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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines
Port "hasvoicemail" change from SIP to other channel drivers
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines
Don't send a BYE on a dialog that is already gone during a REFER.
(closes issue #12865)
Reported by: flefoll
Patches:
chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244)
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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
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been.
(closes issue #12706)
Reported by: falves11
Patches:
chan_sip.c.diff uploaded by rjain (license 226)
Tested by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4 lines
If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed.
(closes issue #12828)
Reported by: ramonpeek
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(Closes AST-38)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #12562)
Reported by: michael-fig
Patches:
20080515__bug12562.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line
add another LOW_MEMORY define I forgot
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line
only define thread storage variable if necessary for LOW_MEMORY
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
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r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
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to realtime less painful in the future.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines
Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines
Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP.
(closes issue #12501)
Reported by: slimey
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r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines
Realtime flag affects construction in multiple ways, so consulting whether
rtcachefriends was set was done too soon (needed to be done inside build_peer,
not just as a flag to build_peer).
Also, fullcontact needed to be reconstructed, because realtime separates the
embedded ';' into multiple fields.
(closes issue #12722)
Reported by: barthpbx
Patches:
20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
Tested by: barthpbx
(Much of the discussion happened on #asterisk-dev for diagnosing this issue)
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 lines
Apply the autoframing setting to dialogs that do not get matched against a user or peer.
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(closes issue #12678)
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bridging dropping audio nearly immediately. The problem was that the lock on the SIP dialog was not being unlocked while the bridge was still active. (Related to issue #12566)
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in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines
Accept text messages even with
Content-Type: text/plain;charset=Södermanländska
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This is to enable video and text in originated calls. Development sponsored
by Omnitor AB, Sweden. (http://www.omnitor.se)
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marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
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r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines
Don't give up on attempting an outbound registration if we receive a 408 Timeout.
(closes issue #12323)
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(closes issue #12597)
Reported by: hooi
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r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines
Avoid putting opaque="" in Digest authentication. This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)
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deprecated function)
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basis in the register line. This comes from a Switchvox patch. (issue AST-24)
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r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines
Don't crash on bad SIP replys.
Fix created in Huntsville together with Mark M (putnopvut)
(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej
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(closes issue #12524)
Reported by: ctooley
Patches:
sip_qualify_peer.diff.2 uploaded by ctooley (license 136)
some modifications for trunk by Corydon76
Tested by: Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11328)
Reported by: rain
Patches:
20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me
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(closes issue #12519)
Reported by: falves11
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r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines
Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.
(closes issue #12513)
Reported by: mneuhauser
Patches:
asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)
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r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines
Only have one max-forwards header in outbound REFERs.
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!
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(closes issue #12514)
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r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines
Add 502 support for both directions, not only one... (see r114571)
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r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines
Treat a 502 just like a 503, when it comes to processing a response code
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