[Asterisk-Users] 2 Sip phones behind un-natted Asterisk

[Shaun Dawson]

What does the Xten diagnostic log say about a single
session?

Also, what does the * SIP debug output say?  I'd be
very interested to see what IPs and ports SIP is
trying to set the RTP connection on.  (Since SIP
appears to be working fine, it's the RTP part that is
breaking).

Are both the Xten and the 7960 trying to listen on the
same RTP port (my Xten is configured to listen on
8000)?

Pardon me if I sound like an idiot, but I'm somewhat
new to VoIP, SIP _and_ Asterisk.  :)

Shaun


--- Bruce Komito <brucek at bagel.com> wrote:
> John, In my case, the two ports are not using the
> same IP port (one is on
> 5060, the other on 5061), but of course, they are on
> the same IP address.
> I think that is what is confusing the NAT server,
> but I don't know what to
> do about it.
> 
> Bruce Komito
> High Sierra Networks, Inc.
> www.servers-r-us.com
> (775) 284-5800 ext 115
> 
> 
> On Mon, 24 May 2004, John Fraizer wrote:
> 
> > Chad Brown wrote:
> >
> > > I have 2 SIP phones (Cisco 7960 & XTen) behind a
> NAT provided by a
> > > Linksys firewall that supports UPnP.  The
> Asterisk server has a public
> > > IP. Here are the problems that I am having with
> this configuration…
> > >
> > >
> > >
> > >    1. The 2 SIP phones can call MeetMe and have
> a conference but cannot
> > >       call each other. (Yes, they connect but no
> audio either direction)
> > >    2. I have verify=yes in the sip.conf for both
> phones. Both phones
> > >       constantly go Unreachable. (However, the
> connection is very fast
> > >       between * and sip phones)
> > >    3. Sometimes but not always when I try to
> call phone1 phone2 rings.
> > >
> > >
> > >
> > > Is this Nat messing with me or something else?
> In any case…Any advice
> > > out there?
> > >
> > >
> > >
> > > Thanks,
> > >
> > > Chad
> > >
> >
> >
> > The problem is probably that both of your SIP
> phones are using the same
> > port.  I played with two 7960's behind a Linksys
> on Saturday and finally
> > got them playing right when I changed the
> following:
> >
> > In Phone 1's SIP[macaddr].cnf:
> >
> > voip_control_port: 5061
> >
> > In Phone 2's SIP[macaddr].cnf:
> >
> > voip_control_port: 5062
> >
> > The default control port is 5060.  Note:  This is
> the port that the
> > PHONE uses to initiate the connection to * and not
> the port it is
> > connecting to.
> >
> > John
> > _______________________________________________
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> >
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