TIPS Call disconnects at 15 minutes.

goblin

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Details:

PIAF Installed Version = 2.0.6.5 under *HARDWARE* │
FreePBX Version = 2.11.0.32
Running Asterisk Version = 1.8.26.1
Asterisk Source Version = 1.8.26.1
Dahdi Source Version = 2.9.0
Libpri Source Version = 1.4.14
Operating System = CentOS release 6.5 (Final)
Kernel Version = 2.6.32-431.1.2.0.1.el6.i686 - 32 Bit


I currently have a pbx that every call gets disconnected at 17 minutes i've been having this issue and cant seem to find the solution. Here is a part of the log when the call gets disconnect.


Any help will be much appreciated.


Code:
From: <sip:[email protected]:5060;user=phone>;tag=as5a8c4126
To: <sip:[email protected]:5060>;tag=c413409a07
Call-ID: e1304eb14505575e
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(1.8.26.1)
Proxy-Authorization: Digest username="800", realm="asterisk", algorithm=MD5, uri="sip:192.168.168.250", nonce="", response="7f1288070cc353989235ce210621a7d0"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
 
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You'd have to put up a sanitized trunk config, name your provider, name router model, describe firewall - port forward settings, static ip?. There are lots of hits on google for this with a variety of causes. It's been covered here too. In many cases its a router - firewall / sip ALG issue (turn ALG off). Sip Debug and wireshark are sometimes used to find the issue. Changing different keepalives sometimes solves the problem. Firewall maybe closing after 15 min.
 

goblin

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Hi there Brian,

I have a private circuit directly to the carrier they installed a router in my office, they just give me an RJ45 and i setup my trunk on my end, so basically there ir no port forward on my side.
 

krzykat

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I recall having a system that would cut out at exactly 30 minutes. What I had to do was go to Settings->Asterisk SIP Settings->Chan SIP, and then add to advanced general settings, Other SIP Settings: session-timers = refuse
I bet that gets it for you.
 

goblin

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I recall having a system that would cut out at exactly 30 minutes. What I had to do was go to Settings->Asterisk SIP Settings->Chan SIP, and then add to advanced general settings, Other SIP Settings: session-timers = refuse
I bet that gets it for you.

Already did this and still i have the same issue..
 
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Assuming sip debug doesn't help you any further....You haven't provided a trunk config. Have you made changes to any other files.? Because you have this kind of setup I think you are going to need to become a Wireshark expert which should take a couple of day of playing. You'll either have to generate a file on the server or use a laptop that is bridged in or get a 'tap'. Check the Wireshark site and wiki.
 

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