After further investigation it looks like it was as simple as both phones trying to listen on the same port. I will continue testing to verify.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shaun Dawson Sent: Monday, May 24, 2004 10:03 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk 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 <[EMAIL PROTECTED]> 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 > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users __________________________________ Do you Yahoo!? Yahoo! Domains - Claim yours for only $14.70/year http://smallbusiness.promotions.yahoo.com/offer _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
