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r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
sip codec negotiation of dynamic rtp payloads error fix
This patch fixes how chan_sip handles dynamic rtp payload types
it does not understand. At the moment if a dynamic payload's mime
type does not match one we understand, the payload does not get
removed from our payload table. As a result of this, the payload
is set to whatever dynamic codec we use internally for that payload
number on outgoing INVITES. This is incorrect.
This patch fixes this by properly checking the rtpmap set function's
return code to make sure it was found. The function can return both
-1 and -2 depending on the source of the mismatch. We were just
checking -1 explicitly.
Review: https://reviewboard.asterisk.org/r/1169/
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Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.
There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation. The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities. A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.
The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.
For example, you may have a single button that when not lit, there is no
active CCSS request. When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel(). If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful. The actual request could ultimately fail. Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.
The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary. The idea is to allow some level of
customization as to the phone's behavior.
As an example, you may want the BLF key to go solid once you have
requested a callback. You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback. You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.
Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine. You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.
You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states. For example, you
may have an extension 3000 that is currently associated with device
SIP/3000. You could then create a feature code for that extension that
may look something like:
exten => *823000,hint,ccss:sip/3000
You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.
(closes issue #18788)
Reported by: p_lindheimer
Patches:
ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski
Review: https://reviewboard.asterisk.org/r/1105/
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r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines
Merged revisions 313579 via svnmerge from
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r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines
Merged revisions 313545 via svnmerge from
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r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines
Asterisk does not hangup a channel after endpoint hangs up.
If the call that the dialplan started an AGI script for is hungup while
the AGI script is in the middle of a command then the AGI script is not
notified of the hangup. There are many AGI Exec commands that this can
happen with. The reported applications have been: Background, Wait, Read,
and Dial. Also the AGI Get Data command.
* Don't wait on the Asterisk channel after it has hung up. The channel is
likely to never need servicing again.
* Restored the AGI script's ability to return the AGI_RESULT_HANGUP value
in run_agi(). It previously only could return AGI_RESULT_SUCCESS or
AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged.
(closes issue #17954)
Reported by: mn3250
Patches:
issue17954_v1.8.patch uploaded by rmudgett (license 664)
issue17954_v1.6.2.patch uploaded by rmudgett (license 664)
issue17954_v1.4.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA SWP-2171
(closes issue #18492)
Reported by: devmod
Tested by: rmudgett
JIRA SWP-2761
(closes issue #18935)
Reported by: nvitaly
Tested by: astmiv, rmudgett
JIRA SWP-3216
(closes issue #17393)
Reported by: siby
Tested by: rmudgett
JIRA SWP-2727
Review: https://reviewboard.asterisk.org/r/1165/
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r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
The CallCompletionRequest()/CallCompletionCancel() dialplan applications
exit nonzero on normal failure conditions. The nonzero exit causes the
dialplan to hangup immediately. The dialplan author has no opportunity to
report success/failure to the user.
* Made always return zero so the dialplan can continue.
* Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also
documented the values set.
* Reduced the warning about no core instance in CallCompletionCancel() to
a debug message. It is a normal event and should not be output at the
WARNING level.
(closes issue #18763)
Reported by: p_lindheimer
Patches:
ccss.patch uploaded by p lindheimer (license 558) Modified
Tested by: p_lindheimer, rmudgett
JIRA SWP-3042
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In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s
ntax remains the same and the method used to track the pattern history will only change when using the length
4 patterns.
(closes issue SWP-3250)
Code:
jrose
rmudgett
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r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines
Merged revisions 310889 via svnmerge from
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r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines
Merged revisions 310888 via svnmerge from
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r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines
Don't delay DTMF in core bridge while listening for DTMF features
This patch is mostly the work of Olle Johansson. I did some cleanup and
added the silence generating code if transmit_silence is set.
When a channel listens for DTMF in the core bridge, the outbound DTMF is not
sent until we have received DTMF_END. For a long DTMF, this is a disaster. We
send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds.
Some products see this delay and the time skew on RTP packets that results and
start ignoring the audio that is sent afterward.
With this change, the DTMF_BEGIN frame is inspected and checked. If it matches
a feature code, we wait for DTMF_END and activate the feature as before. If
transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a
multi-digit feature. If it doesn't match a feature, the frame is forwarded
along with the DTMF_END without delay. By doing it this way, DTMF is not delayed.
(closes issue #15642)
Reported by: jasonshugart
Patches:
issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396)
Tested by: globalnetinc, jde
(closes issue #16625)
Reported by: sharvanek
Review: https://reviewboard.asterisk.org/r/1092/
Review: https://reviewboard.asterisk.org/r/1125/
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r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines
Merged revisions 310635 via svnmerge from
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r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines
Merged revisions 310633 via svnmerge from
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r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines
"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410
The last character in the caller id message is getting a framing error.
The checksum is the last character in the message. A framing error in the
checksum could be because:
1) The sender did not send a full stop bit.
2) The sender cut off the FSK carrier too soon.
3) The sender opted to send zero of the specified zero to 10 trailing mark
bits and round-off errors in the code resulted in the code not being where
it thought it was in the demodulated bit stream.
Bit 8 of 'b' is set when parity error.
Bit 9 of 'b' is set when framing error.
Made ignore the framing and parity error bits if the errored character is
the checksum. We can tolerate a framing/parity error there. The checksum
character validates the message.
(closes issue #18474)
Reported by: nivek
Patches:
callerid.c.1.patch uploaded by nivek (license 636) (with modifications)
Tested by: nivek
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r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines
remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call
If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
(where x = 0/1)
1). ZOMBIE
2). cx->tech_pvt != pvtx
3). gluex != ast_rtp_instance_get_glue(cx->tech->type))
(closes issue #18781)
Reported by: alecdavis
Patches:
bug18781.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, ZX81
Review: https://reviewboard.asterisk.org/r/1128/
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r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines
Add \r\n to remaining http headers passed to ast_http_send
r309204 changed the behavior of ast_http_send. It now requires headers
to be passed with trailing \r\n. This change updates the remaining
instances in the code that did not pass the \r\n.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
res_phoneprov.c.diff uploaded by lathama (license 1028)
manager.diff.txt uploaded by twilson (license 396)
Tested by: lathama
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r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines
Merged revisions 309251 via svnmerge from
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r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
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r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines
Merged revisions 309033-309034 via svnmerge from
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r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines
A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.
Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
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r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines
Clarify meaning, removing double negative (stupid!)
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r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Invalid read in ast_channel_set_caller_event().
Valgrind reported that ast_channel_set_caller_event() was reading data
from a freed buffer when using the pre_set structure.
Rearange things to pre-calculate the name and number pointer before
updating the caller party structure to see if the name or number was
changed.
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r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines
Merged revisions 308814 via svnmerge from
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r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines
Merged revisions 308813 via svnmerge from
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r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines
Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection every
time someone connects via AMI. It should only be sent to the user who
just connected.
(closes issue #18168)
Reported by: FeyFre
Patches:
bug0018168.patch uploaded by FeyFre (license 1142)
Tested by: FeyFre, twilson
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-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
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Guessed the log levels based on info that level 3
is the soft roof. Can we create a page / document
to define the levels?
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Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.
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r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines
No response sent for SIP CC subscribe/resubscribe request.
Asterisk does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only subscribe
for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its call-completion state
is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it
trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
Because this is an invalid state change, it just ignores the message. The
only state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state.
Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
the call by sending a CANCEL.
Asterisk should always send a response. Even if its a negative one.
The fix is to allow for the CCSS core to notify a CC agent that a failure
has occurred when CC is requested. The "ack" callback is replaced with a
"respond" callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that may need
to be communicated to the requester.
(closes issue #18336)
Reported by: GeorgeKonopacki
Tested by: mmichelson, rmudgett
JIRA SWP-2633
(closes issue #18337)
Reported by: GeorgeKonopacki
Tested by: mmichelson
JIRA SWP-2634
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r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines
Merged revisions 307535 via svnmerge from
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r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines
Merged revisions 307534 via svnmerge from
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r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines
Remove color when executing commands via a remote console.
Essentially this makes '-x' imply '-n' on rasterisk. This was done in a
different and incomplete way previously, which I'm reverting here.
(issue #18776)
Reported by: alecdavis
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The nativeformats field was being overwritten when it should have been
appended too. This caused some format capabilities to be lost briefly and
some log warnings to be output.
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r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines
Fix issue with verbose messages not showing on remote console.
This code was reworked recently, and since the logchannel list hadn't been
created yet at this point, and it was a verbose message, it was being dropped
on the floor. Now it'll continue on to where it should be handled.
(closes issue #18580)
Reported by: pabelanger
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r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines
Add a couple of useful channel variables for the CC recall macro.
CC_EXTEN and CC_CONTEXT will allow you to determine the channel
and context that will be called when the recall occurs.
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Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.
The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.
JIRA SWP-2845
JIRA ABE-2736
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r306674 | twilson | 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) | 24 lines
Merged revisions 306673 via svnmerge from
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r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines
Merged revisions 306672 via svnmerge from
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r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines
Don't try to pickup a call in the middle of a masquerade
If A calls B which doesn't answer and C & D both try to do a call pickup, it is
possible for ast_pickup_call to answer the call, then fail to masquerade one of
the calls because the other one is already in the process of masquerading. This
patch checks to see if the channel is in the process of masquerading before
call before selecting it for a pickup.
Review: https://reviewboard.asterisk.org/r/1094/
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r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines
Rearrange a bit of code in the generic CC recall operation.
By waiting to call the callback macro after the CC_INTERFACES,
extension, priority, and context have been set, this information
can be accessed more easily within the callback macro.
Reported by Philippe Lindheimer.
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The logic got reversed, oops. Works properly now when multiple blackfilters are
present.
(closes issue #18283)
Reported by: telecos82
Patches:
ast_managereventfilter.patch uploaded by telecos82 (license 687)
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David Vossel
Jonathan Rose
Richard Mudgett
Matthew Nicholson
Tilghman Lesher
Terry Wilson
Alec L Davis
Jason Parker
Andrew Latham
Tzafrir Cohen
Jeff Peeler
Mark Michelson
Paul Belanger