However I did notice the following in the logs Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:

8F980980E97E42F8EC;nat=yes>

The returning party in this case would be the PSTN GW and it talks

I.e. Softphone --TLS-->Openser ---UDP--->Softphone Should't Openser be responsible for the translation of these parameters?

Sorry my fault..I think got brain busted.. The picture should be Softphone --TLS-->Openser ---UDP-->PSTNGW

Hi All I am using using openser 1.3..if I make a call between two softphones on the same lan or a a pstn call to my mobile phone..and the called/receiver party does hang-up the call. It works fine in UDP mode and the call get’s hang-up. However in TLS mode this does not work. Anything I might have missed here? Since both udp and tls use the same routes, and voice is fine and no one way audio ..etc. I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the logs of openser I can’t see any errors. However on the wire shark I can see icmp destination unreachable…port unreachable. I would have said it is a NAT issue. However it works for simple UDP. However I did notice the following in the logs Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000 Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794 (45900000) Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler: retransmission_handler : done Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request: Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri: <sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B2008F980980E97E42F8EC;nat=yes> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232, <branch> = Shouldn’t the transport=TLS ? Regards ------------------------------------------------------------------------ _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

Hi All I am using using openser 1.3..if I make a call between two softphones

the same lan or a a pstn call to my mobile phone..and the called/receiver party does hang-up the call. It works fine in UDP mode

and the call get's hang-up. However in TLS mode this does not work. Anything I might have missed here? Since both udp and tls use the same

routes, and voice is fine and no one way audio ..etc. I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in

logs of openser I can't see any errors. However on the wire shark I can see icmp destination

unreachable. I would have said it is a NAT issue. However it works for simple UDP. However I did notice the following in the logs Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000 Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794 (45900000) Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler: retransmission_handler : done Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request: Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:

Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232,

<branch> = Shouldn't the transport=TLS ? Regards

_______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

Dear All Please find below the call setup from my softphone to my cell phone,

setup is as follows: LAN -> Office Gateway <--TLS--->Openser<--UDP-->PSTN GW I got the trace below by applying ngrep -W byline -T username -q -d eth0 http://pastebin.com/m2a78a27f I did the same trace for a udp call and it seemed identical to me, as you can see in the lower part of the trace that a BYE packet is being sent to the softphone however the transport is being indicated as UDP not TLS..is this normal ? Any clues apart from that ? Thanks -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: 2008-08-28 18:13 To: Ali Jawad Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS Hi! Send us an ngrep dump: ngrep -P "" -W byline port 5060 Although this will show us only the UDP part (as TLS is encrypted) but

may still show as the problem. Which SIP client do you use? regards klaus Ali Jawad schrieb: on > the same lan or a a pstn call to my mobile phone..and the > called/receiver party does hang-up the call. It works fine in UDP

> and the call get's hang-up. However in TLS mode this does not work. > Anything I might have missed here? Since both udp and tls use the

the unreachable...port

8F980980E97E42F8EC;nat=yes> > Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> > > Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 > > Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type

Ali Jawad schrieb: The Contact is the contact of the caller. Thus, there should be the IP:socket:protocol which is used by the caller (TLS). klaus _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users