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  1. Mar 15, 2017
  2. Mar 14, 2017
    • zuul's avatar
    • Joshua Colp's avatar
    • Matt Jordan's avatar
      configs/samples/hep.conf.sample: Clarify how the HEP stack works · 05713c36
      Matt Jordan authored
      This patch updates the documenation in hep.conf.sample to better specify
      how the various HEP modules interact.
      
      ASTERISK-26717 #close
      
      Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124
      05713c36
    • Matt Jordan's avatar
      main/stasis_cache: Demote the ERROR message when removing a nonexistent item · b03b7271
      Matt Jordan authored
      This patch demotes the ERROR message that is displayed when a
      nonexistent item is removed from the Stasis cache. The genesis of this
      demotion is due to chan_sip's realtime peers and their interaction with
      Asterisk's core ast_endpoint code, but ostensibly it could happen from
      other channel drivers as well.
      
      Since Mark Michelson already did an excellent job of explaining on this
      issue, it is quoted here for posterity:
      
      "Internally, when a realtime peer is retrieved, Asterisk creates an
      ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
      destroyed as well. Part of the destruction of the ast_endpoint involves
      clearing the Stasis cache of all information about that endpoint. The
      problem here is that the act of creating the ast_endpoint is not enough
      to actually put any information in the Stasis cache. Instead, something
      has to happen, such as a state change, in order for the Stasis cache to
      have any information about that endpoint. When a device registers,
      chan_sip creates an ast_endpoint structure, processes the REGISTER, and
      then destroys the ast_endpoint. When the ast_endpoint is destroyed,
      there is nothing to destroy in the Stasis cache, so an error message is
      emitted. When you use rtcachefriends, ast_endpoint structures persist
      for the lifetime of the module and so you do not see this error
      message."
      
      ASTERISK-25237 #close
      
      Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70
      b03b7271
    • Matt Jordan's avatar
      res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue · 2d7e68c0
      Matt Jordan authored
      Tabs > spaces. Always.
      
      Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1
      2d7e68c0
  3. Mar 13, 2017
    • Joshua Colp's avatar
      chan_pjsip: Don't assume a session will have a channel. · 12460b05
      Joshua Colp authored
      When querying for PJSIP specific information using the dialplan
      function CHANNEL() it is possible that the underlying session
      will no longer have a channel associated with it. This is
      most likely to occur when the RTCP HEP module attempts to get
      the channel name. If this happens then a crash will occur.
      
      This change just adds a check that the channel exists on the
      session before querying it.
      
      ASTERISK-26857
      
      Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01
      12460b05
  4. Mar 11, 2017
  5. Mar 10, 2017
  6. Mar 09, 2017
  7. Mar 08, 2017
  8. Mar 07, 2017
    • zuul's avatar
      9cd1a754
    • Mark Michelson's avatar
      res_http_websocket: Fix faulty read logic. · 5d0371d7
      Mark Michelson authored
      When doing some WebRTC testing, I found that the websocket would
      disconnect whenever I attempted to place a call into Asterisk. After
      looking into it, I pinpointed the problem to be due to the iostreams
      change being merged in.
      
      Under certain circumstances, a call to ast_iostream_read() can return a
      negative value. However, in this circumstance, the websocket code was
      treating this negative return as if it were a partial read from the
      websocket. The expected length would get adjusted by this negative
      value, resulting in the expected length being too large.
      
      This patch simply adds an if check to be sure that we are only updating
      the expected length of a read when the return from a read is positive.
      
      ASTERISK-26842 #close
      Reported by Mark Michelson
      
      Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab
      5d0371d7
    • Jean Aunis's avatar
      chan_sip: Call not cancelled after receiving a 422 response · d51ca4b4
      Jean Aunis authored
      When receiving a 422 response, the invitestate variable must be reset to
      INV_CALLING.
      
      ASTERISK-26841
      
      Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099
      d51ca4b4
  9. Mar 06, 2017
    • Daniel Journo's avatar
      Saynumber is trying to get "and" from "digits/" subfolder · 272259a2
      Daniel Journo authored
      * say.c Changed 'digits/and' to 'vm-and' for en_GB
      
      ASTERISK-26598 #close
      
      Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe
      272259a2
    • Sean Bright's avatar
      pbx_spool: Gracefully handle long lines in call files · 5a74abc5
      Sean Bright authored
      Per the linked issue, we aren't checking the buffer filled by fgets()
      to determine if it contains a newline, so we will fail to correctly
      parse the trailing portion of a long line.
      
      This patch increases the buffer size from 256 to 1024, and skips any
      line that exceeds that length, logging a warning in the process.
      
      ASTERISK-17067 #close
      Reported by: Dave Olszewski
      
      Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0
      5a74abc5
  10. Mar 03, 2017
    • Richard Mudgett's avatar
      core: Cleanup ast_get_hint() usage. · c9296b23
      Richard Mudgett authored
      * manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
      if a hint does not exist for the requested extension.  Ran into this when
      developing a testsuite test.  The AMI event ExtensionStatus came out with
      the hint header value containing garbage.  The AMI event PresenceStatus
      also had the same issue.
      
      * manager.c:action_extensionstate() no need to completely initialize the
      hint[].  Only initialize the first element.
      
      * pbx.c:ast_add_hint() Remove unnecessary assignment.
      
      * chan_sip.c: Eliminate an unneeded hint[] local variable.  We only care
      about the return value of ast_get_hint() there.
      
      Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b
      c9296b23
  11. Mar 01, 2017
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