FreePBX/Asterisk dropping calls at 10 minutes

Our FreePBX 2.6.01 / Asterisk 1.4.29.1 system is consistantly dropping calls at 10 minutes.  I am hoping this is something very simple and obvious (just not to me).

It is stock, installed from an AsteriskNOW! download, with trunks from Speakeasy (I have contacted them about a timeout on their end).  The phones and asterisk server are on a physical and logical subnet that uses the firewall as the gateway.  We have a Sonicwall TZ-210 firewall.  I have attached an excerpt from the Asterisk 'full' log with an example of a conference call that was dropped every ten minutes (asteriskfull.txt).

The phones are Linksys 942 & 941.  

Thanks, Harold
hmaupinAsked:
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bwilks99Commented:
Hi, can you get the output from the cli when the call drops? I can not see asteriskfull.txt.

two things I would look at

1. nat firewall problem
2. is there an invite just before the call disconnects
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hmaupinAuthor Commented:
OK, kinda new to this, so run asterisk -r > myfile.txt to capture the CLI output to a file, yes?  Tried attaching the log text again.  Thanks!
asteriskfull.txt
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hmaupinAuthor Commented:
Well, here is a capture of the CLI with the same chunk of the log for time reference.  I was busy and did not find the exact moment it died, but after 20 minutes the phones did not communicate although they thought they were connected.  Will try to test more tomorrow evening.  Any thoughts on where else to look or suggestions as to tools to try would be appreciated, thanks!  


asteriskCLI-log.txt
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bwilks99Commented:
Hi, can you enable sip debugging and capture it again?

Usage: sip set debug
       Enables dumping of SIP packets for debugging purposes

       sip set debug ip <host[:PORT]>
       Enables dumping of SIP packets to and from host.

       sip set debug peer <peername>
       Enables dumping of SIP packets to and from host.
       Require peer to be registered.
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hmaupinAuthor Commented:
Sounds good, thanks.  So with the sip set debug (host) I use the trunk sip server ip as the (host), yes?  And if I enter sip set debug (phone ip) I get the conversation with the phone?  Thanks, this could be very useful!  I will not be back at that site until the afternoon, will post the trace then. Thanks!
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hmaupinAuthor Commented:
Some more information.  Ran a couple of tests to my cell phone, and set up a teliax account so I could dial out via our current vendor and have the call ring another extension via the teliax trunk.  The problem exhibits itself as a lost call at the called station, which properly indicates the call is terminated.  The calling station, our extension off of the asterisk server, remains 'connected' but is silent.  This is all SIP trunks, no DAHDI involved, although there is a DAHDI card in the system for future use.

I have attached several files with CLI SIP DEBUG information.  siptest3.txt is the raw data from the trace.  4014siptest.txt is the filtered (grep 4014) results of the call to my cell phone; 241siptest.txt is the filtered results (grep 241) of the originating extension; 223siptest.txt (grep 223) is the filtered results of the called extension.

Thanks!
siptest3.txt
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hmaupinAuthor Commented:
OK, here are the files in a .zip, oopppss.
speakeasy.zip
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bwilks99Commented:
Hi,

Thanks for the sample I have not had time to look at it.

Here are two links that may help as they solved a 10 minute drop for me on asterisk 1.4. In my case the provider made a change on their end that changed the session timers/RE-INVITE function.

http://forums.whirlpool.net.au/forum-replies.cfm?t=1235239&p=6#r113

http://forums.whirlpool.net.au/forum-replies.cfm?t=1235239&r=21511050#r21511050
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hmaupinAuthor Commented:
Thanks, I will pursue this with Speakeasy.  Do recall what their settings were that worked, perchance?

Thanks, hopefully this will resolve it!
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hmaupinAuthor Commented:
Got a Teliax trunk, limited testing indicates that calls placed on that trunk do not drop, at least up to 50 minutes.  Will turn up the heat on Speakeasy vis-a-vie your suggestion on the session timers/RE-INVITE function.  Thanks!
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hmaupinAuthor Commented:
Thanks, it was the re-invite, and once I pushed the vendor in that direction we figured out the problem.  Your suggestion and info was the catalyst that got everyone focused on the right area.  I will say dealing with Speakeasy was like pushing mud uphill, but once I got past the gate keepers and on to their real techs all was good.  Thanks again!
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gbarrientosCommented:
hmaupin, do you mind sending me a screen shot of your SIP trunk settings and your outbound routes for SpeakEasy. I am having a hard time working with their support staff.
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