Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity
3129 Commits
153 Branches
1671 Tags
316 MiB
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from 'b1d9a040d4'
${ noResults }
Commit Graph

81 Commits (b1d9a040d457e1e79347869900a8743a67bd5e2a)

Author SHA1 Message Date
Mark Spencer 1d473f25dd Add outgoing OSP support (SIP only at this point)
23 years ago
Mark Spencer 2187465cae Set ${DIALSTATUS} from app_dial, make zap return BUSY on a phone that is busy, update macro-stdexten to demo
23 years ago
James Golovich f5c18f62a1 Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h).
23 years ago
Mark Spencer 5546e32355 Merge major BSD mutex and symbol conflict patches (bug #1816) (link patch still pending)
23 years ago
Mark Spencer 8b2b0278c9 Make sure outgoing is not NULL
23 years ago
Mark Spencer 2efcb2e331 Don't stop moh until after announcement finishes (bug #789)
23 years ago
Mark Spencer 502e74e5db Fix app_dial buglet
23 years ago
Anthony Minessale II b168dcdddf I don't know what this last update was for but app_dial.c was broken
23 years ago
Mark Spencer 585afa7e8e Add promiscuous redirect option
23 years ago
James Golovich a054fec8ef Typo in comment in app_dial.c
23 years ago
Mark Spencer bec994b037 Merge Matt's work
23 years ago
Anthony Minessale II cc36c3d475 fix D() issue with to short of a duration
23 years ago
Mark Spencer 194fe2a4a5 Copy hangup cause at end of a good call
23 years ago
Mark Spencer 61ce929656 Merge Mike's announcement enhancements
23 years ago
James Golovich 36f888e06e Clarify app_dial documentation regarding transfers
23 years ago
James Golovich 46b24e8a27 Code cleanup
23 years ago
Anthony Minessale II 66b96f417c Created API call ast_dtmf_stream
23 years ago
Mark Spencer 19a080ddd7 strlen fixes and don't destroy SIP channels that still have pending things on them (bug #1552 et al)
23 years ago
Anthony Minessale II f9226cd9cd Bug Fix
23 years ago
Anthony Minessale II 5385ca0a0e applied final release of bug 1353 per Mark's permission
23 years ago
James Golovich 36a8fb3e4d Make app_dial complain if the timeout passed is non numeric
23 years ago
Mark Spencer bc2a510fb7 Remember when we started/stopped ringing so that ringing twice works as expected
23 years ago
Mark Spencer 4c06525432 Make RDNIS really work right
23 years ago
Mark Spencer ce88c6dd08 Populate RDNIS information
23 years ago
Mark Spencer 7f2e23bb50 Add absolute timeout option to dial that applies when call is up (bug #1107)
23 years ago
James Golovich e8c21dfe67 Check result of malloc in app_dial.c
23 years ago
James Golovich 84cbebcf17 Fix another typo in the app_dial description
23 years ago
James Golovich 9028579d84 Spelling error in app_dial.c
23 years ago
Martin Pycko 3d40b1c03f A little adjustment to the previous one (passing the clid&ani on call_forward)
23 years ago
Martin Pycko e993855421 Pass the callerid & ani to the new b-leg if the current b-leg requests call forwarding
23 years ago
Mark Spencer 036cb819d6 X out announce stuff (bug #760)
23 years ago
Martin Pycko f96c0762ff Allow doing digital PRI to PRI calls automatically
23 years ago
Jeremy McNamara a91dc75020 fix minor typos in descrip. Bug#635
23 years ago
Mark Spencer 26411c1021 Forward language choice
23 years ago
Mark Spencer af19291e01 I have no idea what the heck that was supposed to be
23 years ago
Jim Dixon e1f471f89c Got rid of un-necessary 'c' and 'd' options in app_dial.
23 years ago
Jeremy McNamara f70f05ca35 minor language tweak. Thanks Carp
23 years ago
Mark Spencer 98166ff081 WHen making compatible for bridge, stop generators, etc.
23 years ago
Mark Spencer dd5a2c44a5 Add announce feature to dial
23 years ago
Martin Pycko c0dde4ac5b Add 486 message (SIP) to 17 cause (PRI) translation when you call Dial app and then Hangup:
23 years ago
Martin Pycko cacfd42960 Add usecallingpres keyword to zapata.conf for PRI to PRI call so that we would take the presentation mode and send it to outbound call.
23 years ago
Mark Spencer 8785c02365 Copy musiconhold
23 years ago
Martin Pycko 6e8a074912 Add restrictcid=[yes|no] for chan_zap and chan_sip. You can specify if the sip/zap users will have the callerid transmitted as ANI or CALLERID.
23 years ago
Mark Spencer 3ee27a8a89 Allow "g" option in dial to go to next extension (bug #189)
23 years ago
Martin Pycko c7eef6e141 Save the dstchannel before the call gets answered so that we would know what destination channel was dialed when we get the channel hanged up
23 years ago
Mark Spencer 21b6696e8b Add alternate queueing strategies. Implment ringall, roundrobin, and random
23 years ago
Mark Spencer 159de43db3 Make sure o->chan is non-NULL
23 years ago
Martin Pycko da17bb3e97 Fix the compile problem
23 years ago
Mark Spencer 7fa873f2b2 Don't hold channel resources longer than we need them
23 years ago
Mark Spencer 26fab30972 Fix small leak on app_dial
24 years ago