Commit Verlauf

Autor SHA1 Nachricht Datum
  Walter Doekes 1f4600a641 chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. vor 10 Jahren
  Joshua Colp 532fa8a7b1 Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11 vor 10 Jahren
  Matthew Jordan f838447c09 res_http_websocket: Close websocket correctly and use careful fwrite vor 10 Jahren
  Jonathan Rose 37ef39faaf chan_sip: Add sendrpid trust options vor 11 Jahren
  Jonathan Rose 2aeb1dba96 Reverting r411189 so that it can be put up for public review vor 11 Jahren
  Jonathan Rose ef12ef6942 chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) vor 11 Jahren
  Richard Mudgett f756a5fcca tcptls.c: Made TLS handle a certificate chain file. vor 11 Jahren
  Rusty Newton d38c638ea8 Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 vor 11 Jahren
  Richard Mudgett 004df1fd66 Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. vor 11 Jahren
  Kinsey Moore 04b56c2cca Allow Asterisk to retry after 403 on register vor 11 Jahren
  Walter Doekes d4819bdf19 Add "autoframing" option to sip.conf.sample and h323.conf.sample. vor 11 Jahren
  Kinsey Moore 60b9f4de51 Refactor extraneous channel events vor 11 Jahren
  Richard Mudgett ab80128155 Reimplement bridging and DTMF features related channel variables in the bridging core. vor 11 Jahren
  Matthew Jordan ff7ceb117f Add RFC 3327 Path header support to chan_sip vor 12 Jahren
  Walter Doekes ac9dfd1f89 Remove "registertrying" and add "rtp_engine" from/to sip.conf.sample vor 12 Jahren
  Brent Eagles 0ca22434d7 This change adds a SIP peer configuration feature to allow the peer's vor 12 Jahren
  Joshua Colp 7725c26943 Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default. vor 12 Jahren
  Jonathan Rose c231d00b13 chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry vor 12 Jahren
  Terry Wilson ac5d6cc794 Properly handle UAC/UAS roles for SIP session timers vor 12 Jahren
  Joshua Colp 194643102b Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip. vor 12 Jahren
  Matthew Jordan 383d44bc9f Add named callgroups/pickupgroups vor 12 Jahren
  Mark Michelson eb6f62c59e Add headers from SIPAddHeader to outbound REFER requests. vor 12 Jahren
  Mark Michelson 595ba54f10 Add separate configuration options for subscription and registration minexpiry and maxexpiry. vor 12 Jahren
  Joshua Colp 54f70f7b01 Prevent multiple local candidates from being added with the same information and add support for disabling ICE on a per-peer basis. vor 12 Jahren
  Joshua Colp df4e2ad0aa Add support for SIP over WebSocket. vor 12 Jahren
  Jonathan Rose 8787485cc9 Named ACLs: Introduces a system for creating and sharing ACLs vor 12 Jahren
  Mark Michelson 663e0f8d60 Help mitigate potential reinvite glare scenarios. vor 13 Jahren
  Kinsey Moore 41dc690b14 Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) vor 13 Jahren
  Joshua Colp e03ea59692 Add support for lightweight NAT keepalive. vor 13 Jahren
  Jonathan Rose a31decccf1 Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE vor 13 Jahren