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Voice
asterisk
Repository
2f67f665779c1cbc714c9757143769e79845fd5c
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20
fix_transport_fallback_at_dns_failure
16302-emergency-call-update
14600-wenpeng
asterisk_rdkb
fix_fallback
14666_fxs_no_wideband_codec
fix_srv_records
wenpeng-test-250121
wenpeng-registration-test
fix_deadlock_in_bridge_peer_functions
devel
default
protected
securesip_pjproject_part1
PJ_Example_Review
fix_re_register_authorization_header
fix_internal_calls_auth
fix_internal_calls
Secure_SIPtelephony_part1
SIPtelephony_DECT_syslog
syslog_develop_phase1
merge-asterisk-20.8.1
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22.0.0-pre1
21.4.2
20.9.2
18.24.2
certified-20.7-cert2
certified-18.9-cert11
21.4.1
20.9.1
18.24.1
21.4.0
20.9.0
18.24.0
certified-20.7-cert1
certified-18.9-cert10
21.4.0-rc1
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18.24.0-rc1
21.3.1
20.8.1
18.23.1
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git@dev.iopsys.eu:voice/asterisk.git
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https://dev.iopsys.eu/voice/asterisk.git
Allow "auto" dtmf mode to select between RFC2833 and inband based on peer's offer or answer.
Mark Spencer authored
19 years ago
git-svn-id:
https://origsvn.digium.com/svn/asterisk/trunk@6518
65c4cc65-6c06-0410-ace0-fbb531ad65f3
2f67f665
History
2f67f665
19 years ago
History
Name
Last commit
Last update
agi
apps
build_tools
cdr
channels
codecs
configs
contrib
db1-ast
doc
editline
formats
funcs
images
include
keys
patches
pbx
redhat
res
sounds
stdtime
utils
.cleancount
.cvsignore
BUGS
CHANGES
CREDITS
HARDWARE
LICENSE
Makefile
README
README.fpm
SECURITY
UPGRADE.txt
acl.c
aescrypt.c
aeskey.c
aesopt.h
aestab.c
alaw.c
app.c
ast_expr.y
ast_expr2.c
ast_expr2.fl
ast_expr2.h
ast_expr2.y
ast_expr2f.c
asterisk.8.gz
asterisk.c
asterisk.sgml
astmm.c
autoservice.c
callerid.c
cdr.c
channel.c
chanvars.c
cli.c
coef_in.h
coef_out.h
config.c
db.c
devicestate.c
dlfcn.c
dns.c
dnsmgr.c
dsp.c
ecdisa.h
enum.c
file.c
frame.c
fskmodem.c
image.c
indications.c
io.c
jitterbuf.c
jitterbuf.h
loader.c
logger.c
manager.c
md5.c
mkpkgconfig
muted.c
muted.conf.sample
netsock.c
pbx.c
plc.c
poll.c
privacy.c
rtp.c
sample.call
say.c
sched.c
slinfactory.c
sounds.txt
srv.c
strcompat.c
tdd.c
term.c
translate.c
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README