[Asterisk-Users] Re: Nat...again...

Mark Woods asteriskadmin at fuse.net
Tue Jul 27 08:03:03 MST 2004


Thanks for your reply.

canreinvite has been set to "no" from the beginning...still no luck.

Maybe I'll be able to take a trace of it tonight...we'll see...but any thoughts at all are appreciated!

-Mark


> 
> Hi Mark,
> 
> Are you still having audio problems between outside SIP channels? Make
> sure that you have set the following for all SIP channels in your
> sip.conf
> 
> canreinvite=no
> 
> 
> -- sudhir
> > Message: 2
> > Date: Mon, 26 Jul 2004 22:46:22 -0400
> > From: Leif Madsen <leif.madsen at gmail.com>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] Nat...again....
> > Reply-To: asterisk-users at lists.digium.com
> > 
> > On Mon, 26 Jul 2004 15:14:51 +0000, Mark Woods <asteriskadmin at fuse.net> wrote:
> > > This has probably been answered somewhere, but I'm stumped.
> > > 
> > > I have two Zap channels (FXS and FXO), both working fine.  I
> > > can call from Zap/1 to Zap/2 and reverse.
> > > 
> > > I've also configured SIP channels, both inside and outside of my
> > > firewall.  Inside can call outside, and outside can call inside.
> > > Also, both inside and outside can make and receive calls to/from
> > > Zap/1 & Zap/2.
> > > 
> > > What doesn't work, and makes no sense to me given the above, is
> > > two outside SIP channels can connect but cannot pass traffic.  It
> > > would make sense to me if I couldn't already accept inbound/outbound
> > > SIP calls in some other way, but that isn't the case....
> > > 
> > > Thoughts?  My sip.conf is below (edited slightly for privacy):
> > 
> > Might not be worth much, but I haven't seen a reply yet.
> > 
> > Have you forwarded port 5060 to Asterisk?  Have you forwarded the port
> > range specified in rtp.conf?  By default it is 10000 -> 20000, but I
> > change mine to 10000 -> 10005.
> > 
> > 
> > As for your [general] section, I would maybe try something like this:
> > 
> > > [general]
> > > port = 5060                     ; Port to bind to
> > > context=sip-extensions
> > > externip = xxx.xxx.xxx.xxx
> > > localnet = 192.168.1.0/24
> > 
> > localmask is no longer used.  Not sure if there is backwords
> > compatibility (and I'm doing this from memory, I don't have Asterisk
> > behind NAT anymore)
> > 
> > I don't know, maybe it will work, maybe it won't :)
> > 
> > HTH,
> > Leif Madsen
> > http://www.asteriskdocs.org
> > 
> > --__--__--
> 
> 
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