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r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines
- Fixing issues with "sip show settings"
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
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r128290 | oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines
Adding doxygen comments to missing parts, moving some #define
...trying to get my head around the thoughts behind the TCP/TLS stuff
and figure out what needs to be done to make it useful...
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r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | 38 lines
Merged revisions 127663 via svnmerge from
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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
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r127779 | oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines
Revert some logic for session timers. We do send in-dialog requests that should not have session-timer
require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses
that are related to INVITEs and re-INVITEs.
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r127297 | tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12 lines
Change the global timer B to be dependent on the value of the T1 timer, as
recommended in RFC 3261, instead of being hardcoded to 32 seconds. This is
important for LANs, as it allows autocongestion to occur much more quickly, if
desired by the local PBX administrator. It also corrects a bug: if the T1
timer was increased beyond 500ms, then timer B would have been set at a much
lower value than recommended.
(closes issue #12544)
Reported by: kactus
Patches:
20080616__bug12544.diff.txt uploaded by Corydon76 (license 14)
Tested by: kactus
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r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) | 20 lines
The following patch with some changes for trunk...
Merged revisions 126516 via svnmerge from
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r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines
Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and
also fail if we don't get the very same precious ACK. Based on patch by tsearle, with
my own additions.
(closes issue #12951)
Reported by: tsearle
Patches:
busy_retransmit.patch uploaded by tsearle (license 373)
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r125891 | bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines
Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.
(issue #12799)
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r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 Jun 2008) | 5 lines
Updates all usages of ast_tcptls_session_instance to be managed by reference counts so that they only get destroyed when all threads are done using
them, and memory does not get free'd causing strange issues with SIP.
This code was originally written by russellb in the team/group/issue_11972/ branch.
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r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun 2008) | 19 lines
Merged revisions 123333 via svnmerge from
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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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r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008) | 16 lines
Merged revisions 123110 via svnmerge from
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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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r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) | 14 lines
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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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r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines
Merged revisions 120863,120885 via svnmerge from
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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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r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines
Merged revisions 118646 via svnmerge from
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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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r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008) | 20 lines
Merged revisions 118251 via svnmerge from
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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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r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008) | 13 lines
Merged revisions 115304 via svnmerge from
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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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r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines
Merged revisions 114632 via svnmerge from
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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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