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r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines
Race condition when ISDN CallRerouting/CallDeflection invoked.
The queued AST_CONTROL_BUSY could sometimes be processed before the
call_forward dial string is recognized.
* Moved setting the call_forwarding dial string after sending a response
to the initiator and just queue an empty frame to wake up the media thread
instead of an AST_CONTROL_BUSY.
* Added check for empty rerouting/deflection number and respond with an
error.
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r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) | 9 lines
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(closes issue #18946)
Reported by: GeorgeKonopacki
Patches:
18946.patch uploaded by mmichelson (license 60)
Tested by: GeorgeKonopacki
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r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | 9 lines
Returns with an error notice if CHANNEL function of SIP channel is read without arguments.
(Closes issue #18653)
Reported by: wuwu
Patches:
diff.patch uploaded by jrose (license 1225)
Tested by: jrose
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r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line
Make pri parameter description consistent.
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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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r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines
Fix caller id passed to openr2_chan_make_call
(closes issue #18894)
Reported by: malufrj
Tested by: moy
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Asterisk).
In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants.
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r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
Get real channel of a DAHDI call.
Starting with Asterisk v1.8, the DAHDI channel name format was changed for
ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
There were several reasons that the channel name had to change.
1) Call completion requires a device state for ISDN phones. The generic
device state uses the channel name.
2) Calls do not necessarily have B channels. Calls placed on hold by an
ISDN phone do not have B channels.
3) The B channel a call initially requests may not be the B channel the
call ultimately uses. Changes to the internal implementation of the
Asterisk master channel list caused deadlock problems for chan_dahdi if it
needed to change the channel name. Chan_dahdi no longer changes the
channel name.
4) DTMF attended transfers now work with ISDN phones because the channel
name is "dialable" like the chan_sip channel names.
For various reasons, some people need to know which B channel a DAHDI call
is using.
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
in use by the channel. Use CHANNEL(no_media_path) to determine if the
channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an
Asterisk channel so AMI applications can passively determine the B channel
currently in use. Calls with "no-media" as the DAHDIChannel do not have
an associated B channel. No-media calls are either on hold or
call-waiting.
(closes issue #17683)
Reported by: mrwho
Tested by: rmudgett
(closes issue #18603)
Reported by: arjankroon
Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
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r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines
Merged revisions 309255 via svnmerge from
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r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines
Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.
Since it's a duplicate, nothing is going to be done, so delme doesn't need to
be set at all. Strangely, when this was added, this was being set to 1 in 1.6,
and 0 in trunk.
(issue AST-439)
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r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines
Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal.
Looks like an unintended change when sig_analog.c was extracted from
chan_dahdi.c.
Removed useless conditional around needed code and fixed resulting
compiler warning.
(closes issue #18667)
Reported by: enegaard
Patches:
issue18667.patch uploaded by enegaard (license 1197)
Tested by: enegaard
JIRA SWP-2965
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r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines
Merged revisions 309083 via svnmerge from
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r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines
Fixes thread blocking issue in the sip TCP/TLS implementation.
(closes issue #18497)
Reported by: vois
Patches:
issues_18497.diff uploaded by dvossel (license 671)
Tested by: vois, rossbeer, kowalma, Freddi_Fonet
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r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines
Fix Deadlock with attended transfer of SIP call
Call path
sip_set_rtp_peer (locks chan then pvt)
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper (locks p->owner)
But by the time p->owner lock was attempted, seems as though chan and p->owner were different.
So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.
(closes issue #18837)
Reported by: alecdavis
Patches:
bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, Irontec, ZX81, cmaj
Review: [https://reviewboard.asterisk.org/r/1126/]
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r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines
Merged revisions 308678 via svnmerge from
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r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines
Use remotesecret to authenticate with a remote party
The remotesecret option was only being used for outbound registration
and not for placing calls. This patch uses remotesecret on outbound
calls if it is set, otherwise secret is still used.
Review: https://reviewboard.asterisk.org/r/1107/
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r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines
sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.
(closes issue #18874)
Reported by: cmaj
Patches:
patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830)
JIRA SWP-3172
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audio ConfBridge, and other stuff
-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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List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting. Calls on hold or call-waiting
are not associated with any B channel.
JIRA LIBPRI-27
JIRA SWP-2547
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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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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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r306979 | twilson | 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) | 16 lines
Merged revisions 306973 via svnmerge from
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r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines
Merged revisions 306972 via svnmerge from
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r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines
Fix comparison for REFER Replaces tags with pedantic=yes
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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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r306619 | twilson | 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) | 24 lines
Merged revisions 306618 via svnmerge from
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r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines
Merged revisions 306617 via svnmerge from
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r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines
Don't allow a REFER w/replaces to replace its own dialog
Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces
header that matches the dialog of the REFER. This would be a situation like A
calls B, A calls C, A transfers B to A, which is just silly. This patch makes
the transfer fail instead of making Asterisk freak out and forget to hang other
channels up.
Review: https://reviewboard.asterisk.org/r/1093/
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The display ie handling can be controlled independently in the send and
receive directions with the following options:
* Block display text data.
* Use display text in SETUP/CONNECT messages for name.
* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).
* Pass arbitrary display text during a call. Sent in INFORMATION
messages. Received from any message that the display text was not used as
a name.
If the display options are not set then the options default to legacy
behavior.
The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.
To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.
JIRA SWP-2688
JIRA ABE-2693
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(closes issue #18556)
Reported by: kkm
Review: https://reviewboard.asterisk.org/r/1071/
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r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) | 20 lines
Fix SIP deadlock involving state changes.
Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper)
has caused locking problems. Both of these functions lock the channel when
the channel argument is passed in!
In this case, the suspected problem (the backtrace makes it impossible to tell)
was the private being locked in sip_set_rtp_peer and then:
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper
(Traced to verify that the fix was only required in 1.8 and later.)
(closes issue #18491)
Reported by: cmaj
Patches:
chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830)
Tested by: cmaj
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r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines
Merged revisions 306126 via svnmerge from
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r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines
Merged revisions 306119 via svnmerge from
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r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines
Set hangup cause in local_hangup
When a call involves a local channel (like SIP -> Local -> SIP), the hangup
cause was not being set. This resulted in SIP channels sometimes getting a
503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+
this also can cause issues with CCSS that involve a local channel. This patch
sets the hangupcause for one side of the local channel to the other in
local_hangup for outbound calls.
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
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r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines
Merged revisions 305889 via svnmerge from
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r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines
Merged revisions 305888 via svnmerge from
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r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines
Minor AST_FRAME_TEXT related issues.
* Include the null terminator in the buffer length. When the frame is
queued it is copied. If the null terminator is not part of the frame
buffer length, the receiver could see garbage appended onto it.
* Add channel lock protection with ast_sendtext().
* Fixed AMI SendText action ast_sendtext() return value check.
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Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
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r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb 2011) | 7 lines
Reverse sense of an error test when reading from astdb.
(closes issue #18545)
Reported by: jcovert
Patches:
chan_iax2.c.patch uploaded by jcovert (license 551)
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r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines
Merged revisions 305342 via svnmerge from
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r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines
Merged revisions 305341 via svnmerge from
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r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines
Obtain the pri lock for PRI queue counters.
Need to obtain the pri lock when calling pri_dump_info_str() to avoid a
reentrancy problem when calculating the Q.921 Q count statistic.
JIRA AST-484
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r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines
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r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines
Merged revisions 305252 via svnmerge from
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r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines
Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))
chan_iax2 and other channel drivers already had code to prevent this. The
attempt that app_dial was making to prevent it was not correct, so I fixed that.
(closes issue #18371)
Reported by: gbour
Patches:
18371.patch uploaded by gbour (license 1162)
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r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines
Add connected line chan_dahdi.conf pricpndialplan option.
* Added from_channel value to prilocaldialplan option.
JIRA ABE-2731
JIRA SWP-2842
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r304245 | mnicholson | 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) | 20 lines
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r304244 | mnicholson | 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) | 13 lines
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r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines
This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header.
ABE-2664
Review: https://reviewboard.asterisk.org/r/1059/
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r304150 | rmudgett | 2011-01-26 13:39:35 -0600 (Wed, 26 Jan 2011) | 16 lines
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r304149 | rmudgett | 2011-01-26 13:38:38 -0600 (Wed, 26 Jan 2011) | 9 lines
Merged revisions 304148 from
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r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines
Update documentation for DAHDISendCallreroutingFacility() application.
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r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines
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r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines
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r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines
Guard against retransmitting BYEs indefinitely
In the case of an attended transfer (A calls B, A atxfers to C) where
A becomes unreachable before replying to Asterisk's BYE, Asterisk can
sometimes retransmit the BYE indefinitely. This is because
__sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out,
it will not ever be marked as ALREADYGONE, so when __sip_autodestruct
is called again, we end up starting the cycle over.
This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt
in the case of a BYE that has timed out. This should prevent Asterisk
from trying to transmit new BYE messages in the future.
Review: https://reviewboard.asterisk.org/r/1077/
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r303860 | tilghman | 2011-01-25 12:55:27 -0600 (Tue, 25 Jan 2011) | 12 lines
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r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) | 5 lines
Fix "sip show user <tab>", so that it actually shows results, instead of just completing the last entry.
(closes issue #16675)
Reported by: pj
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r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines
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r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines
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r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines
Sending out unnecessary PROCEEDING messages breaks overlap dialing.
Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing
through Asterisk. There is not enough information available at this point
to know if dialing is complete. The ast_exists_extension(),
ast_matchmore_extension(), and ast_canmatch_extension() calls are not
adequate to detect a dial through extension pattern of "_9!".
Workaround is to use the dialplan Proceeding() application early in
non-dial through extensions.
* Effectively revert issue #16789.
* Allow outgoing overlap dialing to hear dialtone and other early media.
A PROGRESS "inband-information is now available" message is now sent after
the SETUP_ACKNOWLEDGE message for non-digital calls. An
AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE
messages for non-digital calls.
* Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was
inconsistent with the cause codes.
* Added better protection from sending out of sequence messages by
combining several flags into a single enum value representing call
progress level.
* Added diagnostic messages for deferred overlap digits handling corner
cases.
(closes issue #17085)
Reported by: shawkris
(closes issue #18509)
Reported by: wimpy
Patches:
issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664)
Expanded upon issue18509_early_media_v1.8_v3.patch to include analog
and SS7 because of backporting requirements.
Tested by: wimpy, rmudgett
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For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
escaped.
This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.
The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.
The unit tests for these functions have also been updated.
ABE-2705
Review: https://reviewboard.asterisk.org/r/1081/
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r303467 | qwell | 2011-01-24 11:20:03 -0600 (Mon, 24 Jan 2011) | 22 lines
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
Merged revisions 303284 via svnmerge from
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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r303286 | qwell | 2011-01-21 15:50:11 -0600 (Fri, 21 Jan 2011) | 22 lines
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) | 7 lines
Initialize an uninitialized variable.
(closes issue #18640)
Reported by: jcovert
Patches:
chan_sip.c.patch uploaded by jcovert (license 551)
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r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines
Use appropriate type for requested format in chan_local.
We were passing and storing the requested format as an int instead of format_t
resulting in truncation.
(closes issue #18238)
Reported by: whizemen
Patches:
0018238_speex16.patch uploaded by whizemen (license 1143)
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r302314 | mnicholson | 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) | 18 lines
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r302313 | mnicholson | 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) | 11 lines
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r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines
URI encode the user part of the contact header.
ABE-2705
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r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines
Only offer codecs both sides support for directmedia
When using directmedia, Asterisk needs to limit the codecs offered to just
the ones that both sides recognize, otherwise they may end up sending audio
that the other side doesn't understand.
(closes issue #17403)
Reported by: one47
Patches:
sip_codecs_simplified4 uploaded by one47 (license 23)
Tested by: one47, falves11
Review: https://reviewboard.asterisk.org/r/967/
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