On Dec 10, 2007 2:08 PM, James Golovich <[EMAIL PROTECTED]> wrote: > I've continued on the previous work done on the SIP TCP/TLS branch and > it's ready for some additional testing. > > The branch is located at > http://svn.digium.com/view/asterisk/team/group/sip-tcptls/ >
James, This is cool. I've been testing it with a Snom 360 running 7.1.30: *CLI> core show version Asterisk SVN-group-sip-tcptls-r92242-/trunk built by kris @ krislap on a i686 running Linux on 2007-12-10 19:29:26 UTC *CLI> sip show peer snom * Name : snom Secret : <Set> MD5Secret : <Not set> Context : default Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 3028 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Trust RPID : Yes Send RPID : Yes Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 ToHost : Addr->IP : 10.16.5.237 Port 2060 Defaddr->IP : 0.0.0.0 Port 5060 Transport : TLS Def. Username: snom SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No 100 on REG : No Status : OK (25 ms) Useragent : snom360/7.1.30 Reg. Contact : sip:[EMAIL PROTECTED]:2060;transport=tls;line=1sz3a8qe *CLI> sip show tcp Host Port Transport Type 10.16.5.237 2077 TLS Server So far, so good! -- Kristian Kielhofner _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
