Alexandre Savard wrote: > Hi, > > We worked on this issue very recently (beginign of january), so I recommand > you to use the nightly built version if you are not already using it. > > https://launchpad.net/~savoirfairelinux/+archive/sflphone-nightly > > > Concerning your problem, > > SFLphone creates a default SIP UDP transport at startup on port 5060. If you > want to change this transport's settings, you need to go in Edit > Preference > > Ip Calls and then restart SFLphone. > > Once the default transport is created, SFLphone loads your account settings, > and create new transport if required. If a transport is already created on a > given port, no new transport will > be created and the current one will be used. > > To make sure that a new SIP UDP transport with your requested settings will > be created for an account, you should use a specific port number other than > the default 5060. > > OR, if you want to use your VPN for all your accounts, make the change for > the default transport and make sure that all your accounts use the default > transport's port number. > > Thanks for this report, and give us some news about it. > > Alexandre Savard > > > > ----- Mail Original ----- > De: "Jan Wiele" <[email protected]> > À: [email protected] > Envoyé: Mardi 19 Janvier 2010 18h30:19 GMT -05:00 USA/Canada - États de l'Est > Objet: [SFLphone] Cannot connect to asterisk via sip and a vpn > > Hi, > I'm trying to connect to my asterisk server via a OpenVPN. > I have the following ips: > "eth0" -> 192.168.1.10 > "tun0" -> 10.8.0.18 > > The OpenVPN server has the adress 10.8.0.1. > > When I try to connect asterisk logs this: > > ######################################### > <--- SIP read from UDP://10.8.0.18:5060 ---> > REGISTER sip:10.8.0.1 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.1.10:5060;rport;branch=z9hG4bKPjafdb9f53-fbf5-4294-aa2d-e9614a45480b > Max-Forwards: 70 > From: <sip:[email protected]>;tag=785e03b4-0c58-444f-b89b-ae41cb0e072b > To: <sip:[email protected]> > Call-ID: 3d999067-54fa-4dbe-8f23-458212ce93bd > CSeq: 50460 REGISTER > User-Agent: sflphoned/0.9.6 > Contact: "wielejan" <sip:[email protected]:5060> > Expires: 600 > Content-Length: 0 > > > <-------------> > --- (11 headers 0 lines) --- > Sending to 192.168.1.10 : 5060 (no NAT) > > <--- Transmitting (no NAT) to 192.168.1.10:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 192.168.1.10:5060;branch=z9hG4bKPjafdb9f53-fbf5-4294-aa2d-e9614a45480b;received=10.8.0.18;rport=5060 > From: <sip:[email protected]>;tag=785e03b4-0c58-444f-b89b-ae41cb0e072b > To: <sip:[email protected]>;tag=as0c960378 > Call-ID: 3d999067-54fa-4dbe-8f23-458212ce93bd > CSeq: 50460 REGISTER > Server: Asterisk PBX 1.6.1.9 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69154389" > Content-Length: 0 > ######################################### > > With Twinkle it is working: > ######################################### > <--- SIP read from UDP://10.8.0.18:5060 ---> > REGISTER sip:10.8.0.1 SIP/2.0 > Via: SIP/2.0/UDP 10.8.0.18;rport;branch=z9hG4bKfcchtzqc > Max-Forwards: 70 > To: "Jan <sip:[email protected]> > From: "Jan <sip:[email protected]>;tag=uzhfj > Call-ID: bgtvsywndvmq...@joplaptop > CSeq: 983 REGISTER > Contact: <sip:[email protected]>;expires=3600 > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE > User-Agent: Twinkle/1.4.2 > Content-Length: 0 > > > <-------------> > --- (11 headers 0 lines) --- > Sending to 10.8.0.18 : 5060 (no NAT) > > <--- Transmitting (no NAT) to 10.8.0.18:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 10.8.0.18;branch=z9hG4bKfcchtzqc;received=10.8.0.18;rport=5060 > From: "Jan <sip:[email protected]>;tag=uzhfj > To: "Jan <sip:[email protected]>;tag=as1daaf748 > Call-ID: bgtvsywndvmq...@joplaptop > CSeq: 983 REGISTER > Server: Asterisk PBX 1.6.1.9 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="510cbd7b" > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog 'bgtvsywndvmq...@joplaptop' in > 32000 ms (Method: REGISTER) > freeman*CLI> > <--- SIP read from UDP://10.8.0.18:5060 ---> > REGISTER sip:10.8.0.1 SIP/2.0 > Via: SIP/2.0/UDP 10.8.0.18;rport;branch=z9hG4bKfumtltvv > Max-Forwards: 70 > To: "Jan" <sip:[email protected]> > From: "Jan" <sip:[email protected]>;tag=uzhfj > Call-ID: bgtvsywndvmq...@joplaptop > CSeq: 984 REGISTER > Contact: <sip:[email protected]>;expires=3600 > Authorization: Digest > username="janextern",realm="asterisk",nonce="510cbd7b",uri="sip:10.8.0.1",response="6f011aa54d6aada0324ff92481f0bd8b",algorithm=MD5 > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE > User-Agent: Twinkle/1.4.2 > Content-Length: 0 > > > <-------------> > --- (12 headers 0 lines) --- > Sending to 10.8.0.18 : 5060 (no NAT) > -- Registered SIP 'janextern' at 10.8.0.18 port 5060 > Reliably Transmitting (no NAT) to 10.8.0.18:5060: > OPTIONS sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 10.8.0.1:5060;branch=z9hG4bK54785a89;rport > Max-Forwards: 70 > From: "asterisk" <sip:[email protected]>;tag=as3045c12c > To: <sip:[email protected]> > Contact: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 1.6.1.9 > Date: Tue, 19 Jan 2010 23:11:49 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > ######################################### > > It looks like slfphone uses the wrong ip-adress ("Via: SIP/2.0/UDP > 192.168.1.10:5060").i've set tun0 as the adapter to use. > When watching the traffic with wireshark between sflphone and asterisk > on the client side, no package appears from the server. sflphone only > repeats its register packages. > > Any ideas? > > Regards, Jan > > _______________________________________________ > SFLphone mailing list > [email protected] > http://lists.savoirfairelinux.net/mailman/listinfo/sflphone > Hi Alexandre, thanks for your fast reply. I added the nightly ppa to my sources.list but currently sfl-common (snapshot20100119 and snapshot20100118) fails to build.
>> configure: error: You need the Perl-Compatible Regular Expressions library >> (pcre) >> make: *** [configure-stamp] Error 1 >> dpkg-buildpackage: error: debian/rules build gave error exit status 2 So I downloaded snapshot20100117, changed the transport to tun0, restarted and it worked :)! Thanks. Jan _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
