Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity
791 Commits
140 Branches
1561 Tags
340 MiB
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from 'b0bb0ed079'
${ noResults }
Commit Graph

791 Commits (b0bb0ed079f81edb8308e83a6910c26e719daa1b)
 

Author SHA1 Message Date
Mark Spencer b0bb0ed079 Fix transition to operator extension
22 years ago
Mark Spencer f8a6b5e31d Parse initial request
22 years ago
Mark Spencer 1bfcc8d8d0 Don't ACK on anything but INVITE
22 years ago
Mark Spencer ec6cca344b Add missing "Contact" line to Register
22 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
22 years ago
Mark Spencer 14009dbfa1 Fix SIP registration (I hope)
22 years ago
Mark Spencer 9d5e3ee8b9 Clear out old call
22 years ago
Mark Spencer af6aeefada First pass at fixing REGISTER functionality under SIP
22 years ago
Mark Spencer 3bcda42140 Fix potential hangup issues
22 years ago
Mark Spencer e24848eeaa Fix ACL issue, update CREDITS
22 years ago
Jeremy McNamara ea57a86564 fix silly typo
22 years ago
Mark Spencer fa4e465fd4 Add contributed sounds from John Todd and Allison Smith
22 years ago
Mark Spencer fa9c225dd3 Don't send 180 "Ringing" after 183 Session Progres
22 years ago
Mark Spencer 6301ef6baf Minor chan_nbs fix
22 years ago
Mark Spencer 863104131d Check authentication before destination
22 years ago
Mark Spencer 43604eb770 Implement call pickup on SIP, override context if appropriate
22 years ago
Mark Spencer be1d78a3ea Make voicemail timeout configurable
22 years ago
Mark Spencer 2758349cae Make all address comparisons explicit
22 years ago
Mark Spencer 7f844be5e0 Don't send empty RTP frames
22 years ago
Mark Spencer 6978d58552 Send empty voice frame on 183 session progress
22 years ago
Mark Spencer 4bd768a43e Always ack packets
22 years ago
Mark Spencer a0afd5f8fb Update sounds.txt
22 years ago
Mark Spencer 259d29a58d Fix documentation on chanisavail
22 years ago
Jeremy McNamara b49fd4ab95 d'oh! make it so you can acutally use other ports than 1720
22 years ago
Mark Spencer 501d2cdc17 Update to 3261 branch definition
22 years ago
Mark Spencer 16c2861fa8 Fix contact stuff once-and-for-all (I hope)
22 years ago
Mark Spencer b6a6c83b78 Fix peerpoke to also acknowledge
22 years ago
Mark Spencer 451970b3fb Minor extensions.conf update
22 years ago
Mark Spencer 9cc2aa39d0 Unlink local channels
22 years ago
Mark Spencer 4952835d93 Fix some leaks in CLI (thanks James)
22 years ago
Mark Spencer 09e1c34d99 work around buggy redhat 9.o installs
22 years ago
Mark Spencer 2f771c677e After 404 or 484, don't send 503
22 years ago
Mark Spencer 3e8183de43 Add expiry patch from Jim
22 years ago
Mark Spencer d6b38a8b3b Fix retransmission on BYE
22 years ago
Mark Spencer c88abff66a Changelog update
22 years ago
Martin Pycko 60ed66900b Fix a typo
22 years ago
Martin Pycko cd3a6739a4 Add more debug messages for bridging
22 years ago
Mark Spencer fd6cc08ac8 Phone updates
22 years ago
Mark Spencer 736bb7c6dc Fix phone crash
22 years ago
Martin Pycko c0aac70a7b Fix little leak thanks to James G.
22 years ago
Jeremy McNamara 451e3aa6dd fix gatekeeper logic
22 years ago
Mark Spencer 5c18d407c5 Fix contact header, don't transmit BYE when INVITE are still outstanding
22 years ago
Mark Spencer 1680ccf66c Provide authentication for BYE and CANCEL requests
22 years ago
Mark Spencer 85cb2844aa Various management updates
22 years ago
Mark Spencer 67eafb9b74 Fix selection of tag in reqprep
22 years ago
Mark Spencer c42b5f915d Add steven's SIP update
22 years ago
Mark Spencer 201246ea7f PBX updates from andre
22 years ago
Mark Spencer f8755643cd Merge enhanced status changes, add SIP subscribe from Andre
22 years ago
Mark Spencer 77b48c0aed Make call forwarding work properly by using new Local channel
22 years ago
Mark Spencer 864fc0cfce Complete local channel implementation
22 years ago