But the INVITE you posted at http://pastebin.ca/673392 also has only one Record-Route header. regards klaus David Loh schrieb: > Hi, > > Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK > ...but when asterisk disconnected the call and send BYE back to > OpenSER, > the TLS RR header wasn't present, the only 2 RR header was > "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" .... > I'm puzzled ... is there any command to 'fix' this? > > > Regards, > David Loh > > Klaus Darilion wrote: >> The openser proxy should add 2 record-route header (TLS and UDP = >> double record route). This is why it does not work. >> >> regards >> klaus >> >> David Loh schrieb: >>> Hi All, >>> >>> Greeting. >>> >>> I've been struggle with OpenSER TLS implementation for more than >>> a week, since I've ported from UDP to TLS, everything work fine >>> except the "BYE" request from Asterisk (loose route), my >>> implementation was something like below: >>> >>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] >>> >>> My OpenSER.cfg already configured to listen on two port which is >>> :- "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN >>> (or even voicemail) having no problem, >>> but when the callee disconnect the call, caller will never get >>> hang up :( >>> >>> I've attached my ethereal trace/ngrep to pastebin, >>> http://pastebin.ca/673392 >>> >>> Wondering if anyone can help me with the broken "BYE" that >>> returned from Asterisk ? >>> Line #131, supposedly this line should have contain 2 Via header, >>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", >>> but somehow the TLS via header was gone !! (compare to previous >>> ACK (Line #117) /INVITE (Line #51). >>> Due to the missing TLS via header, OpenSER log file was >>> complaining "protocol/port mis-match". >>> >>> The last BYE request (Line #256) is actually firing from Client, >>> which contain the "TLS" via. >>> >>> >>> I've even tried "force_send_socket" to port 5061 (instead of >>> 5060) from loose route, but it complaining TLS certificate error, >>> since Asterisk doesn't support TLS natively, I've no clue why is >>> the ACK/INVITE/CANCEL work but not BYE. >>> if (loose_route) { >>> .... >>> if(is_method("BYE")) { force_send_socket(IP:5061); } >>> } >>> >>> >>> Has any one gone through of this kinda OpenSER over TLS + >>> Asterisk setup, >>> I'm really appreciate if you can share your experience with me, >>> or pin point what's the mistakes I made here. >>> >>> Thanks in advance. >>> >>> Regards, >>> David Loh >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >

Route headers are fine - the problem is the RURI of the BYE: See the Contact header of the INVITE: Contact: <sip:davidloh@x.x.80.178:4294;transport=TLS> This URI must be used in the RURI of the BYE, but Asterisk uses: BYE sip:davidloh@x.x.80.178:4294 SIP/2.0 Thus, the proxy forwards the request with UDP instead of TLS. Thus, this is a bug in Asterisk. Try update Asterisk. Try looking at Asterisk Bug tracker for this bug. If you are unlucky, open a bug report on the Asterisk bug tracker (bugs.digium.com) regards klaus David Loh schrieb: > Hi, > > Arrggghh .. that's one of my attempts to eliminate the broken > "BYE" problem... that's ngrep was captured when I set > "modparam("rr", "enable_double_rr", "0");", > I've paste another ngrep to http://pastebin.ca/674450, this time > the double RR header is enabled. > And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to > decrypt the post is "openser"). > > Even though double RR header is enabled, but for BYE it's still > doesn't process properly :( > For the .cfg file line #130 onward, I did tried t_relay, forward > and force_send_socket, > but none of this will do the trick (force_send_socket was > complaining TLS error due to missing certificate (?) ) > Would appreciate if anyone could enlighten me why is this happen ? > > > Thanks, > David Loh > > > > Klaus Darilion wrote: >> But the INVITE you posted at http://pastebin.ca/673392 also has >> only one Record-Route header. >> >> regards >> klaus >> >> David Loh schrieb: >>> Hi, >>> >>> Yea, OpenSER proxy was add 2 record-route header for the >>> INVITE/ACK ...but when asterisk disconnected the call and send >>> BYE back to OpenSER, >>> the TLS RR header wasn't present, the only 2 RR header was >>> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" .... >>> I'm puzzled ... is there any command to 'fix' this? >>> >>> >>> Regards, >>> David Loh >>> >>> Klaus Darilion wrote: >>>> The openser proxy should add 2 record-route header (TLS and UDP >>>> = double record route). This is why it does not work. >>>> >>>> regards >>>> klaus >>>> >>>> David Loh schrieb: >>>>> Hi All, >>>>> >>>>> Greeting. >>>>> >>>>> I've been struggle with OpenSER TLS implementation for more >>>>> than a week, since I've ported from UDP to TLS, everything >>>>> work fine except the "BYE" request from Asterisk (loose >>>>> route), my implementation was something like below: >>>>> >>>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] >>>>> >>>>> My OpenSER.cfg already configured to listen on two port which >>>>> is :- "tls:eth0:5061" and "udp:eth0:5060", client make p2p or >>>>> PSTN (or even voicemail) having no problem, >>>>> but when the callee disconnect the call, caller will never get >>>>> hang up :( >>>>> >>>>> I've attached my ethereal trace/ngrep to pastebin, >>>>> http://pastebin.ca/673392 >>>>> >>>>> Wondering if anyone can help me with the broken "BYE" that >>>>> returned from Asterisk ? >>>>> Line #131, supposedly this line should have contain 2 Via >>>>> header, one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", >>>>> but somehow the TLS via header was gone !! (compare to >>>>> previous ACK (Line #117) /INVITE (Line #51). >>>>> Due to the missing TLS via header, OpenSER log file was >>>>> complaining "protocol/port mis-match". >>>>> >>>>> The last BYE request (Line #256) is actually firing from >>>>> Client, which contain the "TLS" via. >>>>> >>>>> >>>>> I've even tried "force_send_socket" to port 5061 (instead of >>>>> 5060) from loose route, but it complaining TLS certificate error, >>>>> since Asterisk doesn't support TLS natively, I've no clue why >>>>> is the ACK/INVITE/CANCEL work but not BYE. >>>>> if (loose_route) { >>>>> .... >>>>> if(is_method("BYE")) { force_send_socket(IP:5061); } >>>>> } >>>>> >>>>> >>>>> Has any one gone through of this kinda OpenSER over TLS + >>>>> Asterisk setup, >>>>> I'm really appreciate if you can share your experience with >>>>> me, or pin point what's the mistakes I made here. >>>>> >>>>> Thanks in advance. >>>>> >>>>> Regards, >>>>> David Loh >>>>> >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users(a)openser.org >>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >> >> > >