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authorTerry Wilson <twilson@digium.com>2011-11-02 23:08:46 +0000
committerTerry Wilson <twilson@digium.com>2011-11-02 23:08:46 +0000
commit7f883ef495b57ae9182e47213d01d5e8009dbf3f (patch)
treefea1e82631edc075f9ca3e8ffb19561a37d3e18c /channels/sip
parentf7bdc835a423fb7e969f69983cd615b2c4ff5e0c (diff)
Remove registertrying option in chan_sip
This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/sip')
-rw-r--r--channels/sip/include/sip.h9
1 files changed, 4 insertions, 5 deletions
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index c2924b710..a06605136 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -339,11 +339,10 @@
#define SIP_PAGE2_FAX_DETECT_T38 (2 << 23) /*!< DP: Fax Detection support - detect T.38 reinvite from peer */
#define SIP_PAGE2_FAX_DETECT_BOTH (3 << 23) /*!< DP: Fax Detection support - detect both */
-#define SIP_PAGE2_REGISTERTRYING (1 << 24) /*!< DP: Send 100 Trying on REGISTER attempts */
-#define SIP_PAGE2_UDPTL_DESTINATION (1 << 25) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
-#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 26) /*!< DP: Always set up video, even if endpoints don't support it */
-#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 27) /*< Are we associated with a configured peer context? */
-#define SIP_PAGE2_USE_SRTP (1 << 28) /*!< DP: Whether we should offer (only) SRTP */
+#define SIP_PAGE2_UDPTL_DESTINATION (1 << 24) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
+#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 25) /*!< DP: Always set up video, even if endpoints don't support it */
+#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 26) /*< Are we associated with a configured peer context? */
+#define SIP_PAGE2_USE_SRTP (1 << 27) /*!< DP: Whether we should offer (only) SRTP */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \