Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity
8735 Commits
140 Branches
1567 Tags
344 MiB
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from '09778b268e'
${ noResults }
Commit Graph

1274 Commits (09778b268ea887ef53de4a8dc8f17c5fec615d5c)

Author SHA1 Message Date
Kevin P. Fleming 09778b268e move 'struct ast_rtp' back to rtp.c where it belongs
19 years ago
Olle Johansson 36506dad2d Debug logging fixes
19 years ago
Olle Johansson fee02a9b7d At least reply if we get a fatal error in handling a packet.
19 years ago
Olle Johansson 91732256db Don't crash if chan2 does not exist (one-legged call situations, like echo)
19 years ago
Olle Johansson b796bf414b Improve SIP history
19 years ago
Olle Johansson 89081c3d06 Issue #7294 - Asterisk sends INVITE instead of BYE. Fix by Philippe Sultan. Thanks!
19 years ago
Kevin P. Fleming 472c1ca282 simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
19 years ago
Olle Johansson 479181951a Merge of the "sdpcleanup" branch. Thanks to John Martin for a lot of tests
19 years ago
Olle Johansson 8bc0599b11 Issue 7208 - Handle ; in userinfo part of URI too
19 years ago
Olle Johansson 404f8db694 Do not use ':' as a separator between arguments (kept around for 1.4 for backwards compatibility)
19 years ago
Olle Johansson 62dd0dc6f2 Only override callingpres set by rpid if user/peer preference is set. (Issue #7240)
19 years ago
Olle Johansson 02bc3637da Code simplification
19 years ago
Olle Johansson d520d238f1 - Fix more problems in transfer code (thanks Tholo!)
19 years ago
Olle Johansson 043b765bdc Respond to some of Luigis questions...
19 years ago
Olle Johansson e97507e8fb Fix potential bug. THanks luigi!
19 years ago
Olle Johansson 0d6906e107 - Doxygen fix
19 years ago
Olle Johansson c2b50fd2bd Show max bitrate in "sip show user" and "sip show channel"
19 years ago
Olle Johansson 439e4963ab - Typo in chan_sip (: missing)
19 years ago
Luigi Rizzo 531ed6f290 add a bit of comment on what build_route does,
19 years ago
Luigi Rizzo 88077a4d1f remove some duplicated code;
19 years ago
Luigi Rizzo 2724885b78 small simplification in assignments, and fix a typo in a comment.
19 years ago
Luigi Rizzo 24d1d23a93 mark XXX some places where there is a return right before
19 years ago
Luigi Rizzo 8dbdd8518b fix indentation of a block
19 years ago
Luigi Rizzo d18e363523 some code simplifications using S_OR() to remove some if blocks
19 years ago
Luigi Rizzo ec46b5a4ce use ast_skip_blanks() where appropriate, and use standard
19 years ago
Luigi Rizzo b7b716b618 mark XXX a buggy section of code and implement a probable
19 years ago
Luigi Rizzo 7483e1d2b6 simplify logic in a small block of code
19 years ago
Luigi Rizzo c7a5a96fc6 mark XXX a bug in the code. oej/kevin, any idea what you want
19 years ago
Luigi Rizzo a63b2e4c53 remove duplicate code
19 years ago
Luigi Rizzo 1a38041830 Replace '\"' with '"'.
19 years ago
Olle Johansson e0a8649273 - Formatting
19 years ago
Olle Johansson b7cfe5bd08 Minor change to DEBUG and WARNING messages
19 years ago
Olle Johansson 4d17451c55 Issue #7103 take two. Thanks Mikael!
19 years ago
Russell Bryant 4c76028de9 - add the ability to configure forced jitterbuffers on h323, jingle,
19 years ago
Russell Bryant 62b2f320a1 make a global variable static and remove an unused global variable
19 years ago
Olle Johansson 725e285bc4 Commit of the new SIP transfer support (oej/siptransfer branch)
19 years ago
Russell Bryant 414106a3a6 move the includes of abstract_jb.h to be with the rest of the asterisk includes.
19 years ago
Russell Bryant bb7dd96cfe Add support for using a jitterbuffer for RTP on bridged calls. This includes
19 years ago
Joshua Colp 503486e99d OKAY - Only replace our IP address in the outgoing SIP messages if the actual source address is within the specified local network. (issue #7051 reported by Luke-Jr with mods by me)
19 years ago
Olle Johansson c7007e1f55 Add congestion message to SIP history
19 years ago
Olle Johansson bc0f0a3226 Use ast_channel_trylock for channels
19 years ago
Kevin P. Fleming f890cab292 Merged revisions 30293 via svnmerge from
19 years ago
Kevin P. Fleming 57f6093661 don't send CANCEL for an INVITE that we haven't received a provisional response for yet... mark it pending until a response arrives (issue #7079)
19 years ago
Olle Johansson b8b1fda04c Typo
19 years ago
Kevin P. Fleming 3e99be68d1 add a new option for 'obscuring' SIP user/peer names from fishers
19 years ago
Kevin P. Fleming b31af470e7 Merged revisions 29764 via svnmerge from
19 years ago
BJ Weschke c4227714fd Sanity check code for an extended failure in trying to obtain a channel lock that may have been obtained elsewhere. Prevents the monitor thread of the SIP module from going into an infinite loop, effectively, breaking SIP until you restart Asterisk or the mutex is unlocked, whichever comes first.
19 years ago
Olle Johansson 16a648aaf1 Breaking once will stop us... :-)
19 years ago
Olle Johansson 883e4d6a9b Code formatting
19 years ago
Mark Spencer fa9e0ed768 Handle ringing (early) state properly on SIP
19 years ago