[asterisk-dev] sip-tcptls branch

[Pavel Jezek]

http://svn.digium.com/svn/asterisk/team/bbryant/sip-tcptls

1) I'm testing this branch with eyebeam softphone and sip over tcp,
annoying issue is, that peers are after some minutes unregistered 
without any reason (I must restart softphone to register with asterisk 
again)
logs are continuously filled with messages...

[Nov  9 07:25:57] NOTICE[9824] chan_sip.c: Trying to put 'OPTIONS si' 
onto UDP socket...
[Nov  9 07:25:58] NOTICE[9824] chan_sip.c: Trying to put 'OPTIONS si' 
onto UDP socket...
[Nov  9 07:25:58] NOTICE[9824] chan_sip.c: Trying to put 'OPTIONS si' 
onto UDP socket...
[Nov  9 07:25:58] NOTICE[9824] chan_sip.c: Trying to put 'SIP/2.0 50' 
onto UNKNOWN socket...
[Nov  9 07:25:58] WARNING[9824] chan_sip.c: sip_xmit of 0xb6f956a4 (len 
490) to 192.168.164.154:10173 returned -1: Socket operation on non-socket
[Nov  9 07:26:08] NOTICE[9824] chan_sip.c: Trying to put 'OPTIONS si' 
onto UDP socket...
[Nov  9 07:26:08] NOTICE[9824] chan_sip.c: Trying to put 'OPTIONS si' 
onto UDP socket...


2) It seems, that this branch is not synchronized with current trunk, 
eg. bug
http://bugs.digium.com/view.php?id=11175
was resolved in trunk, but persist in tcp/tls branch