Commit Graph

2371 Commits

Author SHA1 Message Date
Mark Michelson
f805a657a3 Merged revisions 151464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines
  
  Make the sip_standard_port function more granular by allowing separate
  type and port arguments. This is necessary because when building our From
  and Contact headers, we need to be absolutely sure that we are placing our
  source port there and not the peer's source port.
  
  (closes issue #12761)
  Reported by: asbestoshead
  Patches:
        patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 23:15:51 +00:00
Mark Michelson
8c5803b286 Merged revisions 179219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines
  
  Properly free memory and remove scheduler entries when a transmission failure occurs.
  
  Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit 
  was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called,
  this inevitably resulted in the reading and writing of freed memory.
  
  XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet
  at all. The proper action to take is to remove the scheduler entry we just created,
  free the packet's data as well as the packet itself, and unlink it from the list of
  packets on the sip_pvt structure.
  
  (closes issue #14455)
  Reported by: Nick_Lewis
  Patches:
        14455.patch uploaded by mmichelson (license 60)
  Tested by: Nick_Lewis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01 21:52:39 +00:00
Joshua Colp
be2caea0fa Merged revisions 178213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines
  
  Merged revisions 178205 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
    
    Skip check for extension when subscribing for MWI.
    
    Since the remote side is not actually subscribing to a specific extension when
    subscribing for MWI just skip the check to see if the extension exists. They can't use it
    to specify the mailbox either since we require configuration of that in sip.conf
    
    (closes issue #14531)
    Reported by: festr
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 15:20:47 +00:00
Jeff Peeler
5b4fb69323 Remove invalid ast_free calls for static character arrays
(issue #14478)
Reported by: erik_dedecker



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19 17:40:16 +00:00
Joshua Colp
b2a0fc1723 Merged revisions 177005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177005 | file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines
  
  Fix ordering of output for a ChannelUpdate manager event.
  (closes issue #14497)
  Reported by: vinsik
  Patches:
        chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18 17:13:25 +00:00
Dwayne M. Hubbard
24ef2a922e Merged revisions 176705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines
  
  create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38
  
  This is required to create a UDPTL structure in create_addr_from_peer() to handle the
  scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but 
  is defined the peer's context.  I tested this patch by enabling t38pt_udptl in the 
  [general] section on one system and only enabling t38pt_udptl in a peer's context on
  the system sending a fax.  Without the patch, the sending system will fail to initiate
  T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure".
  When this patch is applied the sending side will successfully initiate T38 negotiation.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 22:09:16 +00:00
Tilghman Lesher
dce3b59e28 Oops, merge broke 1.6.0.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 16:27:24 +00:00
Tilghman Lesher
0ab34a15f5 In 1.6.0, the tablename is stored in a variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 14:42:10 +00:00
Tilghman Lesher
d019ac6ef4 Merged revisions 176459 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines
  
  Merged revisions 176426 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines
    
    After a 'sip reload', qualifies for realtime peers weren't immediately
    restarted, instead waiting until the next registration.  We're now
    caching the qualify across a reload/restart and starting the qualify
    immediately upon loading the peer.
    (closes issue #14196)
     Reported by: pdf
     Patches: 
           20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
     Tested by: pdf
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 02:04:58 +00:00
Joshua Colp
d86ea77d16 Merged revisions 176030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines
  
  Merged revisions 176029 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines
    
    Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.
    
    This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
    is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
    pool was used for the value while the old was left untouched/unused. If the current pool was full a new
    pool was created. This would cause memory usage to increase steadily.
    
    (issue #AA50-2332)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 15:37:09 +00:00
Michiel van Baak
25c01347d8 Merged revisions 175952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines
  
  Merged revisions 175921 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines
    
    fix mis-spelling of the word registered.
    Reported by De_Mon on #asterisk-dev.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 09:40:22 +00:00
Russell Bryant
48ade8a53e Merged revisions 175368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines

Remove useless string copy, and make sscanf safe again

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 21:41:20 +00:00
Russell Bryant
869e8bc417 Merged revisions 175295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines

Avoid using ast_strdupa() in a loop.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 20:46:11 +00:00
Joshua Colp
346b766917 Merged revisions 174710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
  
  Only decrease inringing count if above zero.
  (issue #13238)
  Reported by: kowalma
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 20:16:57 +00:00
Joshua Colp
c558922e9c Merged revisions 174543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
  
  Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported.
  (closes issue #14399)
  Reported by: caspy
  (issue #13238)
  Reported by: kowalma
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 15:39:00 +00:00
Mark Michelson
f4113354e4 Merged revisions 174327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines

Fix something I messed up in the merge I just did


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 17:28:52 +00:00
Mark Michelson
0919e13437 Merged revisions 174301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines

Merged revisions 174282 via svnmerge from 
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09 17:25:09 +00:00
Dwayne M. Hubbard
24e312a999 Merged revisions 174084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines

Merged revisions 174082 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 23:59:42 +00:00
Joshua Colp
494e3efac7 Merged revisions 173974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | 15 lines
  
  Merged revisions 173967-173968 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    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
  ........
    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/branches/1.6.0@173986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 17:21:02 +00:00
Matthew Nicholson
76e95662cf Merged revisions 173952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines
  
  Merged revisions 173917 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines
    
    Limit the addition of the Contact header in SIP responses according to various
    SIP RFCs.
    
    (closes issue #13602)
    Reported by: hjourdain
    Tested by: mnicholson
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06 16:33:05 +00:00
Olle Johansson
3873563d82 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 09:56:53 +00:00
Olle Johansson
a43c0e9ad7 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27 15:13:10 +00:00
Olle Johansson
d23360562f 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-26 14:02:52 +00:00
Joshua Colp
9382891b4e 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
  
  ........
    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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 18:10:41 +00:00
Terry Wilson
7cdc49f51d We don't support ~expiry in 1.6.0 register statements
This must have inadvertantly got merged in sometime as the code doesn't handle
this option, and configs/sip.conf.sample doesn't mention it as available. So
just remove it from the WARNING message


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 20:13:18 +00:00
Terry Wilson
82e5da8bc4 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-17 00:29:43 +00:00
Mark Michelson
63c3ccd9b0 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16 22:46:11 +00:00
Olle Johansson
8633f3884f 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 18:14:20 +00:00
Terry Wilson
00647c1f8f 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
  
  ........
    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 22:31:20 +00:00
Mark Michelson
03a951f368 Merged revisions 168485 via svnmerge from
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r168485 | mmichelson | 2009-01-12 09:00:00 -0600 (Mon, 12 Jan 2009) | 13 lines

Merged revisions 168482 via svnmerge from 
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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2009-01-12 15:00:31 +00:00
Tilghman Lesher
8f41e52f44 Merged revisions 168334 via svnmerge from
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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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Mark Michelson
271c6c9529 Merged revisions 168193 via svnmerge from
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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 21:57:01 +00:00
Kevin P. Fleming
cad23358ab 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 17:29:36 +00:00
Kevin P. Fleming
85ac907af2 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 
  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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2009-01-08 17:22:40 +00:00
Mark Michelson
bfbf5afa62 Merged revisions 167180 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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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 17:04:23 +00:00
Tilghman Lesher
6964b59097 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 20:52:19 +00:00
Joshua Colp
665ba5d64b 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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2008-12-17 19:53:42 +00:00
Mark Michelson
3377535a60 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 
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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 23:07:06 +00:00
Russell Bryant
4fa477f58c Merged revisions 164675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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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 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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2008-12-16 16:02:22 +00:00
Joshua Colp
3c6b69d7a6 Merged revisions 164659 via svnmerge from
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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 15:45:46 +00:00
Joshua Colp
004570ef12 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 18:13:13 +00:00
Russell Bryant
1b1801f155 Merged revisions 163670 via svnmerge from
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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 18:52:14 +00:00
Russell Bryant
ab65296510 Merged revisions 163667 via svnmerge from
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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 18:34:56 +00:00
Russell Bryant
a3a784988d 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 18:21:35 +00:00
Joshua Colp
b1eb64beec Merged revisions 162805 via svnmerge from
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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 19:04:13 +00:00
Joshua Colp
b4d8725ec1 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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2008-12-10 17:54:06 +00:00
Mark Michelson
52eea9252e 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 
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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 16:36:38 +00:00
Joshua Colp
62f443578b 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 15:26:49 +00:00
Joshua Colp
77033f4ad0 Merged revisions 161726 via svnmerge from
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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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Tilghman Lesher
2c371ff2b7 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
  
  ........
    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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