[asterisk-users] Erratic calls through NAT-ed server

Rusty Newton rnewton at digium.com
Fri Nov 14 16:39:13 CST 2014

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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