Merged revisions 373501,373505 via svnmerge from

file:///srv/subversion/repos/asterisk/branches/10

................
  r373501 | rmudgett | 2012-09-24 17:11:01 -0500 (Mon, 24 Sep 2012) | 18 lines

  Be consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid>

  When setting CALLERID(pres)=unavailable in the dialplan, the From header
  in the SIP message contains "Anonymous" <sip:Anonymous@anonymous.invalid>.
  For consistency, Asterisk should use a lowercase a in the userpart of the
  URI.

  * Make the From header use a lowercase A in the userpart of the anonymous
  URI.

  (closes issue ASTERISK-19838)
  Reported by: Antti Yrjola
  Patches:
        chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola
  ........

  Merged revisions 373500 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r373505 | mjordan | 2012-09-24 17:17:02 -0500 (Mon, 24 Sep 2012) | 19 lines

  Revert change to res_rtp_asterisk committed in r373236 (1.8)

  The change committed in r373236 attempted to account for endpoints that
  increased their RTP timestamp in DTMF end of event re-transmissions.  This
  change attempted to make Asterisk continue to work with endpoints that
  failed to follow the RFC while maintaining the fix that allowed for out of
  order DTMF to be handled.  Unfortunately, there is no free lunch, and this
  patch broke any system that sent DTMF immediately after an RTP session was
  established or when an SSRC is updated.  As such, that patch is being
  reverted for the previous behavior.

  Endpoints that erroneously increase the RTP timestamp in DTMF end of event
  packets will not work properly with Asterisk.

  (issue ASTERISK-20424)
  ........

  Merged revisions 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373530 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2bb5a6396ca7bf80822a4f68c613805b26862594..d9270851b15a76953d95e2de0cb1a65f6f753ed8 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12573,8 +12573,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
                } else {
                        /* Even if we are using RPID, we shouldn't leak information in the From if the user wants
                         * their callerid restricted */
-                       l = CALLERID_UNKNOWN;
-                       n = l;
+                       l = "anonymous";
+                       n = CALLERID_UNKNOWN;
                        d = FROMDOMAIN_INVALID;
                }
        }

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 53216a21bb4efed8c015f226eabdce0af9c56dff..5bafae292a2bb4adca0e89ae88215d556aeffcd4 100644 (file)
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -1589,9 +1589,7 @@ static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned cha
                new_duration = (new_duration & ~0xFFFF) | samples;
 
                if (event_end & 0x80) {
-                       /* End event.  Absorb re-transmits, and account for some endpoints
-                        * that erroneously increment the timestamp during re-transmissions */
-                       if ((seqno != rtp->last_seqno) && (timestamp > rtp->last_end_timestamp + 320)) {
+                       if ((seqno != rtp->last_seqno) && (timestamp > rtp->last_end_timestamp)) {
                                rtp->last_end_timestamp = timestamp;
                                rtp->dtmf_duration = new_duration;
                                rtp->resp = resp;