tzafrir/asterisk

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

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