[Asterisk-Users] asteisk, sip & NAT

[Jon Fautley]

>----- Original Message -----
>From: "Hervé Thibaud" <ht_asterisk at beltegeuse.org>
>To: "asterisk-users" <asterisk-users at lists.digium.com>
>Sent: Sunday, June 22, 2003 8:13 AM
>Subject: [Asterisk-Users] asteisk, sip & NAT


>hi
>My stations are behinds a firewall, the system is windows 2000 & 98, i
>use sjphone
>aterisk is on the internet gateway where is the firewall Shorewall the
>system is linux debian (sid) kernel 2.4.20
>j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy)
>to write my sip.conf but i can't call an external sip user. (an external
>user can call me)

This sound like the problem that I've been having this weekend. My setup is
a Snom100 and X-Lite connected to my * box, and the same box is the NAT
gateway for the devices. I could have external users call in no problem at
all, but when I tried to call out I got about 1/1.5 seconds of audio and
then all incoming audio died. the other end could hear me, however.

It turned out to be the fact that * sending reinvite requests to fwd, which
was then trying to connect directly to the snom100, and, obviously, failing
because it's behind NAT.

After much hair-pulling from myself and Andy, I stumbled across an unrelated
post that pointed me to the 'canreinvite=no' option. I stuck this in the
[fwd.pulver.com] section of the sip.conf file and magically, it all worked!

Maybe, just maybe, it'll work for you too :)

Jon