[SFLphone] Cannot connect to asterisk via sip and a vpn

Jan Wiele jan at wiele.ath.cx
Wed Jan 20 11:57:27 EST 2010


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" <jan at wiele.ath.cx>
> À: sflphone at lists.savoirfairelinux.net
> 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:janextern at 10.8.0.1>;tag=785e03b4-0c58-444f-b89b-ae41cb0e072b
> To: <sip:janextern at 10.8.0.1>
> Call-ID: 3d999067-54fa-4dbe-8f23-458212ce93bd
> CSeq: 50460 REGISTER
> User-Agent: sflphoned/0.9.6
> Contact: "wielejan" <sip:janextern at 192.168.1.10:5060>
> 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:janextern at 10.8.0.1>;tag=785e03b4-0c58-444f-b89b-ae41cb0e072b
> To: <sip:janextern at 10.8.0.1>;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:janextern at 10.8.0.1>
> From: "Jan <sip:janextern at 10.8.0.1>;tag=uzhfj
> Call-ID: bgtvsywndvmqfai at joplaptop
> CSeq: 983 REGISTER
> Contact: <sip:janextern at 10.8.0.18>;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:janextern at 10.8.0.1>;tag=uzhfj
> To: "Jan <sip:janextern at 10.8.0.1>;tag=as1daaf748
> Call-ID: bgtvsywndvmqfai at 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 'bgtvsywndvmqfai at 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:janextern at 10.8.0.1>
> From: "Jan" <sip:janextern at 10.8.0.1>;tag=uzhfj
> Call-ID: bgtvsywndvmqfai at joplaptop
> CSeq: 984 REGISTER
> Contact: <sip:janextern at 10.8.0.18>;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:janextern at 10.8.0.18 SIP/2.0
> Via: SIP/2.0/UDP 10.8.0.1:5060;branch=z9hG4bK54785a89;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at 10.8.0.1>;tag=as3045c12c
> To: <sip:janextern at 10.8.0.18>
> Contact: <sip:asterisk at 10.8.0.1>
> Call-ID: 3a72c15d1584152f76f59e4d11a8233e at 10.8.0.1
> 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
> SFLphone at lists.savoirfairelinux.net
> 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


More information about the SFLphone mailing list