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r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) | 10 lines
Don't hang up a call on an SRTP unprotect failure
Also make it more obvious when there is an issue en/decrypting.
(closes issue #17563)
Reported by: Alexcr
Patches:
res_srtp.c.patch uploaded by sfritsch (license 1089)
Tested by: twilson
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r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines
Merged revision 287014 from
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r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines
The handling of call transfer signaling for mISDN PTMP is not fully implemented.
The handling of call transfer signaling for mISDN PTMP is not fully
implemented. The signaling of number updates with ISDN/DSS1 ECT
supplementary services (ETS 300 369-1) comes along with a notification
indicator IE and redirection number IE for PTMP. The implementation in
the current Asterisk mISDN channel unfortunately can handle these
information elements only in a NOTIFY message. These information elements
are also signaled in a FACILTY message with a RequestSubaddress facility,
when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of
ETS 300 369-1).
**********
abe_2526_ast.patch
* Added support to handle the notification indicator IE and redirection
number IE with the RequestSubaddress facility.
* Made misdn_update_connected_line() send a NOTIFY message if Asterisk
originated the call and it is not connected yet.
* Made misdn_update_connected_line() send a FACILITY message if the call
is already connected.
This patch requires the presence of the associated mISDN patches to
compile. I had to enhance mISDN to allow the notification indicator IE
and the redirection number IE to be used with a FACILITY message. Earlier
versions of the Digium enhanced mISDN are no longer going to work.
**********
abe_2526_misdn.patch
* Made an incoming FACILITY message allow the presence of the notification
indicator IE and the redirection number IE.
**********
abe_2526_misdnuser_v3.patch
* Added support to send and receive a FACILITY message with the
notification indicator IE and the redirection number IE.
* Added the ability to send a NOTIFY message in PTMP/NT mode to all
responding subcalls in Q.931 states 6, 7, 8, 9, and 25.
**********
Patches:
abe_2526_ast.patch uploaded by rmudgett (license 664)
abe_2526_misdn.patch uploaded by rmudgett (license 664)
abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
Tested by: rmudgett and reporter
JIRA SWP-2146
JIRA ABE-2526
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r287015 | jpeeler | 2010-09-15 15:32:52 -0500 (Wed, 15 Sep 2010) | 21 lines
Merged revisions 286998 via svnmerge from
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r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
Merged revisions 286941 via svnmerge from
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r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
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r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
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r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) | 12 lines
Unable to originate calls using E&M over T1.
When originating a call from Unit Under Test to Reference Unit using E&M
RBS signaling mode, I get the following warning message: "Ring/Off-hook in
strange state 3 on channel 1".
Fixed the sig_analog outgoing flag. It was never set when sig_analog was
extracted from chan_dahdi.
JIRA SWP-2191
JIRA AST-408
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r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) | 1 line
Simplify some code in sig_analog.
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r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) | 16 lines
Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling.
This fixes a regression introduced in r274783.
(closes issue #17960)
Reported by: adriavidal
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mich, mnicholson, adriavidal
(closes issue #17676)
Reported by: outcast
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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r286682 | mnicholson | 2010-09-14 13:04:21 -0500 (Tue, 14 Sep 2010) | 21 lines
Merged revisions 286681 via svnmerge from
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r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines
Merged revisions 286679 via svnmerge from
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r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines
Only drop duplicate answer frames if the channel is bridged.
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework.
ABE-2473
(related to issue #2342)
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r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010) | 9 lines
Rate limit calls to fsync() to 1 per second after astdb updates.
Astdb was determined to be one of the most significant bottlenecks in SIP
registration processing. This patch improved the speed of an astdb load
test by 50000% (yes, Fifty-Thousand Percent). On this particular load test
setup, this doubled the number of SIP registrations the server could handle.
Review: https://reviewboard.asterisk.org/r/825/
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r286189 | twilson | 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) | 30 lines
Merged revisions 286115 via svnmerge from
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r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
Merged revisions 286059 via svnmerge from
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r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
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r286118 | rmudgett | 2010-09-10 15:55:37 -0500 (Fri, 10 Sep 2010) | 25 lines
Merged revisions 286116 via svnmerge from
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r286116 | rmudgett | 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) | 18 lines
Merged revisions 286113 via svnmerge from
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r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines
An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset,
the outgoing leg may not hang up or be delayed in hanging up. (Causes:
PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.)
Just hang up the call if the incoming call leg hangs up before connecting
for any reason. It makes no sense to send a BUSY or CONGESTION control
frame to the outgoing call leg under these circumstances.
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r285962 | tilghman | 2010-09-10 00:32:18 -0500 (Fri, 10 Sep 2010) | 13 lines
Merged revisions 285961 via svnmerge from
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r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines
Another fix for Mac OS X.
While trying to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch.
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r285931 | tilghman | 2010-09-09 20:25:50 -0500 (Thu, 09 Sep 2010) | 21 lines
Merged revisions 285930 via svnmerge from
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285889 via svnmerge from
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r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
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r285564 | dvossel | 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) | 60 lines
Merged revisions 285563 via svnmerge from
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r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines
Fixes interoperability problems with session timer behavior in Asterisk.
CHANGES:
1. Never put "timer" in "Require" header. This is not to our benefit
and RFC 4028 section 7.1 even warns against it. It is possible for one
endpoint to perform session-timer refreshes while the other endpoint does
not support them. If in this case the end point performing the refreshing
puts "timer" in the Require field during a refresh, the dialog will
likely get terminated by the other end.
2. Change the behavior of 'session-timer=accept' in sip.conf (which is
the default behavior of Asterisk with no session timer configuration
specified) to only run session-timers as result of an incoming INVITE
request if the INVITE contains an "Session-Expires" header... Asterisk is
currently treating having the "timer" option in the "Supported" header as
a request for session timers by the UAC. I do not agree with this. Session
timers should only be negotiated in "accept" mode when the incoming INVITE
supplies a "Session-Expires" header, otherwise RFC 4028 says we should
treat a request containing no "Session-Expires" header as a session with
no expiration.
Below I have outlined some situations and what Asterisk's behavior is.
The table reflects the behavior changes implemented by this patch.
SITUATIONS:
-Asterisk as UAS
1. Incoming INVITE: NO "Session-Expires"
2. Incoming INVITE: HAS "Session-Expires"
-Asterisk as UAC
3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header
4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header
5. Outgoing INVITE: HAS "Session-Expires".
Active - Asterisk will have an active refresh timer regardless if the other endpoint does.
Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does.
XXXXXXX - Not possible for mode.
______________________________________
|SITUATIONS | 'session-timer' MODES |
|___________|________________________|
| | originate | accept |
|-----------|------------|-----------|
|1. | Active | Inactive |
|2. | Active | Active |
|3. | XXXXXXXX | Active |
|4. | XXXXXXXX | Inactive |
|5. | Active | XXXXXXXX |
--------------------------------------
(closes issue #17005)
Reported by: alexrecarey
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r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines
Merged revisions 285532 via svnmerge from
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r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines
Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart.
(closes issue #17408)
Reported by: sysreq
Patches:
asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009)
Tested by: sysreq
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