Hello, I've then same probleme with sip rtp packets with different Routers. This is perhaps not a vpn problem ! Sip with french "livebox" France telecom -> don(t work Sip with "Livebox pro" -> ok Sip with Bewan adsl router -> don(t work, but with last firmware, ok.
Their is a problem with ZIP Rtp packets in some routers.... A quick test: an ata286/fxs behind each routers to asterisk. Best regards Laurent BARTHELEMY Sirtem. -----Message d'origine----- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Francois BERGERET Envoyé : samedi 16 juillet 2005 14:42 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] VPN's Sure, I have more than 18 tunnels to manage here, and the only blocking effects are thuse that I have volontary encoded . ;-) I believe that Peter has missed something in the VPN parmeters themselves or not correctly understood how are his IPtables onto this two IPSec secure gateway... Peter, could you post us the content of your "/etc/ipsec.conf" file ? We can take a look here and verify what is not good. Best Regards, Francois BERGERET, France. ----- Original Message ----- From: "Shamsul Arefin" <[EMAIL PROTECTED]> To: "Francois BERGERET" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Friday, July 15, 2005 11:48 PM Subject: Re: [Asterisk-Users] VPN's Hi, We use firewall and VPN togather to connect around 5 remote sites, and never encounter these problems. Make sure that port 10,000 and above mentioned in ur rtp conf files are opened in ur vpn and firewall. also when u connect from remote site don't use public ip use privte behind firewall. If still have problem send me more detail and i will be more then happy to sort this out . Regards Shamsul Arefin Saktek Broadband telephony experts On 7/16/05, Francois BERGERET <[EMAIL PROTECTED]> wrote: > Hi men, > > You have some IP ports blocked ! > I use SuperFreeSwan and I encounter no problem with this kind of > configuration. > Do you have open all ports on your IPsec gateways ? > Think to have a look to your IPchains or any kind of firewall you are > running in your IPSec gateway. > I use shorewall and it is possible to miss some rules or to let pass > few ports only for protections between sites. > You must describe more your configurations to see what... > > Good luck ! > > Francois BERGERET, > [EMAIL PROTECTED], > France. > > ----- Original Message ----- > From: "Armin Schindler" <[EMAIL PROTECTED]> > To: "Peter Osborne" <[EMAIL PROTECTED]> > Cc: <[email protected]> > Sent: Friday, July 15, 2005 8:35 PM > Subject: Re: [Asterisk-Users] VPN's > > > > On Fri, 15 Jul 2005, Peter Osborne wrote: > >> Hi All, > >> > >> I'm using Asterisk for my PBX, I have a remote office that is > >> connected by a VPN link. I am using Openswan on my side and a > >> Linksys box on the remote side. I have a Polycom IP300 on the > >> remote side configured with a static IP address. When I call the > >> phone on the remote side, it rings and establishes the call fine. > >> The problem I am having is that the remote side can hear the call > >> find but the local side hears nothing. Because of the VPN there are > >> no firwalls in the way. Does anyone have some ideas or atleast how > >> I can track down the problem. > > > > I had the same problem with VPN using 'netscreen' (or a similar > > name) boxes. > > When I switched from SIP to IAX protocol, it worked perfectly. > > > > I think the SIP voice UDP packets are blocked somehow, but I didn't > > investigated it further. > > > > Armin > > _______________________________________________ > > 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 > -- Best Regards Shamsul Arefin Saktek , Broadband Telephony experts _______________________________________________ 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
