https://origsvn.digium.com/svn/asterisk/trunk
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r138311 | jpeeler | 2008-08-15 18:46:09 -0500 (Fri, 15 Aug 2008) | 20 lines
Merged revisions 138119,138151,138238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines
Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards).
(closes issue #11017)
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r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line
declared static mutexes using AST_MUTEX_DEFINE_STATIC macro
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r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line
initialize condition variable ss_thread_complete using ast_cond_init
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r137848 | tilghman | 2008-08-14 11:52:43 -0500 (Thu, 14 Aug 2008) | 17 lines
Merged revisions 137847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines
When creating the secondary subchannel name, it is necessary to compare to
the existing channel name without the "Zap/" or "DAHDI/" prefix, since our
test string is also without that prefix.
(closes issue #13027)
Reported by: dferrer
Patches:
chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525)
(Slightly modified by me, to compensate for both names)
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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
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r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) | 9 lines
In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID
was perverted. This change reverts IAX2 to the original meaning, which was,
that the callerid set on the client should be overridden on the server, even if
that means the resulting callerid is blank. In other words, if you set
"callerid=" in the IAX config, then the callerid should be overridden to blank,
even if set on the client. Note that there's a distinction, even on realtime,
between the field not existing (NULL in databases) and the field existing, but
set to blank (override callerid to blank).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
never indicate a match, so nothing would have been returned anyway, but it was
still a poor example of proper usage.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to get rid of a shadow warning (but this seemed legitimate enough to fix
here instead of in my branch).
Thanks to putnopvut for taking a look as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This set of changes enhances IAX2 encryption support by adding key rotation
to provide enhanced security. The key used for encryption is rotated right
after the call gets set up, and then again every few minutes. This was
discussed at the last AstriDevCon. For interoperability with older versions
of Asterisk, there is an option that disables key rotation.
(closes issue #13018)
Reported by: bbryant
Patches:
07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
Tested by: russell, bbryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This ensures that we don't just keep a cache of tiny frames, continually doing
an alloc/free for each data frame, thus negating the point of having a cache.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines
Fix some errant device states by making the devicestate API more strict in
terms of the device argument (only without the unique identifier appended).
(closes issue #12771)
Reported by: davidw
Patches:
20080717__bug12771.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw, jvandal, murf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Use ast_strlen_zero in one place
- check for successful string comparison the way most of Asterisk code does it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines
Small cleanup. Move the declaration of the DAHDI_SPANINFO
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.
Thanks to Tzafrir for pointing this out on #asterisk-dev
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r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line
another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users.
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Allow Spiraled INVITEs to work correctly within Asterisk.
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
(closes issue #7403)
Reported by: stephen_dredge
Modified:
branches/1.4/channels/chan_sip.c
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r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines
ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense'
(closes issue #12160)
Reported by: tzafrir
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r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
commands that conflicted with adding new features to the newer debug
commaands.
(closes issue #13103)
Reported by: mvanbaak
Patches:
2008071901__issue13103_iax2_set_debug_peer.diff uploaded by
mvanbaak (license 7)
Tested by: bbryant, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
buffers=<num of buffers>,<policy>
Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) | 7 lines
Revert part of issue #5620 (revision 6965) as it appears that it was in error.
This should fix talk call progress on analog lines.
(closes issue #12178)
Reported by: michael-fig
Patches:
20080717__bug12178.diff.txt uploaded by Corydon76 (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
instead. DEVICE_STATE is a state change on one server, and DEVICE_STATE_CHANGE is
the "real" state of that device across all servers sharing state. This would have
only been a problem with distributed device state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines
Override the callerid in all cases when the callerid is set in the user, not
just when a remote callerid is set. Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
Reported by: dimas
Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
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r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines
Ensure that a destination callno of 0 will not match for frames that do not
start a dialog (new, lagrq, and ping).
(closes issue #12963)
Reported by: russellb
Patches:
chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
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r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines
Pass the devicestate from an underlying channel up through the Agent channel.
This should make the Agent always report the correct device state, even when
the underlying channel is used for other purposes.
(closes issue #12773)
Reported by: davidw
Patches:
20080710__bug12773.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw
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r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042)
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r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
Reported by: ibc
Patches:
20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: ibc
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r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines
Don't hangup the call if we can't resolve the Contact if there's a proxy
route set for the call.
----
This comment was added a while ago and today it hit me badly.
/* OEJ: Possible issue that may need a check:
If we have a proxy route between us and the device,
should we care about resolving the contact
or should we just send it?
*/
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r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines
Fix handling of when a pvt disappears. Properly return the pvt locked
and don't hold the pvt lock while destroying the ast_channel.
(closes issue #13014)
Reported by: jpgrayson
Patches:
chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492)
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r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines
By using the iaxdynamicthreadcount to identify a thread, it was possible
for thread identifiers to be duplicated. By using a globally-unique monotonically-
increasing integer, this is now avoided.
(closes issue #13009)
Reported by: jpgrayson
Patches:
chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note: I don't think we can start properly without UDP port open, that needs to be tested.
- Removing "bindport" from configuration example, not needed to mention this any more
I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
...trying to get my head around the thoughts behind the TCP/TLS stuff
and figure out what needs to be done to make it useful...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Actually, kill the in-memory structure for type=user and start using the sip_peer
structure for every object. Have only one in-memory list and use them different
ways depending on type=user, type=peer and type=friend - like always.
The idea with this first patch is that configurations should work as before.
Some additional features for realtime peers. By adding a type= field, you
can now have multiple functions.
Let's test this for a while. Won't be integrated in 1.6.0, only in trunk,
for testing.
There's propably more to clean up and simplify here. Help is welcome
and encouraged!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
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require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses
that are related to INVITEs and re-INVITEs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It's mixing peers and users in a strange way and should really not be a CLI command,
since it's not meant for human output. It should be done with an app connecting to
manager.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
recommended in RFC 3261, instead of being hardcoded to 32 seconds. This is
important for LANs, as it allows autocongestion to occur much more quickly, if
desired by the local PBX administrator. It also corrects a bug: if the T1
timer was increased beyond 500ms, then timer B would have been set at a much
lower value than recommended.
(closes issue #12544)
Reported by: kactus
Patches:
20080616__bug12544.diff.txt uploaded by Corydon76 (license 14)
Tested by: kactus
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
fix a bug which caused a crash when a videodevice was
specified after startgui=1 in the config file. This also
involves a slightly different method to determine if the
gui is active or not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Merged revisions 126516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines
Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and
also fail if we don't get the very same precious ACK. Based on patch by tsearle, with
my own additions.
(closes issue #12951)
Reported by: tsearle
Patches:
busy_retransmit.patch uploaded by tsearle (license 373)
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giving you support for up to 9 video sources (e.g. webcams,
or X11 grabbers, etc.) active at once, displaying thumbnails for
each of them in the main GUI window, and with the ability to switch
between them on the fly during a conversation.
The code also implements a 'Picture in Picture' feature,
allowing you to select any source as primary or secondary,
and move the PiP window by just dragging it with the mouse.
The window looks like this:
________________________________________________________________
| ______ ______ ______ ______ ______ ______ ______ |
| | tn.1 | | tn.2 | | tn.3 | | tn.4 | | tn.5 | | tn.6 | | tn.7 | |
| |______| |______| |______| |______| |______| |______| |______| |
| ______ ______ ______ ______ ______ ______ ______ |
| |______| |______| |______| |______| |______| |______| |______| |
| _________________ __________________ _________________ |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | remote video | | | | local video | |
| | | | | | ______ | |
| | | | keypad | | | PIP || |
| | | | | | |______|| |
| |_________________| | | |_________________| |
| | | |
| | | |
| |__________________| |
|________________________________________________________________|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
you have multiple active instances of this grabber;
require v4l device names to start with '/dev/' - prevents some useless
attempt to open a file as a device.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.
(issue #12799)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines
ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number
(issue AST-80)
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008) | 8 lines
When using a Local channel, started by a call file, with a destination of an
AGI script, the AGI script does not always get notified of a hangup if the
underlying channel hangs up early.
(closes issue #11833)
Reported by: IgorG
Patches:
local_hangup-v1.diff uploaded by IgorG (license 20)
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r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008) | 7 lines
It's possible for a hangup to be received, even just after the initial cid
spill.
(closes issue #12453)
Reported by: Alex728
Patches:
20080604__bug12453.diff.txt uploaded by Corydon76 (license 14)
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They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
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them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
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r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines
Cisco BTS sends SIP responses with a tab between the Cseq number and
SIP request method in the Cseq: header. Asterisk did not handle this
properly, but with this patch, all is well.
(closes issue #12834)
Reported by: tobias_e
Patches:
12834.patch uploaded by putnopvut (license 60)
Tested by: tobias_e
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r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines
People expect that if "hasvoicemail" is set in users.conf, even if "mailbox"
isn't set, that SIP will detect a mailbox.
(closes issue #12855)
Reported by: PLL
Patches:
20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: PLL
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r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines
Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP.
(closes issue #12803)
Reported by: lanzaandrea
Patches:
chan_sip.c.diff uploaded by lanzaandrea (license 496)
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function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
Reported by: mostyn
Patches:
iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
(with some additional cleanup by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
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Jeff Peeler
Tilghman Lesher
Jason Parker
Sean Bright
Dwayne M. Hubbard
Michiel van Baak
Richard Mudgett
Mark Michelson
Kevin P. Fleming
Russell Bryant
Brett Bryant
Christian Richter
Olle Johansson
Matthew Fredrickson
Luigi Rizzo
Joshua Colp
Steve Murphy
Doug Bailey