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  1. May 09, 2016
    • George Joseph's avatar
      pjproject_bundled: Check for python-dev and TEST_FRAMEWORK · facce6f6
      George Joseph authored
      The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
      The python bindings are now built only if TEST_FRAMEWORK is set and a
      python development package is installed.
      
      libresample was also disabled.
      
      ASTERISK-25993 #close
      Reported-by: Joshua Colp
      
      Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03
      facce6f6
  2. May 06, 2016
  3. May 05, 2016
  4. May 04, 2016
  5. May 03, 2016
    • Jean Aunis's avatar
      app_chanspy: fix audiohook options in non read-only mode · 0c9faaee
      Jean Aunis authored
      When option 'o' was not set, ChanSpy created its audiohook with the flag
      AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one
      direction only.
      
      ASTERISK-25866 #close
      
      Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0
      0c9faaee
    • Alexei Gradinari's avatar
      res_pjsip/AMI: add contact.updated event · a4cfcda0
      Alexei Gradinari authored
      With the old SIP module AMI sends PeerStatus event on every
      successfully REGISTER requests, ie, on start registration,
      update registration and stop registration.
      
      With PJSIP AMI sends ContactStatus only when status is changed.
      Regarding registration:
      on start registration - Created
      on stop registration - Removed
      but on update registration nothing
      
      This patch added contact.updated event.
      
      ASTERISK-25904
      
      Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f
      a4cfcda0
    • zuul's avatar
      Merge "pjsip: Added "reg_server" to contacts." · c339d4c6
      zuul authored
      c339d4c6
    • Joshua Colp's avatar
    • George Joseph's avatar
      pjproject_bundled: Various fixes discovered during testing of OSes · e61716b7
      George Joseph authored
      For all OSes:
      * Disabled third-party codecs in pjproject and added
        '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
        configure options since we don't use the pjsip codec capability.
      
      FreeBSD:
      * Added FreeBSD support to install_prereq.
      * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
      * Added __progname and environ to asterisk.exports.in.
      * Reverted the use of ldconfig to create shared library symlinks to ln.
      * Only enable epoll in pjproject if `uname -s` is Linux.
      * Added a patch to pjproject to take the name of the 'make' command from
        an environment variable if supplied.  This is needed for the python bindings.
        (merged by Teluu into pjproject trunk 5/3/2016)
      FreeBSD support isn't complete.  Still some general issues regarding
      make/gmake having nothing to do with pjproject.  With some handholding it DOES
      build successfully.
      
      CentOS:
      Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
      CentOS 6/7 32/64 build and run the pjsip testsuite successfully.
      
      Ubuntu:
      No changes required.
      Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.
      
      Debian:
      No changes required.
      Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.
      
      There will utimately be a follow-up patch to create an install_prereq for
      the testsuite as I've discovered a few missing requirements.
      
      ASTERISK-25968 #close
      
      Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c
      e61716b7
    • zuul's avatar
      9c4c4a7d
    • Andrew Nagy's avatar
      app_voicemail: always copy dynamic struct to avoid race condition · 080c6216
      Andrew Nagy authored
      Voicemail email addresses can be corrupt or voicemail
      emails can end up being sent to the wrong email address if asterisk is
      reading voicemail.conf during a reload and processing an email at the
      same time. This patch always copies the struct that would otherwise only
      be copied once.
      
      ASTERISK-24463 #close
      Reported by: John Campbell
      Tested by: Etienne Lessard
      Tested by: Andrew Nagy
      Change-Id: I3a0643813116da84e2617291903d0d489b7425fb
      080c6216
  6. May 02, 2016
    • Alexei Gradinari's avatar
      pjsip: Added "reg_server" to contacts. · 2b1edee7
      Alexei Gradinari authored
      If the Asterisk system name is set in asterisk.conf, it will be stored
      into the "reg_server" field in the ps_contacts table to facilitate
      multi-server setups.
      
      ASTERISK-25931
      
      Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8
      2b1edee7
  7. May 01, 2016
  8. Apr 29, 2016
  9. Apr 28, 2016
  10. Apr 27, 2016
    • George Joseph's avatar
      res_pjsip: Add ability to identify by Authorization username · 4ebf9a93
      George Joseph authored
      A feature of chan_sip that service providers relied upon was the ability to
      identify by the Authorization username.  This is most often used when customers
      have a PBX that needs to register rather than identify by IP address.  From my
      own experiance, this is pretty common with small businesses who otherwise
      don't need a static IP.
      
      In this scenario, a register from the customer's PBX may succeed because From
      will usually contain the PBXs account id but an INVITE will contain the caller
      id.  With nothing recognizable in From, the service provider's Asterisk can
      never match to an endpoint and the INVITE just stays unauthorized.
      
      The fixes:
      
      A new value "auth_username" has been added to endpoint/identify_by that
      will use the username and digest fields in the Authorization header
      instead of username and domain in the the From header to match an endpoint,
      or the To header to match an aor.  This code as added to
      res_pjsip_endpoint_identifier_user rather than creating a new module.
      
      Although identify_by was always a comma-separated list, there was only
      1 choice so order wasn't preserved.  So to keep the order, a vector was added
      to the end of ast_sip_endpoint.  This is only used by res_pjsip_registrar
      to find the aor.  The res_pjsip_endpoint_identifier_* modules are called in
      globals/endpoint_identifier_order.
      
      Along the way, the logic in res_pjsip_registrar was corrected to match
      most-specific to least-specific as res_pjsip_endpoint_identifier_user does.
      
      The order is:
      
      username@domain
      username@domain_alias
      username
      
      Auth by username does present 1 problem however, the first INVITE won't have
      an Authorization header so the distributor, not finding a match on anything,
      sends a securty_alert.  It still sends a 401 with a challenge so the next
      INVITE will have the Authorization header and presumably succeed.  As a result
      though, that first security alert is actually a false alarm.
      
      To address this, a new feature has been added to pjsip_distributor that keeps
      track of unidentified requests and only sends the security alert if a
      configurable number of unidentified requests come from the same IP in a
      configurable amout of time.  Those configuration options have been added to
      the global config object.  This feature is only used when auth_username
      is enabled.
      
      Finally, default_realm was added to the globals object to replace the hard
      coded "asterisk" used when an endpoint is not yet identified.
      
      The testsuite tests all pass but new tests are forthcoming for this new
      feature.
      
      ASTERISK-25835 #close
      Reported-by: Ross Beer
      
      Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
      4ebf9a93
    • Joshua Colp's avatar
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