https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines
Don't do an SRV lookup if a port is specified
RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.
(closes issue #14419)
Reported by: klaus3000
Patches:
patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)
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r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines
check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()
The reporter didn't actually upload a properly-formed patch, instead a
modified chan_sip.c file was uploaded. I created a patch to determine the
changes, then modified the suggested changes to create a proper fix. The
summary above is a complete description of the changes.
(closes issue #13547)
Reported by: tecnoxarxa
Patches:
chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa
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r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines
Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
(closes issue #14350)
Reported by: fhackenberger
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r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines
Remove a debug message I put in by accident.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
supporting devices. The devices (snoms, specifically) need to receive a SIP
URI instead of just an extension. This adds that functionality.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Also, change a function in app.c to return a userful value instead of always returning 0.
Patch by fnordian, changed by Corydon76 and myself.
This does not close the bug report, as fnordian had an additional change we're still discussing.
(related to issue #14059)
Reported by: fnordian
Patches:
chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines
Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.
The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!
(closes issue #14294)
related to issue #13385
Reported by: klaus3000 and adomjan
Patches:
bug14294b.diff uploaded by oej (license 306)
Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Merged revisions 171527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines
Use the same branch tag in CANCEL as in INVITE
Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems.
Thanks Fredrik for pointing out where the bug in the SIP messaging was.
(closes issue #14346)
Reported by: oej
Patches:
bug14346.diff uploaded by oej (license 306)
Tested by: oej
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ccesario on IRC pointed out that his sip peers were not displayed
properly when he would issue the command "sip show peers." The problem
was that the onlymatchonip field was used to determine if the endpoint
was a "peer" or "user." The tricky part is that a "friend" is supposed
to be treated as both a "user" and a "peer" but the logic would not allow
"friends" to show up as "peers" since onlymatchonip was set to FALSE
for friends.
I have modified the sip_peer structure to more explicitly keep track of
what type endpoint it is so that the various manager and CLI commands
will display the expected information
Reported by ccesario via IRC
Tested by ccesario
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an ast_str expands to hold more data, any pointers that were pointing
to the data prior to the expansion will be pointing at invalid memory. This
change makes such pointers used in chan_sip.c instead be offsets from the
beginning of the string so that the same math may be applied no matter where
in memory the string resides.
To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added
to chan_sip.c so that given a sip_request and an offset, the string at that
offset is returned.
(closes issue #14220)
Reported by: riksta
Tested by: putnopvut
Review http://reviewboard.digium.com/r/126/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines
Account for possible NULL pointer when we receive a 408 in response to a REGISTER
It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash
This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER
(closes issue #14211)
Reported by: aborghi
Patches:
14211.diff uploaded by putnopvut (license 60)
Tested by: aborghi
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In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically
allocated buffer to hold the text of the request. There was a check in the
add_line function to not attempt to write the line into the buffer if we
did not have room for it.
In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str
structure is used to hold the text. Since it may grow to fit an arbitrarily
sized string, this check in add_line is no longer valid.
I found this oddity while attempting to fix issue #14220; however, I do not
believe that this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be printed had this
condition been satisfied.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan 2009) | 5 lines
I am reverting the fix made in revision 168128 (and its upward merges)
after being contacted by Olle Johansson and being shown how this fix is
incorrect. Thanks to Olle for clearing this up for me.
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r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines
Add check_via calls to more request handlers
INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests
were not checking the topmost Via to determine where
to send the response. Adding check_via calls to those
request handlers solves this.
(closes issue #13071)
Reported by: baron
Patches:
check_via.patch uploaded by baron (license 531)
Tested by: baron
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the functionality to app_queue of calculating
the average amount of time that channels are bridged for a
queue. The algorithm used to calculate the average is the same
exponential average currently used to calculate the average holdtime.
See the CHANGES file to see the methods you may use to view this
information.
(closes issue #13960)
Reported by: coolmig
Patches:
app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines
When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available.
http://reviewboard.digium.com/r/123/
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r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines
A couple of changes to T.38 SDP attribute handling
There are some boolean attributes for T.38 such
as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we
should treat these as a "true" value. The current
code, however, was requiring a 1 or 0 as the value
of the attribute in order to parse it. This is due
to the fact that there are some T.38 endpoints and
gateways that also transmit this information
incorrectly. This patch follows the "be liberal in
what you accept and strict in what you send"
philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending
information as it is supposed to be sent.
It was also discovered that a particular type of
T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram
and T38MaxBitRate, it used T38MaxDatagram and
T38FaxMaxRate respectively. We now will properly
accept these attributes as well.
Note that there are a lot of patches cited in
the below commit message template. This is
because the person who submitted these patches is
an awesome person and wrote 1.4, 1.6.0, and 1.6.1
variants.
(closes issue #13976)
Reported by: linulin
Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648)
Tested by: arcivanov
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Note that in the SIPS URI scheme, transport is independent of TLS,
and thus "sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
enabled (either globally or for a specific peer), chan_sip will treat any SDP
data it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session version
received is different from the current SDP session version. This option is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with Microsoft OCS
which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/
(closes issue #13958)
Reported by: toc
Tested by: toc
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r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines
After looking through SIP registration code most of the day, this
is one of the few things I could find that was just plain wrong.
Even though it probably isn't possible for it to happen, it seems weird
to have code that checks if a pointer is NULL and then immediately dereferences
that pointer if it was NULL.
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