Ali Jawad schrieb:
Hi Klaus
You are referring to line 17 right ? That part of traffic is from openser to the pstn gw ..and both of those are UDP..should it be transport=tls there or transport=udp?
The Contact is the contact of the caller. Thus, there should be the IP:socket:protocol which is used by the caller (TLS).
klaus
Thanks
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +961 1 373725
Fax: +961 1 375554
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: 2008-08-29 14:34 To: Ali Jawad Cc: users@lists.kamailio.org Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
- INVITE:
Contact: sip:username@IP.OF.LAN.GW:2949;transport=UDP;rinstance=D98C1DD404B2008F 980980E97E42F8EC;nat=yes.
As you see the caller announces UDP as contact. Either a bug in the caller client or do you rewrite the contact in openser?
Further it is strange that the caller sends frmo 127.0.0.1 (Via header, SDP) but announces a different IP in contact.
regards klaus
Ali Jawad schrieb:
Dear All
Please find below the call setup from my softphone to my cell phone,
The
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
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@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:
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=D98C1DD404B200
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
232,
<branch> =
Shouldn't the transport=TLS ?
Regards
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