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  1. Sep 05, 2017
  2. Sep 01, 2017
  3. Aug 31, 2017
  4. Aug 30, 2017
    • George Joseph's avatar
      pjsip_message_ip_updater: Fix issue handling "tel" URIs · f78f5278
      George Joseph authored
      sanitize_tdata was assuming all URIs were SIP URIs so when a non
      SIP uri was in the From, To or Contact headers, the unconditional
      cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused
      a segfault when trying to access uri->other_param.
      
      * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri)
        checks before attempting to cast or use the returned uri.
      
      ASTERISK-27152
      Reported-by: Ross Beer
      
      Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f
      f78f5278
    • Corey Farrell's avatar
      AST-2017-006: Fix app_minivm application MinivmNotify command injection · 1bf3dfff
      Corey Farrell authored
      An admin can configure app_minivm with an externnotify program to be run
      when a voicemail is received.  The app_minivm application MinivmNotify
      uses ast_safe_system() for this purpose which is vulnerable to command
      injection since the Caller-ID name and number values given to externnotify
      can come from an external untrusted source.
      
      * Add ast_safe_execvp() function.  This gives modules the ability to run
      external commands with greater safety compared to ast_safe_system().
      Specifically when some parameters are filled by untrusted sources the new
      function does not allow malicious input to break argument encoding.  This
      may be of particular concern where CALLERID(name) or CALLERID(num) may be
      used as a parameter to a script run by ast_safe_system() which could
      potentially allow arbitrary command execution.
      
      * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
      instead of ast_safe_system() to avoid command injection.
      
      * Document code injection potential from untrusted data sources for other
      shell commands that are under user control.
      
      ASTERISK-27103
      
      Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
      1bf3dfff
    • Joshua Colp's avatar
      res_rtp_asterisk: Only learn a new source in learn state. · 7f2a60fb
      Joshua Colp authored
      This change moves the logic which learns a new source address
      for RTP so it only occurs in the learning state. The learning
      state is entered on initial allocation of RTP or if we are
      told that the remote address for the media has changed. While
      in the learning state if we continue to receive media from
      the original source we restart the learning process. It is
      only once we receive a sufficient number of RTP packets from
      the new source that we will switch to it. Once this is done
      the closed state is entered where all packets that do not
      originate from the expected source are dropped.
      
      The learning process has also been improved to take into
      account the time between received packets so a flood of them
      while in the learning state does not cause media to be switched.
      
      Finally RTCP now drops packets which are not for the learned
      SSRC if strict RTP is enabled.
      
      ASTERISK-27013
      
      Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c
      7f2a60fb
    • Joshua Colp's avatar
      res_rtp_asterisk: Allow remote SSRC to change on an RTP instance. · 5ba82ced
      Joshua Colp authored
      When SDP renegotiation occurs it is possible for an RTP
      instance to be reused for a new stream, resulting in the remote
      SSRC changing if it is part of a bundle group. This change
      allows this and updates its mapping in the current bundle
      group.
      
      ASTERISK-27231
      
      Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490
      5ba82ced
    • Jenkins2's avatar
      36c15131
    • Andre Nazario's avatar
      chan_pjsip: Add tag info in CHANNEL function · 71be8d5b
      Andre Nazario authored
      Create local_tag and remote_tag in CHANNEL info to get tag from From and
      To headers of a SIP dialog.
      
      ASTERISK-27220
      
      Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524
      71be8d5b
  5. Aug 29, 2017
  6. Aug 28, 2017
  7. Aug 25, 2017
  8. Aug 24, 2017
  9. Aug 23, 2017
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