Commit Graph

2428 Commits

Author SHA1 Message Date
Olle Johansson
3c4e4a8355 Merged revisions 172234 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r172234 | oej | 2009-01-29 12:19:29 +0100 (Tor, 29 Jan 2009) | 7 lines

Make sure register= line supports both port and expiry at the same time.
(closes issue #14185)
Reported by: Nick_Lewis
Patches: 
      chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@172235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 11:24:30 +00:00
Olle Johansson
815b9b5691 Merged revisions 172173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines

Merged revisions 172169 via svnmerge from 
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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2009-01-29 10:36:28 +00:00
Olle Johansson
a68cd825fe Merged revisions 171528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 lines

Solving the same issue, but a bit different in trunk...

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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2009-01-27 15:19:52 +00:00
Olle Johansson
7481ad589a Merged revisions 171326 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | 17 lines

Merged revisions 171264 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines

Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes

(closes issue #14284)
Reported by: klaus3000
Patches: 
      patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000


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2009-01-26 14:58:04 +00:00
Joshua Colp
85f8608a8e Merged revisions 170505 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170504 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines
    
    Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
    (closes issue #14295)
    Reported by: klaus3000
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2009-01-23 18:11:35 +00:00
Mark Michelson
b7799bc31a Merged revisions 169791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r169791 | mmichelson | 2009-01-21 15:53:55 -0600 (Wed, 21 Jan 2009) | 18 lines

Further fix some oddities in sip show users and sip show peers logic

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


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2009-01-21 21:57:33 +00:00
Mark Michelson
3a69dbe8cc Merged revisions 169557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r169557 | mmichelson | 2009-01-20 14:10:31 -0600 (Tue, 20 Jan 2009) | 19 lines

Convert the character pointers in a sip_request to be pointer offsets

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/


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2009-01-20 20:15:50 +00:00
Terry Wilson
eb427ada5e Merged revisions 169044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r169044 | twilson | 2009-01-16 18:03:39 -0600 (Fri, 16 Jan 2009) | 8 lines
  
  Fix port :0 added to SIP INVITE URI when outboundproxy used
  
  (closes issue #14233)
  Reported by: chris-mac
  Patches: 
        asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176)
  Tested by: jamesgolovich, chris-mac, otherwiseguy
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2009-01-17 00:32:56 +00:00
Mark Michelson
382e0d37be Merged revisions 168976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168976 | mmichelson | 2009-01-16 16:43:09 -0600 (Fri, 16 Jan 2009) | 26 lines

Merged revisions 168975 via svnmerge from 
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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2009-01-16 22:46:33 +00:00
Mark Michelson
31749d07fa Merged revisions 168728 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168728 | mmichelson | 2009-01-15 13:16:29 -0600 (Thu, 15 Jan 2009) | 3 lines

Fix the compactheaders option in sip.conf


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2009-01-15 19:17:43 +00:00
Mark Michelson
33ad5ed729 Merged revisions 168725 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168725 | mmichelson | 2009-01-15 13:00:06 -0600 (Thu, 15 Jan 2009) | 17 lines

Remove an unneeded condition for line addition to a SIP request/response

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.


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2009-01-15 19:00:52 +00:00
Olle Johansson
ea715d8f7f Merged revisions 168712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168712 | oej | 2009-01-15 19:08:59 +0100 (Tor, 15 Jan 2009) | 3 lines

Make sure that we have the same terminology in sip.conf.sample and the source code warning.
Thanks Nick Lewis for pointing this out in the bug tracker.

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2009-01-15 18:18:34 +00:00
Mark Michelson
e31bbaea8b Merged revisions 168610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168610 | mmichelson | 2009-01-14 14:13:48 -0600 (Wed, 14 Jan 2009) | 9 lines

Restore the "sip show users" and "sip show user" CLI commands

(closes issue #14180)
Reported by: amorsen
Patches:
      sip_show_users_161v3.diff uploaded by putnopvut (license 60)
Tested by: blitzrage, amorsen


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@168611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-14 20:18:55 +00:00
Terry Wilson
af9322a5bf Merged revisions 168578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168578 | twilson | 2009-01-13 16:22:34 -0600 (Tue, 13 Jan 2009) | 14 lines
  
  Merged revisions 168551 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) | 7 lines
    
    Don't pass a value with a side effect to a macro
    
    (closes issue #14176)
    Reported by: paraeco
    Patches: 
          chan_sip.c.diff uploaded by paraeco (license 658)
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2009-01-13 22:38:21 +00:00
Mark Michelson
45340d2526 Merged revisions 168486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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2009-01-12 15:05:43 +00:00
Tilghman Lesher
02dd0d784f Merged revisions 168334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r168334 | tilghman | 2009-01-09 19:42:45 -0600 (Fri, 09 Jan 2009) | 2 lines
  
  sizeof for a stringfield is 4.  Kinda low for reconstructing a field value.
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2009-01-10 01:44:33 +00:00
Mark Michelson
4736407e8d Merged revisions 168193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan 2009) | 21 lines

Merged revisions 168128 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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2009-01-09 21:57:46 +00:00
Kevin P. Fleming
b205a18a8d Merged revisions 167720 via svnmerge from
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  r167720 | kpfleming | 2009-01-08 11:26:03 -0600 (Thu, 08 Jan 2009) | 9 lines
  
  Merged revisions 167714 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan 2009) | 1 line
    
    remove an unnecessary argument to queue_request()
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2009-01-08 17:28:42 +00:00
Kevin P. Fleming
4f2b0aa24c Merged revisions 167700 via svnmerge from
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  r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan 2009) | 12 lines
  
  Merged revisions 167620 via svnmerge from 
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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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Mark Michelson
0e408f80fe Merged revisions 167180 via svnmerge from
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r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan 2009) | 49 lines

Merged revisions 167179 via svnmerge from 
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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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2009-01-05 17:10:08 +00:00
Joshua Colp
75d4f7e1c2 Per kpfleming add a note describing why you must never change the first element of peer_finding_info.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-29 14:52:22 +00:00
Tilghman Lesher
827f655dac Merged revisions 166696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r166696 | tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines
  
  Allow semicolons and extended characters in user-specified SIP headers.
  (closes issue #14110)
   Reported by: gork
   Patches: 
         20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14)
   Tested by: gork, putnopvut
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2008-12-23 20:56:10 +00:00
Joshua Colp
4142e300f7 Merged revisions 165792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165792 | file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines
  
  Numerous documentation updates.
  (closes issue #13970)
  Reported by: pkempgen
  Patches:
        __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10)
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2008-12-18 21:24:00 +00:00
Joshua Colp
1f96800e79 Merge in changes to return chan_sip to matching based on how it was previously done and how it is done in trunk. It will do name based for users and friends and IP based for peers.
(closes issue #14107)
Reported by: jsmith


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2008-12-18 17:59:37 +00:00
Joshua Colp
f9973f81a3 Merged revisions 165216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines
  
  Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly!
  (closes issue #14055)
  Reported by: chris-mac
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Mark Michelson
cbad8e6dca Merged revisions 164978 via svnmerge from
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r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec 2008) | 15 lines

Merged revisions 164977 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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2008-12-16 23:07:33 +00:00
Russell Bryant
072033cdb0 Merged revisions 164675 via svnmerge from
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r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines

Merged revisions 164672 via svnmerge from 
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r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines

Fix a memory leak related to the use of the "setvar" configuration option.

The problem was that these variables were being appended to the list of vars
on the sip_pvt every time a re-registration or re-subscription came in.
Since it's just a waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying the vars.

(closes issue #14037)
Reported by: marvinek
Tested by: russell

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Joshua Colp
d2b0a64b61 Merged revisions 164659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r164659 | file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines
  
  When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself.
  (closes issue #13634)
  Reported by: performer
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2008-12-16 15:47:10 +00:00
Joshua Colp
d303ab8bee Merged revisions 164351 via svnmerge from
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  r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) | 13 lines
  
  Merged revisions 164350 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines
    
    Do not try to unlock a non-existant channel if the transfer fails.
    (closes issue #13800)
    Reported by: dwagner
    Patches:
          asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608)
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2008-12-15 18:14:04 +00:00
Russell Bryant
7129aba1a5 Merged revisions 163670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines

Rename a number of tcptls_session variables.  There are no functional changes here.

The name "ser" was used in a lot of places.  However, it is a relic from when
the struct was a server_instance, not a session_instance.  It was renamed since
it represents both a server or client connection.

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2008-12-12 18:53:37 +00:00
Russell Bryant
bb31aab74c Merged revisions 163667 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r163667 | russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines

Fix a small race condition in sip_tcp_locate().

We must increase the reference count on the tcptls_session _before_ unlocking
the thread list.

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2008-12-12 18:37:08 +00:00
Russell Bryant
3b2988893a Merged revisions 163642 via svnmerge from
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r163642 | russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines

Resolve crashes when using SIP TCP/TLS with qualify.

The problem was a reference count error on the tcptls_session structure.

(closes issue #13989)
Reported by: Nugget

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2008-12-12 18:22:47 +00:00
Joshua Colp
e4d276a957 Merged revisions 163629 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r163629 | file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines
  
  When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found.
  (closes issue #13811)
  Reported by: pj
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-12 18:19:33 +00:00
Joshua Colp
79d555f70c Merged revisions 163579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines
  
  Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven.
  (closes issue #13525)
  Reported by: pj
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-12 16:57:04 +00:00
Joshua Colp
6bb78bff53 Merged revisions 162997 via svnmerge from
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  r162997 | file | 2008-12-11 11:05:49 -0400 (Thu, 11 Dec 2008) | 4 lines
  
  When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it.
  (closes issue #13525)
  Reported by: pj
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2008-12-11 15:07:55 +00:00
Joshua Colp
25ed056a69 Merged revisions 162805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) | 13 lines
  
  Merged revisions 162804 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines
    
    Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
    (closes issue #12560)
    Reported by: vsauer
    Patches:
          patch001.diff uploaded by ramonpeek (license 266)
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2008-12-10 19:05:02 +00:00
Joshua Colp
777d07ee11 Merged revisions 162739 via svnmerge from
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  r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) | 13 lines
  
  Merged revisions 162738 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines
    
    When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
    (closes issue #13599)
    Reported by: hjourdain
    Patches:
          chan_sip.c.diff uploaded by hjourdain (license 583)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-10 17:54:58 +00:00
Mark Michelson
e67941f5cb Merged revisions 162664 via svnmerge from
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r162664 | mmichelson | 2008-12-10 10:34:35 -0600 (Wed, 10 Dec 2008) | 19 lines

Merged revisions 162663 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines

Revert fix for issue 13570. It has caused more problems than
it helped to fix.

(closes issue #13783)
Reported by: navkumar


(closes issue #14025)
Reported by: ffs


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2008-12-10 16:37:14 +00:00
Joshua Colp
a78825a181 Merged revisions 162619 via svnmerge from
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  r162619 | file | 2008-12-10 11:22:26 -0400 (Wed, 10 Dec 2008) | 4 lines
  
  When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server.
  (closes issue #13633)
  Reported by: performer
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2008-12-10 15:28:02 +00:00
Joshua Colp
f99abb26d6 Merged revisions 161726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r161726 | file | 2008-12-08 13:53:32 -0400 (Mon, 08 Dec 2008) | 13 lines
  
  Merged revisions 161725 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines
    
    Make the usereqphone option work again.
    (closes issue #13474)
    Reported by: mmaguire
    Patches:
          20080912_bug13474.diff uploaded by mmaguire (license 571)
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2008-12-08 17:55:45 +00:00
Matthew Nicholson
cebc3198bc Merged revisions 161721 via svnmerge from
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  r161721 | mnicholson | 2008-12-08 11:23:41 -0600 (Mon, 08 Dec 2008) | 7 lines
  
  Fix a crash that can occur on a transfer in chan_sip when attempting to collect
  rtp stats.
  
  (closes issue #13956)
  Reported by: chris-mac
  Tested by: chris-mac
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2008-12-08 17:24:32 +00:00
Tilghman Lesher
af7f07ac17 Merged revisions 160481 via svnmerge from
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  r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines
  
  Merged revisions 160480 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
    
    Jon Bonilla (Manwe) pointed out on the -dev list:
    "I guess that having only ip-phones in mind is not a good approach. Since it is
    possible to have a sip proxy connected to asterisk we could receive a 407
    (unauthorized) or 483 (too many hops) as response and dialog ending would not be
    a good behavior."
    So modified.
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2008-12-03 14:14:13 +00:00
Tilghman Lesher
a99affdaf8 Merged revisions 160308 via svnmerge from
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  r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines
  
  Merged revisions 160297 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines
    
    When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion
    fails, and the resulting integer is garbage.  Thus, we must initialize the
    integer and check it afterwards for success.
    (closes issue #14000)
     Reported by: folke
     Patches: 
           asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
           asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
           asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
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2008-12-02 18:03:01 +00:00
Kevin P. Fleming
c3347e79a5 Merged revisions 159818 via svnmerge from
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  r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines
  
  incorporates r159808 from branches/1.4:
  ------------------------------------------------------------------------
  r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
  
  update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
  
  since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
  
  format attributes in a consistent way
  
  
  ------------------------------------------------------------------------
  
  in addition:
  
  move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
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2008-11-29 18:16:50 +00:00
Mark Michelson
257123d217 Merged revisions 158265-158266 via svnmerge from
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r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines

Use some magic constants to get the right size
for this sscanf statement. Thanks Richard!


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r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines

Use a more expressive constant for a 64-bit scanned int


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2008-11-21 01:23:45 +00:00
Mark Michelson
ab27cc2a63 Merged revisions 158262 via svnmerge from
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r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines

Fix the build for 32-bit systems. %lu is only 32-bits
on 32-bit systems, so we need to use %llu instead. Of course
%llu is 128-bits on 64-bit systems, so we have to cast to
unsigned long long. No harm, but it's sure annoying.


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2008-11-21 01:00:58 +00:00
Mark Michelson
a55d5d6c3b Merged revisions 158230 via svnmerge from
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r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines

Change the remote user agent session version variable
from an int to a uint64_t. This prevents potential comparison
problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could
not properly place a call on hold since the version in the
SDP of the re-INVITE to place the call on hold greatly 
exceeded INT_MAX.

This also aligns with RFC 2327 better since it recommends
using an NTP timestamp for the version (which is a 
64-bit number).


(closes issue #13531)
Reported by: sgofferj
Patches:
      13531.patch uploaded by putnopvut (license 60)
Tested by: sgofferj


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2008-11-20 23:14:43 +00:00
Mark Michelson
8ffb368853 Change this so it actually compiles. Thanks, Terry!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-20 21:37:07 +00:00
Mark Michelson
a76dd11ef2 Merged revisions 158133 via svnmerge from
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r158133 | mmichelson | 2008-11-20 12:20:00 -0600 (Thu, 20 Nov 2008) | 10 lines

Merged revisions 158072 via svnmerge from 
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r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines

Begin on a crusade to end trailing whitespace!

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2008-11-20 18:20:39 +00:00
Mark Michelson
55e0bfb25a Merged revisions 158082 via svnmerge from
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r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines

Merged revisions 158071 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines

We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a 
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.


(closes issue #12994)
Reported by: pabelanger
Patches:
      12994.patch uploaded by putnopvut (license 60)

Closes AST-129


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