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r291507 | rmudgett | 2010-10-13 14:01:48 -0500 (Wed, 13 Oct 2010) | 18 lines
Merged revision 291504 from
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r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, 13 Oct 2010) | 11 lines
Hold off ast_hangup() from destroying the ast_channel.
Must get the ast_channel lock before proceeding with release_chan() and
release_chan_early() to hold off ast_hangup() from destroying the
ast_channel.
Missed this change for -r291468.
JIRA ABE-2598
JIRA SWP-2317
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r291469 | rmudgett | 2010-10-13 13:10:21 -0500 (Wed, 13 Oct 2010) | 23 lines
Merge revision 291468 from
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r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed, 13 Oct 2010) | 16 lines
Memory overwrites when releasing mISDN call.
Phone <--> Asterisk
<-- ALERTING
--> DISCONNECT
<-- RELEASE
--> RELEASE_COMPLETE
* Add lock protection around channel list for find/add/delete operations.
* Protect misdn_hangup() from release_chan() and vise versa using the
release_lock.
JIRA ABE-2598
JIRA SWP-2317
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r291394 | russell | 2010-10-13 10:46:39 -0500 (Wed, 13 Oct 2010) | 20 lines
Merged revisions 291393 via svnmerge from
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r291393 | russell | 2010-10-13 10:29:21 -0500 (Wed, 13 Oct 2010) | 13 lines
Merged revisions 291392 via svnmerge from
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r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines
Lock pvt so pvt->owner can't disappear when queueing up a frame.
This fixes a crash due to a hangup race condition.
ABE-2601
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r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines
Gtalk enhancements and general code cleanup.
This patch includes several chan_gtalk enhancements.
Two new gtalk.conf options have been added, externip
and stunadd. Setting externip allows us to
manually specify what the external IP address is
outside of a NAT environment. Setting the stunaddr
option to a valid stun server allows for that external
ip to be retrieved via a STUN server automatically. This
external IP is then advertised during call setup as
a possible candidate.
I have also attempted to clean up chan_gtalk's code
so it meets our coding guidelines. During this cleanup
I noticed several things that need to be done in the
code and made a TODO section at the top of the file.
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r291112 | rmudgett | 2010-10-11 13:48:15 -0500 (Mon, 11 Oct 2010) | 20 lines
Merged revisions 291110-291111 via svnmerge from
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r291110 | rmudgett | 2010-10-11 13:34:22 -0500 (Mon, 11 Oct 2010) | 9 lines
Merged revisions 291109 via svnmerge from
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r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line
Add missing unlock to an exception condition in reload_config().
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r291111 | rmudgett | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line
Make exit from handle_request_do() consistent.
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r291113 | rmudgett | 2010-10-11 13:51:13 -0500 (Mon, 11 Oct 2010) | 1 line
Move declaration closer to where now used.
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r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08 Oct 2010) | 12 lines
Make outbound Google Voice calls.
This patch allows for outbound Google Voice calls to be
dialed from Asterisk using chan_gtalk. Below is an example
dialstring.
exten -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,)
In this example, 'asterisk' is the jabber.conf profile configured
to connect to your gmail account. In order to receive Google Voice
calls make sure to enable 'allowguest=yes' in gtalk.conf.
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r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07 Oct 2010) | 6 lines
Add Philippe Sultan to chan_gtalk author list.
Philippe has made some notable contributions to the
gtalk channel driver. His name deserves to be listed
amoung the authors of that file. Thanks Philippe!
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r290828 | dvossel | 2010-10-07 16:44:58 -0500 (Thu, 07 Oct 2010) | 5 lines
Outbound gtalk calls now work correctly.
There was a problem with how the candidates were being
built on an outbound call. This patch fixes that.
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r290674 | dvossel | 2010-10-06 16:22:51 -0500 (Wed, 06 Oct 2010) | 4 lines
Fixes commented out code to use #if 0 instead.
Thanks to rmudgett for catching this!
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r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 Oct 2010) | 12 lines
Fixes gtalk outbound DTMF to work properly.
Outbound DTMF with gtalk needs to be done within the RTP stream. I discovered
this after investigating a packet capture from the gmail client. Instead of
performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive
on the RTP stream using RFC2833 way of doing things. Chan_gtalk also had an issue
with negotiating RTP payload type 106 for the telephony-event and then sending
DTMF as payload 101. This has been resolved by always negotiating 101 as the payload
type like we do everywhere else. With this patch, incoming google voice calls forwarded
to Asterisk via gtalk work.
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r290506 | dvossel | 2010-10-05 17:23:00 -0500 (Tue, 05 Oct 2010) | 2 lines
Fixes uninitialized memory problem in 'iax2 set debug peer' option.
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r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) | 6 lines
Fixes chan_gtalk to work with gmail client
This patch was written by Philippe Sultan (phsultan). Thanks
for keeping this up to date!
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r290378 | dvossel | 2010-10-05 15:09:06 -0500 (Tue, 05 Oct 2010) | 11 lines
Resolves dnsmgr memory corruption in chan_iax2.
(closes issue #17902)
Reported by: afried
Patches:
issue_17902.rev1.txt uploaded by russell (license 2)
Tested by: afried, russell, dvossel
Review: https://reviewboard.asterisk.org/r/965/
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r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines
Merged revisions 289798 via svnmerge from
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r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289797 via svnmerge from
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r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
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r289701 | jpeeler | 2010-10-01 11:22:19 -0500 (Fri, 01 Oct 2010) | 28 lines
Merged revisions 289700 via svnmerge from
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r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines
Merged revisions 289699 via svnmerge from
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r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines
Ensure user portion of SIP URI matches dialplan when using encoded characters.
This commit takes a simliar approach to 288112 and checks the dialplan to
determine the proper action for an incoming contact header as to whether or not
it should be decoded or not. sip_new was blindly always decoding the extension,
which also caused the outgoing contact header to be incorrect as well as failing
to match the encoded extension in the dialplan.
(closes issue #17892)
Reported by: wdoekes
Patches:
bug17892-1.patch uploaded by jpeeler (license 325)
Tested by: wdoekes
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On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed.
Review: https://reviewboard.asterisk.org/r/901/
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r289554 | mnicholson | 2010-09-30 14:53:10 -0500 (Thu, 30 Sep 2010) | 11 lines
Merged revisions 289553 via svnmerge from
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r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) | 4 lines
Properly handle channel allocation failures duing invites with replaces.
ABE-2588
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r289549 | rmudgett | 2010-09-30 14:28:36 -0500 (Thu, 30 Sep 2010) | 17 lines
Merged revision 289547 from
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r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu, 30 Sep 2010) | 10 lines
In chan_misdn, the DivertingLegInformation2 DivertingNr is garbage when the number is restricted.
The same thing happens with DivertingLegInformation1 DivertedTo number.
The misdn_PresentedNumberUnscreened_extract() extracted the Unscreened
PartyNumber field unconditionally. It now checks the presented number
unscreened type to see if the PartyNumber was even present.
JIRA ABE-2595
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r289057 | rmudgett | 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010) | 5 lines
Avoid deadlock processing incoming AOC-E messages.
Deadlock avoidance for the owner channel was not done when processing
incoming AOC-E messages.
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r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) | 1 line
Break up long ast_manager_event_multichan() event lines.
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r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27 Sep 2010) | 1 line
Revert stuff not ready for commit in -r289054.
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r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) | 5 lines
Still build SIP, even if res_crypto cannot be built (use, not depend).
(closes issue #18062)
Reported by: a user on the mailing list
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r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) | 5 lines
Append Retry-After header on 500 error response to Re-INVITE according to RFC3261 section 14.2.
ABE-2301
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r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) | 4 lines
Inspect Require header on BYE transaction according to RFC3261 section 8.2.2.3.
ABE-2293
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r288748 | twilson | 2010-09-24 09:02:27 -0700 (Fri, 24 Sep 2010) | 19 lines
Merged revisions 288747 via svnmerge from
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r288747 | twilson | 2010-09-24 08:37:39 -0700 (Fri, 24 Sep 2010) | 12 lines
Merged revisions 288746 via svnmerge from
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r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) | 5 lines
Don't fail a masquerade if it is already being hung up
This avoids noise on some Local channel situations where we don't use /n.
Thanks to Alec Davis for the suggestion.
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r288507 | twilson | 2010-09-22 16:18:27 -0700 (Wed, 22 Sep 2010) | 22 lines
Merged revisions 288500 via svnmerge from
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r288500 | twilson | 2010-09-22 16:10:09 -0700 (Wed, 22 Sep 2010) | 15 lines
Merged revisions 288499 via svnmerge from
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r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) | 8 lines
Don't let a Local channel get bridged to itself
If a local channel gets bridged to itself, it becomes orphaned with no devices
left to actually tell it to hang up. This patch modifies local_fixup() to detect
this case and deny it.
Review: https://reviewboard.asterisk.org/r/934
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r288418 | dvossel | 2010-09-22 12:49:56 -0500 (Wed, 22 Sep 2010) | 18 lines
Merged revisions 288417 via svnmerge from
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r288417 | dvossel | 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) | 11 lines
Merged revisions 288416 via svnmerge from
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r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines
RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response.
ABE-2458
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r288345 | dvossel | 2010-09-22 11:59:14 -0500 (Wed, 22 Sep 2010) | 16 lines
Merged revisions 288344 via svnmerge from
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r288344 | dvossel | 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) | 9 lines
Merged revisions 288343 via svnmerge from
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r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines
During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup.
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r288194 | rmudgett | 2010-09-21 19:06:21 -0500 (Tue, 21 Sep 2010) | 40 lines
Merged revisions 288193 via svnmerge from
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r288193 | rmudgett | 2010-09-21 19:03:37 -0500 (Tue, 21 Sep 2010) | 33 lines
Merged revisions 288192 via svnmerge from
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r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) | 26 lines
In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel.
Near the beginning of schedule_delivery(), ast_bridged_channel() is called
on iaxs[fr->callno]->owner. However, the channel is not locked, which can
result in ast_bridged_channel() crashing should owner->tech change to a
technology that doesn't implement bridged_channel.
I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since
the owner lock was not held there either.
Converted the existing channel deadlock avoidance to use
iax2_lock_owner(). Using the new function simplified some awkward code.
In the process of fixing the locking on ast_bridged_channel(), I also
found a memory leak in socket_process() for v1.6.2 and v1.8. The local
struct variable ies.vars is not freed on early/abnormal function exits.
(closes issue #17919)
Reported by: rain
Patches:
issue17919_v1.4.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/926/
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r288159 | tilghman | 2010-09-21 17:57:22 -0500 (Tue, 21 Sep 2010) | 29 lines
Merged revisions 288113 via svnmerge from
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r288113 | tilghman | 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) | 22 lines
Merged revisions 288112 via svnmerge from
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r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines
Try both the encoded and unencoded subscription URI for a match in hints.
When a phone sends an encoded URI for a subscription, the URI is not matched
with the actual hint that is in decoded format. For example, if we have an
extension with a hint that is named: "#5601" or "*5601", the subscription will
work fine if the phone subscribes with an already decoded URI, but when it's
decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the
correct hint.
(closes issue #17785)
Reported by: ramonpeek
Patches:
20100831__issue17785.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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r288157 | pabelanger | 2010-09-21 18:26:15 -0400 (Tue, 21 Sep 2010) | 15 lines
Merged revisions 288147 via svnmerge from
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r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue, 21 Sep 2010) | 9 lines
Setup timer before set_config().
(closes issue #18019)
Reported by: Netview
Patches:
issue_0018019.patch uploaded by pabelanger (license 224)
Tested by: Netview
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needdestroy and rtptimeout
adding two dialog container, one for dialogs which need destroy, another for rtptimeout checks.
both container will be checked on every loop of do_monitor instead of iterate through all dialogs.
(closes issue #17912)
Reported by: schmidts
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/917/
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r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) | 4 lines
Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header.
ABE-2258
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r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) | 10 lines
Don't use ast_strdupa() from within the arguments to a function.
(closes issue #17902)
Reported by: afried
Patches:
issue_17902.rev1.txt uploaded by russell (license 2)
Tested by: russell
Review: https://reviewboard.asterisk.org/r/927/
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r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) | 9 lines
Anonymous callerid needs a "sip:" uri prefix.
(closes issue #17981)
Reported by: avalentin
Patches:
sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
(plus an additional fix by me)
Tested by: avalentin
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r287683 | rmudgett | 2010-09-20 18:14:42 -0500 (Mon, 20 Sep 2010) | 9 lines
The inalarm flag was not set in sig_analog struct if the port is initially in alarm.
Fixed initial inalarm value for sig_analog ports.
Along with -r261007, this gets the inalarm flag in sync with chan_dahdi
for sig_analog ports.
(closes issue #16983)
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r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) | 9 lines
Fixes issue with registrations not working properly with pedantic=yes.
(closes issue #18017)
Reported by: schmidts
Patches:
issues_18017_v1.diff uploaded by dvossel (license 671)
Tested by: schmidts
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r287643 | qwell | 2010-09-20 16:29:46 -0500 (Mon, 20 Sep 2010) | 15 lines
Merged revisions 287642 via svnmerge from
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r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) | 8 lines
Don't crash when parking a non-bridged call.
(closes issue #17680)
Reported by: jmhunter
Patches:
chan_skinny-park-v1.txt uploaded by DEA (license 3)
Tested by: jmhunter, DEA
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SIP messages. Adding error based on RFC 3398 recommendations.
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r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines
Merged revision 287014 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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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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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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r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) | 2 lines
Sets subscribed type for outgoing MWI subscriptions so correct Event header is used.
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r286758 | mnicholson | 2010-09-14 14:28:38 -0500 (Tue, 14 Sep 2010) | 27 lines
Merged revisions 286757 via svnmerge from
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r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines
Merged revisions 286756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines
Don't clear the username from a realtime database when a registration expires.
Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either.
(closes issue #17551)
Reported by: ricardolandim
Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96)
Tested by: ricardolandim, mnicholson
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r286457 | qwell | 2010-09-13 14:40:05 -0500 (Mon, 13 Sep 2010) | 12 lines
Merged revisions 286456 via svnmerge from
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r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines
Remove "Internal IP" from sip show settings, as it's not at all useful to display.
(closes issue #17840)
Reported by: oej
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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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