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r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar 2008) | 41 lines
Merged revisions 108737 via svnmerge from
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r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines
Fix a race condition in the SIP packet scheduler which could cause a crash.
chan_sip uses the scheduler API in order to schedule retransmission of reliable
packets (such as INVITES). If a retransmission of a packet is occurring, then the
packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if
a response is received from the packet as previously transmitted, then when we
ACK the response, we will remove the packet from the scheduler and free the packet.
The problem is that both the ACK function and retrans_pkt attempt to acquire the
same lock at the beginning of the function call. This means that if the ACK function
acquires the lock first, then it will free the packet which retrans_pkt is about to
read from and write to. The result is a crash.
The solution:
1. If the ACK function fails to remove the packet from the scheduler and the retransmit
id of the packet is not -1 (meaning that we have not reached the maximum number of
retransmissions) then release the lock and yield so that retrans_pkt may acquire the
lock and operate.
2. Make absolutely certain that the ACK function does not recursively lock the lock in
question. If it does, then releasing the lock will do no good, since retrans_pkt will
still be unable to acquire the lock.
(closes issue #12098)
Reported by: wegbert
(closes issue #12089)
Reported by: PTorres
Patches:
12098-putnopvutv3.patch uploaded by putnopvut (license 60)
Tested by: jvandal
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r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) | 18 lines
Merged revisions 108530 via svnmerge from
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r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines
Make a tweak that gets the LEDs on polycom phones to blink when an extension that
has been subscribed to goes on hold. Otherwise, they just stay on like it does
when an extension is in use.
(closes issue #11263)
Reported by: russell
Patches:
notify_hold.rev1.txt uploaded by russell (license 2)
Tested by: russell
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r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines
Merged revisions 108288 via svnmerge from
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r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines
Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip.
The scheduler callback will always return 0. This means that this id
is never rescheduled, so it makes no sense to loop trying to delete
the id from the scheduler queue. If we fail to remove the item from the
queue once, it will fail every single time.
(Yes I realize that in this case, the macro would exit early because the
id is set to -1 in the callback, but it still makes no sense to use
that macro in favor of calling ast_sched_del once and being done with it)
This is the first of potentially several such fixes.
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r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines
Merged revisions 108086 via svnmerge from
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r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines
if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP
closes issue #11475
Reported by: andrebarbosa
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r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar 2008) | 10 lines
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r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines
don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"
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r105734 | russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines
Fix some bugs in the SIP tcp helper thread.
- fix a spot where a lock wouldn't get unlocked in an error condition
- call ast_mutex_destroy() on the lock before freeing its memory
(related to issue #11972)
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r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines
Merged revisions 105674 via svnmerge from
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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
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r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar 2008) | 15 lines
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r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines
when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
(closes issue #11917)
Reported by: mavetju
Tested by: mavetju
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automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6 lines
Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device.
(closes issue #9044)
Reported by: queuetue
Patches:
sip-gui-friend.diff uploaded by qwell (license 4)
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r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6 lines
If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog.
(closes issue #10727)
Reported by: s0l4rb03
Patches:
10727-2.diff uploaded by file (license 11)
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r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6 lines
Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank.
(closes issue #12061)
Reported by: flefoll
Patches:
chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244)
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r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19 Feb 2008) | 8 lines
Account for the fact that the "other" channel can disappear while the local pvt
is not locked.
(fixes a problem introduced in rev 100581)
(closes issue #12012)
Reported by: stevedavies
Patch by me
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Reported by: julianjm
Patches:
chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm (license 99)
Patch fixes problem of device state incorrectly reporting idle before PBX answers incoming call on FXO channel. Device status is updated now during new channel creation.
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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
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r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb 2008) | 10 lines
Fix a linked list corruption that under the right circumstances
could lead to a looped list, meaning it will traverse forever.
(closes issue #11818)
Reported by: michael-fig
Patches:
11818.patch uploaded by putnopvut (license 60)
Tested by: michael-fig
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(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!
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r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8 lines
Handle ACK and CANCEL in an invite transaction - even if we get INFO transactions during the actual call setup.
(closes issue #10567)
Reported by: jacksch
Tested by: oej
Patch by: oej inspired by suggestions from neutrino88 in the bug tracker
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r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) | 5 lines
Change the SDP_SAMPLE_RATE macro. It turns out that even though G.722 is 16 kHz,
it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed
to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but
people follow it anyway, because it's the spec ...)
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r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31 Jan 2008) | 8 lines
Add some more sanity checking on IAX2 dial strings for the case that no peer
or hostname was provided, which is the one part of the dial string that is
absolutely required. If it's not there, bail out.
(closes issue #11897)
Reported by sokhapkin
Patch by me
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r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008) | 2 lines
Add missing locking to the find_agent() function.
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r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31 Jan 2008) | 3 lines
Move the locking from find_agent() into the agent dialplan function handler to
ensure that the agent doesn't disappear while we're looking at it.
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broke realtime configurations which still used the "username" field. This was taken
care of properly when reading from realtime but was not handled properly when updating
a realtime peer. This change also adds a deprecation NOTICE CLI message that will print
if using the deprecated "username" field.
(closes issue #11880)
Reported by: cabal95
Patches:
11880.patch uploaded by putnopvut (license 60)
Tested by: cabal95
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r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008) | 5 lines
For some reason, the use of this strdupa() is leading to memory corruption on
freebsd sparc64. This trivial workaround fixes it.
(closes issue #10300, closes issue #11857, reported by mattias04 and Home-of-the-Brave)
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r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines
Make some deadlock related fixes. These bugs were discovered and reported
internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
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that was just merged from 1.4, so this is a changeover to those APIs to use the
macro versions, so that we properly detect errors from ast_sched_del, instead
of simply ignoring the return values.
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r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines
When deleting a task from the scheduler, ignoring the return value could
possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9 lines
Make sure we don't cancel destruction on calls in CANCEL state, even if we
get 183 while waiting for answer on our CANCEL.
(issue #11736)
Reported by: MVF
Patches:
bug11736.txt uploaded by oej (license 306)
Tested by: MVF
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r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 lines
Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language
over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old
head to avoid too heavy memory allocations on some systems.
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as a channel variable BRIDGEPVTCALLID
This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.
The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.
Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.
Inspired by (issue #11816)
Reported by: ctooley
Patch by oej
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r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 Jan 2008) | 12 lines
Fixing an issue wherein monitoring local channels was not possible. During a channel
masquerade, the monitors on the two channels involved are swapped. In 99% of the cases
this results in the desired effect. However, if monitoring a local channel, this caused
the monitor which was on the local channel to get moved onto a channel which is immediately
hung up after the masquerade has completed. By swapping the monitors prior to the masquerade,
we avoid the problem by tricking the masquerade into placing the monitor back onto the channel
where we want it.
During the investigation of the issue, the channel's monitor was the only thing that was swapped
in such a manner which did not make sense to have done. All other variable swapping made sense.
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is currently active for the Asterisk CLI, or to set it. Also, knock multiple device
support off of the to-do list.
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- Add support for multiple devices. All devices are configured in console.conf.
- Add "console list devices" CLI command to show configured devices. Also, changed
the old "list devices" to be "list available", which queries PortAudio for all
audio devices that are available for use.
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines
Have IAX2 optimize the codec translation path just like chan_sip does it. If
the caller's codec is in our codec list, move it to the top to avoid transcoding.
(closes issue #10500)
Reported by: stevedavies
Patches:
iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh
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This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Tested by: jtodd, rjain, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines
Fix a deadlock in chan_local in local_hangup. There was contention because
the local_pvt was held and it was attempting to lock a channel, which is the
incorrect locking order.
(closes issue #11730)
Reported by: UDI-Doug
Patches:
11730.patch uploaded by putnopvut (license 60)
Tested by: UDI-Doug
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines
Don't drop the old record route information when dealing with packets related to a reinvite.
(closes issue #11545)
Reported by: kebl0155
Patches:
reinvite-patch.txt uploaded by kebl0155 (license 356)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines
Change a buffer in check_auth() to be a thread local dynamically allocated
buffer, instead of a massive buffer on the stack. This fixes a crash reported
by Qwell due to running out of stack space when building with LOW_MEMORY defined.
On a very related note, the usage of BUFSIZ in various places in chan_sip is
arbitrary and careless. BUFSIZ is a system specific define. On my machine,
it is 8192, but by definition (according to google) could be as small as 256.
So, this buffer in check_auth was 16 kB. We don't even support SIP messages
larger than 4 kB! Further usage of this define should be avoided, unless it
is used in the proper context.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note: NoAnswer support is currently not implemented, as it would take a
significant amount of work to figure out how to do correctly.
Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines
1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running. Don't just ignore it.
(Closes issue #11698)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- support scrolling of message window;
- simplify the code for creating a message window,
and try it using a second one in the top of
the keypad (where we echo the dialed number).
The 'skin' that supports these two windows will be
committed separately.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines
Set the caller id within the gtalk_alloc function.
As underlined in issue #10437 by Josh, we need to prevent a possible
memory leak. We only set the name part of the caller id, the number
part is not relevant when dealing with JIDs.
Closes issue #11549.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a number to dial in the 'message' area under the
keypad.
Now you can make calls using the keypad as a regular phone
(or the keyboard for chars not present on the keypad)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
commands so you can start and stop the gui even outside
of a call. This is convenient for testing, and also for
using the keypad to pick up a call, and to dial a number
(the latter not yet implemented, but should be close).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main code to implement the textarea is in console_board.c,
and uses a simple png image with the font, blitting characters
on the designated areas of the main screen.
Additionally we provide some annotations in the image used
as a skin to indicate which areas are used for text messages.
(images will be committed separately).
At the moment the dialog area is only used to display a running
counter, just as a proof of concept.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This way it can contain additional elements (e.g. fonts, buttons,
widgets) without having to use a zillion files to store them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines
Make use of the temporary channel pointer while the pvt is unlocked.
(closes issue #11675)
Reported by: flefoll
Patches:
chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
of a hack. It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call. But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Russell Bryant
Mark Michelson
Kevin P. Fleming
Tilghman Lesher
Jason Parker
Terry Wilson
Matthew Fredrickson
Philippe Sultan
Olle Johansson
Steve Murphy
Doug Bailey
Jeff Peeler
Michiel van Baak
James Golovich
Christian Richter
Luigi Rizzo