since you did not have a connection properly with your NAT this happens. To me this is a bug in asterisk that you can make a call using SIP via UDP but then the VOICE datagrams get lost.
Asterisk cannot know the details of your NAT and firewall environment. You have to provide that information to it.
Not necessarily. 32 sec == 64*T1 (where T1 is one of multiple SIP timers). In most cases it’s a signaling (SIP) issue.