Files · c65f1a04ab9cb4ddabdcf582b9337264e113c14a · Voice / asterisk · GitLab

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there is no reason to check the channeltype of the channel calling SIPAddHeader

Russell Bryant authored 19 years ago

because it really only matters what the channeltype is of the *created* channel
in the subsequent dial (bug #4896)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

c65f1a04

c65f1a04 19 years ago

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pbx
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BUGS
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acl.c
aescrypt.c
aeskey.c
aesopt.h
aestab.c
alaw.c
app.c
ast_expr.y
ast_expr2.fl
ast_expr2.y
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
config_old.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
ulaw.c
utils.c