asterisk users - Nov 2014 - SIP call drops after 32 seconds, but only when....

Problem fixed. The issue was that the timeout for the Options packet was longer
than our NAT timeout.

Best wishes,
Norman
________________________________________
From: asterisk-users-bounces at lists.digium.com on behalf of Rusty Newton
Sent: Saturday, November 15, 2014 12:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Erratic calls through NAT-ed server

On Thu, Nov 13, 2014 at 8:23 AM, Norman Laidla
<norman.laidla at telegrupp.ee> wrote:
> Morning,
>
> We recently pushed our Asterisk video bridge into a DMZ and since then,
> local calls have been unreliable to say the least. While offsite calls work
> nicely, calls on our internal server usually fail to ring the far end. Two
> test calls that were made 4 minutes apart yielded different results: one
> rang the far end, the other kept trying to transmit the Invite. The
> configuration didn't change at all between the two calls. I've been
going
> over the debug logs, but haven't noticed any possible reasons why one
call
> failed. It's the same all the way to the part where the far end is
called.
>
> The endpoints use different ports for UDP signaling and Asterisk is set to
> expect UDP packets from those ports. The RTP port range is the same between
> the ends (at least where it's configurable), Asterisk and the firewall.
All
> ports that we're using have been opened in the firewall and incoming
UDP
> traffic is routed to Asterisk. In Asterisk settings, localnet is defined as
> the LAN that both endpoints are on, externip is the public address of the
> server. Directrtpsetup and directmedia are both set to "no" and
nat is set
> to "yes".
>
> So, what could be causing this issue?
If out of multiple calls, some work and some don't - you either have
found a bug or something is really changing between the calls. That is
assuming the failing/working behavior does not fit an obvious pattern
(e.g. unique to a particular dialed remote party).

If you pastebin two Asterisk logs that show the working and failing
calls then someone may be able to look through them and spot an issue.

Be sure the Asterisk logs show VERBOSE and DEBUG channels at level 5
or above. See:
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

You might also mention the exaction version of Asterisk you are using
and which channel driver (though it sounds like chan_sip based on the
options described).

--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

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hi,

I have a really strange problem which is driving me crazy for days now.

If I register my asterisk (tried all versions from 1.6 up to 13.x) with 
one sip registrar,
everything works... calls go out and call come in... no 32 seconds limit.

but as soon as I configure another sip registration on another server, 
outgoing
calls  drop after 32 seconds.

as far as I know, there is no firewall in between...

I tried to "work around" this by increasing the settings for
"timerb"...
but I
realized that asterisk does not care at all, what I set this value to...
"sip show settings" always gives me 32000ms, and it does not make any
difference if I configure timerb in the general context or in the phone 
context...

any ideas?

thanks,
yves

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