78ec9a057c
clean up various whitespace issues (issue #7236 , casper)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 20:19:50 +00:00
414106a3a6
move the includes of abstract_jb.h to be with the rest of the asterisk includes.
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These used to be wrapped in a #ifdef
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 17:29:12 +00:00
0384330d64
update the rest of the channel drivers that use RTP so that their channel
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tech structures indicate that they create jitter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 17:21:21 +00:00
bb7dd96cfe
Add support for using a jitterbuffer for RTP on bridged calls. This includes
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854 , Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 16:56:50 +00:00
503486e99d
OKAY - Only replace our IP address in the outgoing SIP messages if the actual source address is within the specified local network. (issue #7051 reported by Luke-Jr with mods by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 16:31:41 +00:00
c7007e1f55
Add congestion message to SIP history
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-31 14:36:47 +00:00
bc0f0a3226
Use ast_channel_trylock for channels
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2006-05-31 14:28:29 +00:00
705b5459c1
Bug 7237 - Replace recoded thread_safe_rand with the existing ast_random API
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-30 20:24:40 +00:00
4790dd5d9e
when we receive an IAX2 registration request with both a plaintext secret and an MD5 challenge, prefer the MD5 challenge for authentation (reported on asterisk-dev)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-30 18:01:52 +00:00
1203424ce2
remove duplicate static keywords, oops
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2006-05-29 14:52:55 +00:00
1617adb055
make some variables static ... committed from xcode :)
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2006-05-27 18:47:44 +00:00
79161454a3
Few more expire_registry changes
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2006-05-26 20:00:48 +00:00
ec42807ebd
That goes for jingle too :)
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2006-05-26 05:21:41 +00:00
730f75cc78
only display a debug message if option_debug is in use
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-25 22:39:57 +00:00
f890cab292
Merged revisions 30293 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r30293 | kpfleming | 2006-05-25 12:18:01 -0500 (Thu, 25 May 2006) | 2 lines
allow SIPCHANINFO(peername) to work for calls from users as well (issue #7215 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-25 17:22:26 +00:00
e0612c9835
Safely traverse the thread lists and wait until each thread is done before moving on to the next.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-25 16:14:17 +00:00
336fd1a2de
Only get the low 16 bits if we actually have a message count
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2006-05-25 15:42:11 +00:00
eefc9193e7
added EVENT_NEW_CHANNEL. We change the channel name now when we got the real channel, also changed name generation to new stringfield api
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 23:21:03 +00:00
57f6093661
don't send CANCEL for an INVITE that we haven't received a provisional response for yet... mark it pending until a response arrives (issue #7079 )
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2006-05-24 20:37:45 +00:00
b8b1fda04c
Typo
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2006-05-24 19:58:36 +00:00
11d13e1d4b
Merged revisions 29969 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r29969 | kpfleming | 2006-05-24 11:17:26 -0500 (Wed, 24 May 2006) | 2 lines
respect 'usecallingpres' in zapata.conf even if CLID has not been set for the channel (issue #7123 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 16:19:57 +00:00
8122c35675
fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 07:58:52 +00:00
3e99be68d1
add a new option for 'obscuring' SIP user/peer names from fishers
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use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 03:28:49 +00:00
7370cb3221
Merged revisions 29849 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r29849 | russell | 2006-05-23 17:44:52 -0400 (Tue, 23 May 2006) | 2 lines
fix the sourceaddress option (issue #7213 , alphaque)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23 21:46:26 +00:00
547c6ebd86
Bump up the echo tail length option
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2006-05-23 21:10:55 +00:00
39ac1a5b83
added a l1watcher timeout, therefore removed the old behaviour of guessing the l1state.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23 19:40:16 +00:00
b31af470e7
Merged revisions 29764 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r29764 | kpfleming | 2006-05-23 13:16:40 -0500 (Tue, 23 May 2006) | 2 lines
simplify/fix lock retry, and fix comment
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2006-05-23 18:17:40 +00:00
c4227714fd
Sanity check code for an extended failure in trying to obtain a channel lock that may have been obtained elsewhere. Prevents the monitor thread of the SIP module from going into an infinite loop, effectively, breaking SIP until you restart Asterisk or the mutex is unlocked, whichever comes first.
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2006-05-23 17:31:05 +00:00
5d51260c36
finish cleaning up some more stuff before russell
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gets a chance to.
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2006-05-23 16:43:58 +00:00
32a28ed9cf
update chan_jingle to reflect the recent change to the indicate prototype
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23 16:25:37 +00:00
16a648aaf1
Breaking once will stop us... :-)
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2006-05-23 12:39:48 +00:00
adfae0960f
* export_ies uses now _VAR else the vars are not copied to the dest chan
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* when receiving a connect from the NT Side we wait until we have the final
l3id until we queue the answer to asterisk to avoid bridging conflicts
* when not bridged to misdn we had a segfault after receiving the connect
due to a strcasecmp bug.. this didn't happen before, cause we hadn't had
the bridge before
* cleanup of the bchannels is queued now, due to possible race conditions
* added mISDN_clear_stack when cleaning the bchannel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23 12:38:06 +00:00
883e4d6a9b
Code formatting
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2006-05-23 12:14:35 +00:00
fa9e0ed768
Handle ringing (early) state properly on SIP
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2006-05-23 04:49:31 +00:00
7aa1a77e75
asterisk-xmpp merge in
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2006-05-22 21:12:30 +00:00
a0f95cd526
Merged revisions 29464 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r29464 | file | 2006-05-22 13:33:03 -0300 (Mon, 22 May 2006) | 2 lines
Preserve presentation bit when going through chan_local (issue #7002 reported by acunningham)
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2006-05-22 16:36:13 +00:00
19d46333bf
added callcounters for incoming and outgoing calls
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2006-05-22 15:02:03 +00:00
76c47813fe
remove code duplication generating some ast_verbose messages .
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 03:16:38 +00:00
82dbf5a4c1
use the functional form of time(), and mark XXX a couple of places
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with dubious assignments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 02:43:24 +00:00
afe7c46924
more constifications
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2006-05-22 02:31:58 +00:00
322aff00c6
various constification of function arguments to prevent erroneous
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usage and to ease understanding of the function themselves.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 02:27:33 +00:00
ba195ee2d4
the msg argument of the various transmi_response*() functions is const.
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2006-05-22 02:22:39 +00:00
75c6b453a3
mark const the 'header' argument of gettag
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2006-05-22 02:19:14 +00:00
1a709886e3
mark const the source of copy_request
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2006-05-22 02:17:52 +00:00
c76882a94a
fix indentation to make a table more readable
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2006-05-22 02:16:04 +00:00
3ade391d1d
simplify register_peer_exten() removing some duplicated code.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 02:02:57 +00:00
734535011a
use strsep() to trim away ;xxx from SIP messages.
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Note that in a couple of places the code overwrites the string
in the buffer - mark the two places with XXX but do not fix the
bug yet.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 01:27:52 +00:00
8414445108
use a more compact method to trim the trailing ;... from SIP lines.
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These are just the first two instances, there are many more to replace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 01:08:41 +00:00
917182e286
annotate const parameters of copy_header()
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2006-05-22 00:46:04 +00:00
27f4fb4990
src is const in parse_copy()
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2006-05-22 00:32:36 +00:00
564ca6394a
minor logic simplification in get_sip_pvt_byid_locked()
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2006-05-22 00:29:12 +00:00
15b7465cd2
simplify logic in function create_addr()
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2006-05-22 00:17:46 +00:00
a234f92337
simplify logic in realtime_peer(), removing an unnecessary
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initialization, an extra variable (newpeername) and a
variable newpeername, and an 'else' branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22 00:07:48 +00:00
94d3a99089
strcmp() is the right function to use here, not strncmp().
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On passing simplify an ast_cli message nearby.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-21 23:54:28 +00:00
ed0b86c916
fix the possibility of writing one byte past the end of a buffer.
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(issue #7189 , Mithraen)
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2006-05-20 19:54:14 +00:00
df6767fa52
Merged revisions 28896 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28896 | kpfleming | 2006-05-19 19:55:31 -0500 (Fri, 19 May 2006) | 2 lines
don't try to predict where the compiler will place things on the stack... put them in the right place explicitly (issues #7029 and #7100 , maybe others)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-20 01:01:20 +00:00
6d92d53d41
attributes need trailing CR-LF
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2006-05-19 23:08:47 +00:00
0debd3dec2
allow chan_h323 to build (issue #7006 , casper)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-19 22:55:09 +00:00
42e2cf61d8
Merged revisions 28794 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28794 | kpfleming | 2006-05-19 14:39:55 -0500 (Fri, 19 May 2006) | 2 lines
use the specified 'subscribecontext' for a peer rather than the context found via the target domain (domain contexts are for calls, not for subscriptions) (issue #7122 , reported by raarts)
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2006-05-19 19:43:33 +00:00
d0c36296d2
As requested by kpfleming, renaming messagecount to inboxcount and messagecount2 to messagecount.
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2006-05-19 18:21:31 +00:00
ebc762c82e
Clean up some chan_zap code to make it more readable. (issue #6834 reported by ppyy)
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2006-05-19 17:15:29 +00:00
fa13a5f113
Merged revisions 28698 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28698 | file | 2006-05-19 14:04:02 -0300 (Fri, 19 May 2006) | 2 lines
Make the minidle option actually exist as documented (issue #7159 reported by imran)
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2006-05-19 17:05:22 +00:00
5d24426e40
fix up a patch that did not merge properly from 1.2 ... weird
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(issue #6866 , flefoll)
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2006-05-19 14:51:08 +00:00
1783cb3a91
Adding send_text capability to chan_local
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2006-05-19 08:20:01 +00:00
908c153840
First stab at supporting video in chan_local
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2006-05-19 08:03:28 +00:00
a3cc752008
Add simple devicestate for chan_local
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2006-05-19 07:56:56 +00:00
cd03597f0c
Formatting cleanup
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2006-05-19 07:52:08 +00:00
0c73d47618
Merged revisions 28380,28384 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28380 | kpfleming | 2006-05-18 15:24:07 -0500 (Thu, 18 May 2006) | 2 lines
handle incoming multipart/mixed message bodies in SIP and find the SDP, if presnet (issue #7124 reported and patched by eborgstrom, but very different fix)
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r28384 | kpfleming | 2006-05-18 15:43:42 -0500 (Thu, 18 May 2006) | 2 lines
fix up a few more places to find the SDP properly (fallout from fix for #7124 )
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2006-05-18 21:02:21 +00:00
42cf0b0a8f
add another media path reinvite 'flavor', where we will only redirect our media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
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also, documented the 'canreinvite=update' option in the sample config file
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2006-05-18 16:57:59 +00:00
48c36f4b89
adding new code should require following the formatting guidelines :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18 16:38:26 +00:00
6d603ec09c
Allow contexts in regexten so that extensions can be added to multiple contexts when peer registers (issue #6869 reported by and created by Marquis)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18 14:07:46 +00:00
4694f5e02e
this is not a warning, it might happen during call initialization
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18 08:45:43 +00:00
5769ed6ea2
Fix receiving message count information from a remote IAX2 peer. (issue #7163 reported by and fixed by akohlsmith)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-18 00:12:54 +00:00
a6480b61b0
Merged revisions 27973 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r27973 | file | 2006-05-17 19:34:08 -0300 (Wed, 17 May 2006) | 2 lines
Fix codec priority stuff during authentication (issue #6194 reported by jkoopmann)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 22:35:53 +00:00
6eb1d09ad9
Only set the RTP DTMF information of an RTP structure does indeed exist.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 22:02:15 +00:00
10c0ed16f0
Issue #7176 - Crash in expire_register (imported from 1.2)
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This is a bad fix, since peer should never be 0. Why is it zero
on Royk's system?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 19:37:51 +00:00
270d5f6191
Adding a debug line
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 18:19:17 +00:00
15358932ec
Add distinctive ring detection with Caller ID for Australia, New Zealand, and other countries. (issue #3596 reported by deon patch by dbowerman with minor mods by moi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 16:51:47 +00:00
3fb070853b
fixed bug that we've lost the orginator of the call.. now we can hear to disconnect indications again
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 15:29:01 +00:00
770a4a30c5
Use pointer instead of un-needed buffer. Thanks Kevin!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-17 06:18:09 +00:00
c3dce57d95
mark RTP sessions that are not carrying DTMF
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allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-16 22:11:02 +00:00
99648dd713
Add option for enabling and disabling echo cancellation
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-16 21:43:20 +00:00
eeb2184920
Send a=recvonly when put on hold, otherwise a=sendrecv to indicate that we want two streams.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-16 20:13:10 +00:00
e4c0a90649
Show all formats for an active SIP call
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-16 18:06:27 +00:00
a309a2db9f
fixed very bad memory overwrite which leaded to segfaults on some systems. removed redundant buffer betweend mISDN reading thread and ast_read in favour of the already existing pipe, this clarifies the way a voice frame takes between mISDN and asterisk a lot. centralized debugging of NumberPlan. removed a compiler warning.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-16 14:34:21 +00:00
c625a539be
One sipfrag attachment is enough...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-16 12:44:04 +00:00
a3ed43f0dc
Move WITH_SMDI from the Makefiles to be handled by menuselect. It will now be
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defined in buildopts.h. Also, remove a few more stray spaces in the gcc commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-13 10:54:40 +00:00
f9c578a8a0
- The recent change to linklists.h broke the build on linux for some reason.
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So, I have removed all of the uses of AST_LIST_HEAD_INIT and replaced them
with the equivalent static initializations.
- On passing, fix a memory leak in the unload_module() function of chan_agent.
The agents list mutex was never destroyed, and the elements in the agents
list were not freed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 20:07:44 +00:00
87e909476d
simplify determine_firstline_parts
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 14:55:34 +00:00
f3e6fa56ab
remove last instance of add_blank_header()
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 14:47:12 +00:00
b87aea8d60
remove almost all useless instances of add_blank_header() but one,
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for which i need to investigate a bit more (but i am almost 100%
sure that one is also useless).
I do the commit in two steps so it is easier to track the problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 14:35:06 +00:00
5237a0e06d
- Use systemname for realm in sip, if we have no configuration for realm
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- Optionally send systemname in manager (cool when you have a manager proxy)
- Use systemname in CLI prompt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 13:54:00 +00:00
945ec73b09
remove trailing space
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 10:26:39 +00:00
589241e300
remove a duplicate prototype
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 10:26:00 +00:00
96478ea3a2
remove duplicated code in add_header
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 10:22:18 +00:00
d06427978c
simplify init_req()
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As the rest of the chan_sip.c changes i am committing, this is
Approved by: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 09:38:53 +00:00
5cd1ec4e17
cleanup init_resp with proper variable names and arguments.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 09:36:10 +00:00
e08e646252
staticize a variable.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 09:32:51 +00:00
798b82e184
replace list unlinking with UNLINK macro
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(when the list becomes an astobj this will be revisited;
however at the moment the change would be too intrusive).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 09:29:37 +00:00
067c472abb
simplify sip_pretend_ack also removing a bug in the existing
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code which called ast_skip_blanks erroneously.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-11 09:24:35 +00:00
Russell Bryant
Joshua Colp
Olle Johansson
Tilghman Lesher
Kevin P. Fleming
Mark Spencer
Christian Richter
Matthew Fredrickson
BJ Weschke
Matt O'Gorman
Luigi Rizzo