Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity
823 Commits
150 Branches
1605 Tags
304 MiB
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from '1cdcc857d9'
${ noResults }
Commit Graph

276 Commits (1cdcc857d982b7360cdeb87d0f3f0a8fd98ff55c)

Author SHA1 Message Date
Mark Spencer 2cf87c7c50 Fix ALSA/OSS leak
23 years ago
Jeremy McNamara ac16b57304 tweak internal codec order
23 years ago
Jeremy McNamara 979de9323a make non matching IP addresses go to invalid extension if default context exists
23 years ago
Mark Spencer a02b54bfb8 Add MGCP audit
23 years ago
Mark Spencer 7cabdde1ce Fix zombie *8# channels in SIP
23 years ago
Mark Spencer c3098625ea Fix blind transfer with macro
23 years ago
Mark Spencer bbcac3aa85 Fix deferred dialing on FXO
23 years ago
Mark Spencer 256093b5da Suppress 487 message since people think it is an error
23 years ago
Mark Spencer 13dbf301c6 update CREDTS file
23 years ago
Mark Spencer 5654b1b886 Use hostname instead of IP for sip registration
23 years ago
Mark Spencer f94a62d64e Don't double ack subscribes
23 years ago
Mark Spencer aa39431646 Put number in name field if name field is NULL *or* has 0 length
23 years ago
Mark Spencer a6cc0b233e When swapping subs, queue a null frame, extend delay for hangup-on-flash
23 years ago
Jeremy McNamara 4ffdfcbec7 remove debug
23 years ago
Jeremy McNamara c75fdd13cf more debug
23 years ago
Mark Spencer 93cd3c18e7 Only do call progress on outgoing calls
23 years ago
Jeremy McNamara ae23339452 add some debug for testing
23 years ago
Mark Spencer 10a7938b6e Only consider a RINGOFFHOOK as "UP" when a call is outgoing
23 years ago
Mark Spencer 15b1b0b0f1 Fix double transfer bug
23 years ago
Mark Spencer 39094852db Merge and edit Nick's dial support
23 years ago
Mark Spencer b430999a96 Ignore ;user=phone on register
23 years ago
Mark Spencer f8a6b5e31d Parse initial request
23 years ago
Mark Spencer 1bfcc8d8d0 Don't ACK on anything but INVITE
23 years ago
Mark Spencer ec6cca344b Add missing "Contact" line to Register
23 years ago
Mark Spencer c7c91a93f2 Allow "fromuser" option and don't append port number if it's specified (you can do it with fromdomain=temp.com:5060
23 years ago
Mark Spencer 14009dbfa1 Fix SIP registration (I hope)
23 years ago
Mark Spencer 9d5e3ee8b9 Clear out old call
23 years ago
Mark Spencer af6aeefada First pass at fixing REGISTER functionality under SIP
23 years ago
Mark Spencer 3bcda42140 Fix potential hangup issues
23 years ago
Jeremy McNamara ea57a86564 fix silly typo
23 years ago
Mark Spencer fa9c225dd3 Don't send 180 "Ringing" after 183 Session Progres
23 years ago
Mark Spencer 6301ef6baf Minor chan_nbs fix
23 years ago
Mark Spencer 863104131d Check authentication before destination
23 years ago
Mark Spencer 43604eb770 Implement call pickup on SIP, override context if appropriate
23 years ago
Mark Spencer 2758349cae Make all address comparisons explicit
23 years ago
Mark Spencer 6978d58552 Send empty voice frame on 183 session progress
23 years ago
Mark Spencer 4bd768a43e Always ack packets
23 years ago
Jeremy McNamara b49fd4ab95 d'oh! make it so you can acutally use other ports than 1720
23 years ago
Mark Spencer 501d2cdc17 Update to 3261 branch definition
23 years ago
Mark Spencer 16c2861fa8 Fix contact stuff once-and-for-all (I hope)
23 years ago
Mark Spencer b6a6c83b78 Fix peerpoke to also acknowledge
23 years ago
Mark Spencer 9cc2aa39d0 Unlink local channels
23 years ago
Mark Spencer 2f771c677e After 404 or 484, don't send 503
23 years ago
Mark Spencer 3e8183de43 Add expiry patch from Jim
23 years ago
Mark Spencer d6b38a8b3b Fix retransmission on BYE
23 years ago
Mark Spencer fd6cc08ac8 Phone updates
23 years ago
Mark Spencer 736bb7c6dc Fix phone crash
23 years ago
Jeremy McNamara 451e3aa6dd fix gatekeeper logic
23 years ago
Mark Spencer 5c18d407c5 Fix contact header, don't transmit BYE when INVITE are still outstanding
23 years ago
Mark Spencer 1680ccf66c Provide authentication for BYE and CANCEL requests
23 years ago