Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
> canreinvite=no [general] port = 5060 srvlookup = yes nat = yes tos = lowdelay disallow = all allow = ulaw allow = gsm allow = alaw context= INVALID Currently my IP phones haves this in the sip.conf [4403] type= friend username= 4403 secret = password nat = yes host = dynamic context = toll-access accountcode = barry mailbox= 4403 dtmfmode = inband canreinvite= yes I'll try changing the can reinvite to NO Currently my setup is 192.168.0.0 192.168.1.0 IP Phone <-> Linux gateway <-> Internet & VPN connection <-> Linux gateway <-> * server <-> IP Phone The two networks are linked via internet and a VPN connect whereby I can ping the local address from either network Thanks all Barry ___ 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
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
Since I always use canreinvite=no, you're probably right. John Bruce Komito wrote: Not to beat a dead horse, but I had the problem even with the two lines on different ports. The canreinvite=no thing solved the problem. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, John Fraizer wrote: Bruce Komito wrote: > In sip.conf, try setting canreinvite=no for both lines. > > Bruce Komito > High Sierra Networks, Inc. > www.servers-r-us.com > (775) 284-5800 ext 115 canreinvite=no will sometimes make a difference but, I believe that what most people are running into is what I described in my previous post. NAT devices track IP flows. When you have two "flows" from the asterisk server that look like: ast.erisk.serv.er:5060<->nat.wan.ip.addr:5060<->NAT<->10.1.1.1:5060 ast.erisk.serv.er:5060<->nat.wan.ip.addr:5060<->NAT<->10.1.1.2:5060 The NAT device has no way of deciding which of the two or more SIP devices (hardphone or softphone) on the inside the flow is supposed to go to. It looks through it's flow database and sees that there are two potential candidates on the inside that could be the owner of the flow. When you make sure that your SIP devices behind the NAT use unique source ports, you end up with this: ast.erisk.serv.er:5060<->nat.wan.ip.addr:5061<->NAT<->10.1.1.1:5061 ast.erisk.serv.er:5060<->nat.wan.ip.addr:5062<->NAT<->10.1.1.2:5062 I guess that many people don't really truly understand how the NAT works on the raw level and it's a plug and play device to them. For that matter, I have been running large scale networks for nearly 16 hears and I got bit by the same thing because I assumed that the Cisco would do the "Right_Thing" (TM) and choose a random source port. Once I discovered that it didn't do that, it was a simple matter of telling it to do so. I suppose that this is also somewhat the fault of the NAT device (linksys/cisco) as well since it should have noticed that there were two devices on the inside of the NAT trying to create identical flows. It could have easilly done some mapping to change the source port on the WAN side of the connection of one of the flows. OK. This is the * list and not routing 701 so, I'll stop now. Suffice it to say that it is a good idea to have unique source ports if you have multiple devices on the inside contacting the same device:port on the outside. John Fraizer EnterZone, Inc http://www.enterzone.net/ ___ 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
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
Not to beat a dead horse, but I had the problem even with the two lines on different ports. The canreinvite=no thing solved the problem. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, John Fraizer wrote: > Bruce Komito wrote: > > > In sip.conf, try setting canreinvite=no for both lines. > > > > Bruce Komito > > High Sierra Networks, Inc. > > www.servers-r-us.com > > (775) 284-5800 ext 115 > > canreinvite=no will sometimes make a difference but, I believe that what > most people are running into is what I described in my previous post. > NAT devices track IP flows. When you have two "flows" from the asterisk > server that look like: > > ast.erisk.serv.er:5060<->nat.wan.ip.addr:5060<->NAT<->10.1.1.1:5060 > ast.erisk.serv.er:5060<->nat.wan.ip.addr:5060<->NAT<->10.1.1.2:5060 > > The NAT device has no way of deciding which of the two or more SIP > devices (hardphone or softphone) on the inside the flow is supposed to > go to. It looks through it's flow database and sees that there are two > potential candidates on the inside that could be the owner of the flow. > > When you make sure that your SIP devices behind the NAT use unique > source ports, you end up with this: > > ast.erisk.serv.er:5060<->nat.wan.ip.addr:5061<->NAT<->10.1.1.1:5061 > ast.erisk.serv.er:5060<->nat.wan.ip.addr:5062<->NAT<->10.1.1.2:5062 > > I guess that many people don't really truly understand how the NAT works > on the raw level and it's a plug and play device to them. For that > matter, I have been running large scale networks for nearly 16 hears and > I got bit by the same thing because I assumed that the Cisco would do > the "Right_Thing" (TM) and choose a random source port. Once I > discovered that it didn't do that, it was a simple matter of telling it > to do so. > > I suppose that this is also somewhat the fault of the NAT device > (linksys/cisco) as well since it should have noticed that there were two > devices on the inside of the NAT trying to create identical flows. It > could have easilly done some mapping to change the source port on the > WAN side of the connection of one of the flows. > > OK. This is the * list and not routing 701 so, I'll stop now. Suffice > it to say that it is a good idea to have unique source ports if you have > multiple devices on the inside contacting the same device:port on the > outside. > > John Fraizer > EnterZone, Inc > http://www.enterzone.net/ > > > ___ 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
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
Bruce Komito wrote: > In sip.conf, try setting canreinvite=no for both lines. > > Bruce Komito > High Sierra Networks, Inc. > www.servers-r-us.com > (775) 284-5800 ext 115 canreinvite=no will sometimes make a difference but, I believe that what most people are running into is what I described in my previous post. NAT devices track IP flows. When you have two "flows" from the asterisk server that look like: ast.erisk.serv.er:5060<->nat.wan.ip.addr:5060<->NAT<->10.1.1.1:5060 ast.erisk.serv.er:5060<->nat.wan.ip.addr:5060<->NAT<->10.1.1.2:5060 The NAT device has no way of deciding which of the two or more SIP devices (hardphone or softphone) on the inside the flow is supposed to go to. It looks through it's flow database and sees that there are two potential candidates on the inside that could be the owner of the flow. When you make sure that your SIP devices behind the NAT use unique source ports, you end up with this: ast.erisk.serv.er:5060<->nat.wan.ip.addr:5061<->NAT<->10.1.1.1:5061 ast.erisk.serv.er:5060<->nat.wan.ip.addr:5062<->NAT<->10.1.1.2:5062 I guess that many people don't really truly understand how the NAT works on the raw level and it's a plug and play device to them. For that matter, I have been running large scale networks for nearly 16 hears and I got bit by the same thing because I assumed that the Cisco would do the "Right_Thing" (TM) and choose a random source port. Once I discovered that it didn't do that, it was a simple matter of telling it to do so. I suppose that this is also somewhat the fault of the NAT device (linksys/cisco) as well since it should have noticed that there were two devices on the inside of the NAT trying to create identical flows. It could have easilly done some mapping to change the source port on the WAN side of the connection of one of the flows. OK. This is the * list and not routing 701 so, I'll stop now. Suffice it to say that it is a good idea to have unique source ports if you have multiple devices on the inside contacting the same device:port on the outside. John Fraizer EnterZone, Inc http://www.enterzone.net/ ___ 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
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
Barry Fawthrop wrote: 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. I'm having a similar problem with snom 200s would changing the port work there also or is that just a 7960 issue? Do you or any other know where I would change that on a snom 200 ?? thanks in advance Barry It's not phone specific. It's part the way your nat is working. It tracks things based on flows and a flow is sourceIP:sourcePort <-> destIP:destPort. When you have two devices inside your NAT sourcing from the same port and going to the same destIP:port, it doesn't know what to do with packets returning from the destIP:port because it has more than one flow that matches. So, if you have multiple devices (hard or soft phones) on the inside that are trying to register on the same sip proxy or * server, you need to make sure that they use unique source ports. 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
RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
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
RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
> > I'm having a similar problem with snom 200s would changing the port > work there also or is that just a 7960 issue? Do you or any other > know where I would change that on a snom 200 ?? > > thanks in advance > > Barry > ___ try adding Canreinvite=no To UA in question inside sip.conf! ___ 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
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
In sip.conf, try setting canreinvite=no for both lines. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, Barry Fawthrop wrote: > > 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. > > > > I'm having a similar problem with snom 200s would changing the port > work there also or is that just a 7960 issue? Do you or any other > know where I would change that on a snom 200 ?? > > thanks in advance > > Barry > ___ > 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
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
> 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. > I'm having a similar problem with snom 200s would changing the port work there also or is that just a 7960 issue? Do you or any other know where I would change that on a snom 200 ?? thanks in advance Barry ___ 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
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
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
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
Re: [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 ___ 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
RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
I'm not the original poster, but I have the same problem with a Sipura. In my configuration, I have line 1 set to port 5060 and line 2 set to port 5061. I assume that is what you are suggesting, right? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, Senad Jordanovic 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. > > Have you tried to make sure that each user agent use differnet sip port? > > ___ > 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
RE: [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? ___ 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
Re: [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 > > ___ 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
Re: [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 ___ 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] 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… The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 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) 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
