Commit Graph

7550 Commits (2bb01eb0ff9a87bfc95899de19a777746c0e94b7)

Author SHA1 Message Date
Joshua Colp fb74294b92 Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
13 years ago
Richard Mudgett c7e2bf3187 chan_dahdi/SS7: Made reject incoming call for an in-alarm or blocked channel.
13 years ago
Mark Michelson a65fbf8012 Multiple revisions 375993-375994
13 years ago
Joshua Colp 5f28931a1f Fix a bug where our Motif ICE candidates were not quite proper, and make us more forgiving.
13 years ago
Matthew Jordan f0cd27e027 Refactor ast_timer_ack to return an error and handle the error in timer users
13 years ago
Richard Mudgett 93d85a0087 Things don't need to be that const.
13 years ago
Damien Wedhorn f4fb271601 Fix for chan_skinny leaving RTP ports open
13 years ago
Richard Mudgett 35e96f995e Multiple revisions 375519-375524
13 years ago
Michael L. Young 2fce31c09a Fix Wrong Result In Debug Message For SDP Origin Processing
13 years ago
Jonathan Rose 509f348639 chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry
13 years ago
Mark Michelson d51cc27812 Prevent resetting of NATted realtime peer address on reload.
13 years ago
Richard Mudgett 421fbee8d8 chan_dahdi: Fix segfault dereferencing a NULL tech_pvt.
13 years ago
Walter Doekes 0ee22cfd14 Fixes to the fd-oriented SIP TCP reads.
13 years ago
Walter Doekes 5fc8671fb7 Update sip_request_call SIP dial string documentation.
13 years ago
Joshua Colp a318db28e3 Remove a log message that was left in accidentally from call-id logging development.
13 years ago
Mark Michelson 94c0fa9098 Fix some potential misuses of ast_str in the code.
13 years ago
Igor Goncharovskiy 5b1a89e1b1 Fix underscreen buttons warnings apeared while transfer process
13 years ago
Mark Michelson ccf01fbfdc Do not use a FILE handle when doing SIP TCP reads.
13 years ago
Joshua Colp 963f94e99f Fix a bug where audio on Google Voice would not work due to ignoring candidates.
13 years ago
Joshua Colp 385b30fbc6 Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures.
13 years ago
Mark Michelson b5f231501b Don't make chan_sip export global symbols.
13 years ago
Joshua Colp d5dc7d8b03 Consider the Google Talk content stanza name (jin:content) valid.
13 years ago
Joshua Colp 332407b5f8 Improve logging for DTLS-SRTP failure situations.
13 years ago
Joshua Colp 749bd15c6f Add a log message for when DTLS-SRTP is requested and the underlying engine does not support it.
13 years ago
Richard Mudgett f76557db58 Merged revisions 374515-374535 from
13 years ago
Matthew Jordan 8943656ccc Fix a variety of ref counting issues
13 years ago
Matthew Jordan 30d590a970 Fix ref leak when adding ICE candidates to an SDP
13 years ago
Joshua Colp f8e894e031 Make res_http_websocket an optional dependency on supported platforms for chan_sip.
13 years ago
Joshua Colp 5e0aff508c Fix an issue where Local channels dialed by app_queue are considered in use immediately.
13 years ago
Mark Michelson 70cb09cd56 Move handling of 408 response so there is no misleading warning message.
13 years ago
Mark Michelson d9e1cec84a Remove dead code and documentation for nonexistent feature.
13 years ago
Joshua Colp 59c9a7205a Fix T.38 support when used with chan_local in between.
13 years ago
Terry Wilson ba4e0c1591 Properly handle UAC/UAS roles for SIP session timers
13 years ago
Jonathan Rose 57771ffe11 chan_sip: Set Quality of Service for video rtp instance
13 years ago
Richard Mudgett fcd5d7f458 Be consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid>
13 years ago
Richard Mudgett 26e45bbfca Fix potential reentrancy problems in chan_sip.
13 years ago
Joshua Colp f3e09ab823 Fix a deadlock caused by a race condition between removing a hint and reloading the dialplan and subscribing to the removed hint.
13 years ago
Joshua Colp b40fecd9ab Fix an issue with H.264 format attribute comparison and fix an issue with improper SDP being produced.
13 years ago
Jonathan Rose 388509cfa9 iax2-provision: Fix improper return on failed cache retrieval
13 years ago
Joshua Colp 42ebea2f2f Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
13 years ago
Kinsey Moore 19fcfcb280 Correct handling of unknown SDP stream types
13 years ago
Richard Mudgett 7687370500 Made companding law for SS7 calls only determined by SS7 signaling type.
13 years ago
Matthew Jordan 9e396da730 Resolve memory leaks in TLS initialization and TLS client connections
13 years ago
Joshua Colp 0b9f1c4e0d Skip any non-content information when looking for and handling content.
13 years ago
Mark Michelson cc8afceba5 Add channel name to a warning to make debugging easier.
13 years ago
Jonathan Rose 79d0efd393 chan_local: Switch from using a random 4 digit hex identifier to unique id
13 years ago
Kinsey Moore b7aa658cf9 Ensure iax2 debug output is displayed when expected
13 years ago
Kinsey Moore 05cccdea8c Deprecate chan_gtalk, chan_jingle, and res_jabber
13 years ago
Matthew Jordan 0067aba7e8 Only re-create an SRTP session when needed
13 years ago
Richard Mudgett 1af1164d43 Fix loss of MOH on an ISDN channel when parking a call for the second time.
13 years ago