1. 19 Dec, 2021 1 commit
  2. 08 Dec, 2021 1 commit
  3. 06 Dec, 2021 1 commit
  4. 05 Dec, 2021 1 commit
  5. 10 Jun, 2020 3 commits
  6. 09 Jun, 2020 3 commits
  7. 08 Jun, 2020 1 commit
  8. 05 Jun, 2020 4 commits
    • Joshua C. Colp's avatar
      res_fax: Don't consume frames given to fax gateway on write. · d2500c62
      Joshua C. Colp authored
      In a particular fax gateway scenario whereby it would
      have to translate using the read translation path on a
      channel the frame being translated would be consumed.
      When the frame is in the write path it is not permitted
      to free the frame as the caller expects it to continue
      to exist.
      
      This change makes it so that the frame is only consumed
      on the read path where it is acceptable to free it.
      
      ASTERISK-28900
      
      Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d
      d2500c62
    • Alexander Traud's avatar
      pjproject_bundled: Honor --without-pjproject. · 0a4dffe6
      Alexander Traud authored
      The previous change missed that 'make' uses 'PJPROJECT_BUNDLED' anyway.
      
      ASTERISK-28929
      
      Change-Id: I7ef0e78a06ea391b59d95b99d46bbed3fec4fed9
      0a4dffe6
    • Pirmin Walthert's avatar
      res_pjsip_logger: use the correct pointer when logging tx_messages to pcap · e8c6e9ae
      Pirmin Walthert authored
      When writing tx messages to pcap files, Asterisk is using the wrong
      pointer resulting in lots of wasted space. This patch fixes it to use
      the correct pointer.
      
      ASTERISK-28932 #close
      
      Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23
      e8c6e9ae
    • sungtae kim's avatar
      bridge.c: Fixed null pointer exception · 25ae412f
      sungtae kim authored
      If the bridge show all command could not get the bridge snapshot, it causes null pointer exception.
      Fixed it to check the snapshot is null.
      
      ASTERISK-28920
      
      Change-Id: I3521fc1b832bfc69644d0833f2c78177e1e51f58
      25ae412f
  9. 02 Jun, 2020 1 commit
    • George Joseph's avatar
      Scope Tracing: A new facility for tracing scope enter/exit · ca3c22c5
      George Joseph authored
      What's wrong with ast_debug?
      
        ast_debug is fine for general purpose debug output but it's not
        really geared for scope tracing since it doesn't present its
        output in a way that makes capturing and analyzing flow through
        Asterisk easy.
      
      How is scope tracing better?
      
        Scope tracing uses the same "cleanup" attribute that RAII_VAR
        uses to print messages to a separate "trace" log level.  Even
        better, the messages are indented and unindented based on a
        thread-local call depth counter.  When output to a separate log
        file, the output is uncluttered and easy to follow.
      
        Here's an example of the output. The leading timestamps and
        thread ids are removed and the output cut off at 68 columns for
        commit message restrictions but you get the idea.
      
      --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001
      	--> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
      		--> res_pjsip_session.c:3669 handle_incoming_response PJSIP/
      			--> chan_pjsip.c:3265 chan_pjsip_incoming_response_after
      				--> chan_pjsip.c:3194 chan_pjsip_incoming_response P
      					    chan_pjsip.c:3245 chan_pjsip_incoming_respon
      				<-- chan_pjsip.c:3194 chan_pjsip_incoming_response P
      			<-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after
      		<-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/
      	<-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173
      <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001
      
        The messages with the "-->" or "<--" were produced by including
        the following at the top of each function:
      
        SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session));
      
        Scope isn't limited to functions any more than RAII_VAR is.  You
        can also see entry and exit from "if", "for", "while", etc blocks.
      
        There is also an ast_trace() macro that doesn't track entry or
        exit but simply outputs a message to the trace log using the
        current indent level.  The deepest message in the sample
        (chan_pjsip.c:3245) was used to indicate which "case" in a
        "select" was executed.
      
      How do you use it?
      
        More documentation is available in logger.h but here's an overview:
      
        * Configure with --enable-dev-mode.  Like debug, scope tracing
          is #ifdef'd out if devmode isn't enabled.
      
        * Add a SCOPE_TRACE() call to the top of your function.
      
        * Set a logger channel in logger.conf to output the "trace" level.
      
        * Use the CLI (or cli.conf) to set a trace level similar to setting
          debug level... CLI> core set trace 2 res_pjsip.so
      
      Summary Of Changes:
      
        * Added LOG_TRACE logger level.  Actually it occupies the slot
          formerly occupied by the now defunct "event" level.
      
        * Added core asterisk option "trace" similar to debug.  Includes
      	ability to specify global trace level in asterisk.conf and CLI
      	commands to turn on/off and set levels.  Levels can be set
      	globally (probably not a good idea), or by module/source file.
      
        * Updated sample asterisk.conf and logger.conf.  Tracing is
          disabled by default in both.
      
        * Added __ast_trace() to logger.c which keeps track of the indent
          level using TLS. It's #ifdef'd out if devmode isn't enabled.
      
        * Added ast_trace() and SCOPE_TRACE() macros to logger.h.
          These are all #ifdef'd out if devmode isn't enabled.
      
      Why not use gcc's -finstrument-functions capability?
      
        gcc's facility doesn't allow access to local data and doesn't
        operate on non-function scopes.
      
      Known Issues:
      
        The only know issue is that we currently don't know the line
        number where the scope exited.  It's reported as the same place
        the scope was entered.  There's probably a way to get around it
        but it might involve looking at the stack and doing an 'addr2line'
        to get the line number.  Kind of like ast_backtrace() does.
        Not sure if it's worth it.
      
      Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027
      ca3c22c5
  10. 01 Jun, 2020 1 commit
    • Pirmin Walthert's avatar
      res_pjsip_logger.c: correct the return value checks when writing to pcap · c16937cd
      Pirmin Walthert authored
      files
      
      fwrite() does return the number of elements written and not the
      number of bytes. However asterisk is currently comparing the return
      value to the size of the written element what means that asterisk logs
      five WARNING messages on every packet written to the pcap file.
      
      This patch changes the code to check for the correct value, which will
      always be 1.
      
      ASTERISK-28921 #close
      
      Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2
      c16937cd
  11. 27 May, 2020 1 commit
    • Joshua C. Colp's avatar
      res_pjsip: Use correct pool for storing the contact_user value. · 9c2871ed
      Joshua C. Colp authored
      When replacing the user portion of the Contact URI the code
      was using the ephemeral pool instead of the tdata pool. This
      could cause the Contact user value to become invalid after a
      period of time.
      
      The code will now use the tdata pool which persists for the
      lifetime of the message instead.
      
      ASTERISK-28794
      
      Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5
      9c2871ed
  12. 22 May, 2020 1 commit
  13. 21 May, 2020 1 commit
    • Joshua C. Colp's avatar
      bridge: Don't try to match audio formats. · afa2c9a8
      Joshua C. Colp authored
      When bridging channels we were trying to match the audio
      formats of both sides in combination with the configured
      formats. While this is allowed in SDP in practice this
      causes extra reinvites and problems. This change ensures
      that audio streams use the formats of the first existing
      active audio stream. It is only when other stream types
      (like video) exist that this will result in re-negotiation
      occurring for those streams only.
      
      ASTERISK-28871
      
      Change-Id: I22f5a3e7db29e00c165e74d05d10856f6086fe47
      afa2c9a8
  14. 20 May, 2020 6 commits
    • Joshua C. Colp's avatar
      res_sorcery_config: Always reload configuration on errors. · ec7890d7
      Joshua C. Colp authored
      When a configuration file in Asterisk is loaded
      information about it is stored such that on a
      reload it is not reloaded if nothing has changed.
      This can be problematic when an error exists in
      a configuration file in PJSIP since the error
      will be output at start and not subsequently on
      reload if the file is unchanged.
      
      This change makes it so that if an error is
      encountered when res_sorcery_config is loading
      a configuration file a reload will always read
      in the configuration file, allowing the error
      to be seen easier.
      
      Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed
      ec7890d7
    • Alexander Traud's avatar
      res_srtp: Set all possible flags while selecting the Crypto Suite. · 4de0e50c
      Alexander Traud authored
      The flags of a previous selection could have been set within the
      object 'srtp', for example, when the previous selection returned
      failure after setting just 'some' flags. Now, not to clutter the
      code, all possible flags are cleared first, and then the selected
      flags are set as before.
      
      ASTERISK-28903
      
      Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee
      4de0e50c
    • Joshua C. Colp's avatar
      bridge_softmix: Always remove audio from mixed frame. · e8c8d69d
      Joshua C. Colp authored
      When receiving audio from a channel we determine if it
      is talking or silence based on a threshold value. If
      this threshold is met we always mix the audio into the
      conference bridge. If this threshold is not met we also
      mix the audio into the conference bridge UNLESS the
      drop silence option is enabled.
      
      The code that removed the audio from the mixed frame
      assumed that it was always not present if it did not
      meet the threshold to be considered talking. This is
      incorrect. If it has been stated that the audio was
      mixed into the mixed frame then it has been mixed into
      the mixed frame. By not removing audio that was
      considered non-talking it was possible for a channel
      to receive a slight echo of audio of itself at times.
      
      This change ensures that the audio is always removed
      from the mixed frame going back to the channel so it
      no longer receives the slight echo.
      
      ASTERISK-28898
      
      Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb
      e8c8d69d
    • Ben Ford's avatar
      res_stir_shaken: Add unit tests for signing and verification. · f506cc48
      Ben Ford authored
      Added two unit tests, one for signing and another for verifying.
      stir_shaken_sign checks to make sure that all the required parameters
      are passed in and then signs the actual payload. If a signature is
      produced and a payload returned as a result, the test passes.
      stir_shaken_verify takes the signature from a signed payload to verify.
      This unit test also verifies that all the required information is passed
      in, and then attempts to verify the signature. If verification is
      successful and a payload is returned, the test passes.
      
      Change-Id: I9fa43380f861ccf710cd0f6b6c102a517c86ea13
      f506cc48
    • Joshua C. Colp's avatar
      res_pjsip_logger: Expand functionality to improve logging. · a7aaee70
      Joshua C. Colp authored
      The PJSIP packet logger now has the following CLI commands:
      
      pjsip set logger pcap <filename>
      
      When used this will create a pcap file containing the incoming
      and outgoing SIP packets, in unencrypted form.
      
      pjsip set logger verbose <on / off>
      
      This allows you to toggle logging to verbose on and off.
      
      pjsip set logger host <IP/subnet mask> add
      
      This allows you to add an additional IP address or subnet
      mask to logging, allowing you to log multiple instead of
      just a single IP address or all traffic.
      
      The normal "pjsip set logger host" CLI command has also been
      expanded to allow subnet masks as well.
      
      ASTERISK-28895
      
      Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e
      a7aaee70
    • Nicholas John Koch's avatar
      res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings · fef97a9a
      Nicholas John Koch authored
      A warning was triggered that there may be a problem regarding file
      extension (which is correct and should not be set anyway). The warning
      also appeared if there was dot within the path itself.
      
      E.g.
      [sales-queue-hold]
      mode=playlist
      entry=/var/www/domain.tld/moh/funky_music
      
      The music played correctly but you get a warning message.
      
      Now there will be a check if the position of a potential dot character
      is after the last position of a slash character. This dot charachter
      will be treated as a extension naming. Dots within the path then ignored.
      
      ASTERISK-28892
      Reported-By: Nicholas John Koch
      
      Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e
      fef97a9a
  15. 18 May, 2020 1 commit
  16. 15 May, 2020 1 commit
    • Joshua C. Colp's avatar
      ari: Allow variables to be set on channel create. · 15cbff9d
      Joshua C. Colp authored
      This change adds the same variable functionality that
      is available for originating a channel to the create
      call. Now when creating a channel you can specify
      dialplan variables to set instead of having to do another
      API call.
      
      ASTERISK-28896
      
      Change-Id: If13997ba818136d7c070585504fc4164378aa992
      15cbff9d
  17. 13 May, 2020 2 commits
    • Roger James's avatar
      pjsip_resolver.c: Ensure AAAA dns requests are made. · c8dec423
      Roger James authored
      1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups
         when an IPV6 transport type has been requested.
      
      2. Rename all occurrences of pjsip_transport_get_type_name to
         pjsip_transport_get_type_desc. This ensures that the log/debug info
         shows whether the transport is IPv6 or IPv4.
      
      3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport
         types. This results in invalid values. Use a bitwise or instead.
      
      ASTERISK-26780
      Patches:
          pjsip_resolver.c uploaded by Peter Sokolov (License #7070)
      
      Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58
      c8dec423
    • Ben Ford's avatar
      res_stir_shaken: Added dialplan function and API call. · e29df34d
      Ben Ford authored
      Adds the "STIR_SHAKEN" dialplan function and an API call to add a
      STIR_SHAKEN verification result to a channel. This information will be
      held in a datastore on the channel that can later be queried through the
      "STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results
      including identity, attestation, and verify_result. Here are some
      examples:
      
      STIR_SHAKEN(count)
      STIR_SHAKEN(0, identity)
      STIR_SHAKEN(1, attestation)
      STIR_SHAKEN(2, verify_result)
      
      Getting the count can be used to iterate through the results and pull
      information by specifying the index and the field you want to retrieve.
      
      Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82
      e29df34d
  18. 11 May, 2020 4 commits
  19. 08 May, 2020 1 commit
    • Pirmin Walthert's avatar
      app.c: make sure that no non-async-signal-safe syscalls are used after · 6b2d9451
      Pirmin Walthert authored
      fork before exec
      
      Posix does only allow async-signal-safe syscalls after fork before exec.
      As asterisk ignores this, functions like TrySystem or System sometimes
      end up in a deadlocked child process. The patch prevents the use of
      non-async-signal-safe syscalls.
      
      ASTERISK-28776
      
      Change-Id: Idc76365c0592ee3f3b3bd72a4f48f7a098978e8e
      6b2d9451
  20. 06 May, 2020 2 commits
    • George Joseph's avatar
      streams: Fix one memory leak and one formats ref issue · 7fbfbe7d
      George Joseph authored
      ast_stream_topology_create_from_format_cap() was setting the
      stream->formats directly but not freeing the default formats.  This
      causes a memory leak.
      
      * ast_stream_topology_create_from_format_cap() now calls
        ast_stream_set_formats() which properly cleans up the existing
        stream formats.
      
      When cloning a stream, the source stream's format caps _pointer_ is
      copied to the new stream and it's reference count bumped.  If
      either stream is set to "removed", this will cause _both_ streams
      to have their format caps cleared.
      
      * ast_stream_clone() now creates a new format caps object and copies
        the formats from the source stream instead of just copying the
        pointer.
      
      ASTERISK-28870
      
      Change-Id: If697d81c3658eb7baeea6dab413b13423938fb53
      7fbfbe7d
    • Nathan Bruning's avatar
      app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions · f217fcdc
      Nathan Bruning authored
      Add a new "masquarade" channel event, and use it in app_queue to track unique id's.
      
      Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210
      
      ASTERISK-28829 #close
      ASTERISK-25844 #close
      
      Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6
      f217fcdc
  21. 05 May, 2020 2 commits
  22. 01 May, 2020 1 commit