diff options
| author | Russell Bryant <russell@russellbryant.com> | 2008-11-11 16:07:36 +0000 |
|---|---|---|
| committer | Russell Bryant <russell@russellbryant.com> | 2008-11-11 16:07:36 +0000 |
| commit | 72d5d58069b930da8a451b45a922cb849bde7b18 (patch) | |
| tree | 062a078c950746bc6d567eabc65247e6f42dab61 /channels | |
| parent | b07eba0c150c2ff04054a6a2f581cad9170347cc (diff) | |
Remove commentary from the issues list for SIP TCP/TLS
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 15 |
1 files changed, 3 insertions, 12 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d3d79fd08..3a0908fb0 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -87,18 +87,9 @@ * the sip_hangup() function */ -/*! \page sip_tcp_tls SIP TCP and TLS support - * The TCP and TLS support is unfortunately implemented in a way that is not - * SIP compliant and tested in a SIP infrastructure. We hope to fix this for - * at least release 1.6.2. This code was new in 1.6.0 and won't be fixed for - * that release, due to the current release policy. Only bugs compared with - * the working functionality in 1.4 will be fixed. Bugs in new features will - * be fixed in the next release. As 1.6.1 is already in release - * candidate mode, there will be a buggy SIP channel in that release too. - * - * If you have opinions about this release policy, send mail to the asterisk-dev - * mailing list. - * +/*! + * \page sip_tcp_tls SIP TCP and TLS support + * * \par tcpfixes TCP implementation changes needed * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more * \todo Save TCP/TLS sessions in registry |