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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040524/301f32d2/attachment.htm
Stephen Davies
2004-May-24 01:25 UTC
[Asterisk-Users] 2 Sip phones behind un-natted Asterisk
On Mon, 24 May 2004, Chad Brown wrote:> 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?Yes - I think your NAT firewall is messing with you. I suspect that if you configure the two phones in different ports - IE move one away from 5060, then you'll probably unconfuse your firewall. Steve
Bruce Komito
2004-May-24 05:38 UTC
[Asterisk-Users] 2 Sip phones behind un-natted Asterisk
I am having exactly the same problem with two phnes connected to a Sipura behind a Linksys. I'm sure this is NAT, because it works fine when I move the Sipura out from behind the Linksys. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, 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 > >
Senad Jordanovic
2004-May-24 06:35 UTC
[Asterisk-Users] 2 Sip phones behind un-natted Asterisk
>> 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.Have you tried to make sure that each user agent use differnet sip port?
John Fraizer
2004-May-24 07:46 UTC
[Asterisk-Users] 2 Sip phones behind un-natted Asterisk
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
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: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Shaun Dawson Sent: Monday, May 24, 2004 10:03 AM To: asterisk-users@lists.digium.com 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 <brucek@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 > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > >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 > Asterisk-Users@lists.digium.com >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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users