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@gmail.com>
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Nat...again....
> Reply-To: asterisk-users@lists.digium.com
>
> On Mon, 26 Jul 2004 15:14:51 +0000, Mark Woods
<asteriskadmin@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
>
> --__--__--
