| Age | Commit message (Collapse) | Author |
|---|
|
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines
Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly.
(issue #14460)
Reported by: moliveras
Tested by: russell
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
(closes issue #14460)
Reported by: moliveras
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines
Don't send DTMF for infinite time if we do not receive an END event.
I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf,
dtmftimeout, that was intended to handle this situation. However, in between
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.
The default timeout is 3 seconds. However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:
Limiting the time period of extending the tone is necessary
to avoid that a tone "gets stuck". Regardless of the
algorithm used, the tone SHOULD NOT be extended by more than
three packet interarrival times. A slight extension of tone
durations and shortening of pauses is generally harmless.
Three seconds will pretty much _always_ be far more than three packet
interarrival times. However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.
Code from svn/asterisk/team/russell/issue_14460
(closes issue #14460)
Reported by: moliveras
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines
Don't crash if RTCP is not enabled on an RTP structure but statistics are output.
(closes issue #14234)
Reported by: jcovert
Patches:
rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
rtp.c.patch-svn-165599 uploaded by jcovert (license 551)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines
Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
(closes issue #13545)
Reported by: davidw
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines
Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
(closes issue #12983)
Reported by: vt
Patches:
dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
(closes issue #13209)
Reported by: ip-rob
Patches:
13209.diff uploaded by file (license 11)
Tested by: ip-rob, bujones
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines
Take video into account when early bridging RTP.
(closes issue #13535)
Reported by: davidw
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines
(closes issue #13835)
Reported by: matt_b
Tested by: jpeeler
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Closes AST-142.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines
Remove the potential for a division by zero error.
(Closes issue #13810)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) | 6 lines
Allow for "G.729" if offered in an SDP even though
it is not RFC 3551 compliant. Some Cisco switches
will send this in an SDP, and it doesn't hurt to
be able to accept this.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue #13319)
Reported by: srt
Patches:
13319_rtcp_manager_event_headers.diff uploaded by srt (license 378)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
files from main/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines
Since adding the AST_CONTROL_SRCUPDATE frame type,
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
(closes issue #13247)
Reported by: pj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines
Fix a problem where inbound rfc2833 audio would be sent to the
core instead of being P2P bridged. When the core regenerated
the rfc2833 packet for the outbound leg, the SSRC would be different
than the RTP audio on the call leg causing DTMF detection issues on
the far end.
(closes issue #12955)
Reported by: tonyredstone
Patches:
dynamic_rtp.patch uploaded by tsearle (license 373)
Tested by: tonyredstone
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue #13002)
Reported by: caio1982
Patches:
janitor_arraylen5.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(if we have RTCP reports
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) | 7 lines
Check for rtcp structure before trying to delete schedule.
(closes issue #12872)
Reported by: destiny6628
Patches:
20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
Tested by: destiny6628
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines
Add ast_assert(), which can be used to handle fatal errors. It is only compiled
in if dev-mode is enabled, and only aborts if DO_CRASH is defined.
(inspired by issue #12650)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 lines
Don't change the SSRC when a new source comes into play, this might happen quite often and depending on the remote side... they might not like this.
(closes issue #12353)
Reported by: dimas
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue #12409)
Reported by: candlerb
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines
Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things.
(closes issue #12212)
Reported by: bamby
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
to reload any file that references a given configuration file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines
Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue #11429)
Reported by: sperreault
Patches:
11429-frametype.diff uploaded by qwell (license 4)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines
Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value.
(AST-2008-002)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines
Properly initialize rtp->schedid
(Closes issue #12154)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.
main/rtp.c:
- When a G722 frame is read from the smoother, the number of samples in the
frame must be divided by 2 before being sent out over the network. Even
though G722 is 16 kHz, an error in some previous spec has made it so that
we have to list the number of samples such as if it was 8 kHz.
main/file.c:
- When scheduling the next time to expect a frame, take into account that the
format of the file we're reading from may not be 8 kHz.
codecs/codec_g722.c:
- When converting from G722 to slinear, g722_decode() expects its samples
parameter to be in the silly (real samples / 2) format. Make it so.
- When converting from slinear to G722, properly set the number of samples in
the frame to be the number of bytes of output * 2.
formats/format_pcm.c:
- This format module handles G722, among a number of other formats. However,
the read() and seek() functions did not account for the fact that G722 has
2 samples per byte.
(closes issue #12130, reported by rickross, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines
Fix a bug that I just noticed in the RTP code. The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use. The len field
represents the number of ms of audio that the frame contains. It would have
set the value to be twice what it should be.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines
In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue #12078)
Reported by: dimas
Patches:
v1-12078.patch uploaded by dimas (license 88)
(closes issue #12074)
Reported by: Ivan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines
When a SIP channel is being auto-destroyed, it's possible for it to still be
in bridge code. When that happens, we crash. Delay the RTP destruction until
the bridge is ended.
(closes issue #11960)
Reported by: norman
Patches:
20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: norman
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
- change function to void because it always returned the same value and no one read it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
