Commit Graph

6367 Commits

Author SHA1 Message Date
Olle Johansson
64b76fa41a Merged revisions 242226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3 lines

Initialize notify_types to NULL


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-22 09:28:34 +00:00
Tilghman Lesher
04aa8b1149 Formats are inconsistent between even 32-bit and 64-bit Linux. Use casts to ensure both compile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-21 15:14:55 +00:00
Kevin P. Fleming
237f57cafe Fix up compile breakage from ast_tvdiff_ms() API change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-20 13:01:00 +00:00
Alec L Davis
c7be027151 Update CDR variables as pbx starts
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call,
by executing ast_cdr_update() early in __ast_pbx run.
Reverts sig_pri changes in trunk that are specific to isdn technology only.

(closes issue #16638)
Reported by: alecdavis
Patches: 
      cdr_update.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-20 08:18:45 +00:00
Jeff Peeler
716d3880fc small correction from 241314
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-19 18:51:19 +00:00
Jeff Peeler
4bb10fe6a9 Merged revisions 241227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 Jan 2010) | 13 lines
  
  Fix deadlock in agent_read by removing call to agent_logoff.
  
  One must always lock the agents list lock before the agent private. agent_read
  locks the private immediately, so locking the agents list lock is not an
  option (which is what agent_logoff requires). Because agent_read already 
  has access to the agent private all that is necessary is to do the required
  hanging up that agent_logoff performed.
  
  (closes issue #16321)
  Reported by: valon24
  Patches: 
        bug16321.patch uploaded by jpeeler (license 325)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-19 18:46:11 +00:00
Alec L Davis
800313cbd9 Update CDR variables before pbx starts (overlap dial)
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.

(issue #16638)
Reported by: alecdavis
Patches: 
      cdr_update.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-19 00:24:35 +00:00
Jeff Peeler
568c057c4c Extend max call limit duration from 24.8 days to 292+ million years.
If the limit was set past MAX_INT upon answering, the call was immediately
hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup).
The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been
changed to return an int64_t to prevent overflow. Also the reporter suggested
adding a message indicating the reason for the call hanging up. Given that the
new limit is so much higher, the message (which would only really be useful in
the overflow scenario) has been made a debug message only.

(closes issue #16006)
Reported by: viraptor


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-18 22:31:25 +00:00
Alec L Davis
12306debf2 Update CDR variables before pbx starts
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.

(closes issue #16638)
Reported by: alecdavis
Patches: 
      cdr_update.diff.txt uploaded by alecdavis (license 585)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-18 22:00:24 +00:00
Sean Bright
e612d87695 Convert a few places to use ast_calloc_with_stringfields where applicable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15 18:21:50 +00:00
Olle Johansson
c15a667733 SIP Show channelstats fix - use float division to show proper stats
(closes issue #15819)
Reported by: klaus3000
Patches: 
      asterisk-sip-show-channelstats-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000, oej

This patch is for trunk only and will be blocked in 1.6.2



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13 10:24:23 +00:00
David Vossel
fa0ef8031d fixes text support in sdp answer
The code that handled setting 'm=text' in the sdp was not executing
in the correct order.  The check to see if text was needed came after
the check to add 'm=text' to the sdp, this resulted in 'm=text' always
being set to 0 because it looked like text was never required.

(closes issue #16457)
Reported by: peterj
Patches:
      textportinsdp.diff uploaded by peterj (license 951)
      issue16457.diff uploaded by dvossel (license 671)
Tested by: peterj


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-12 16:14:41 +00:00
Tilghman Lesher
9c1b53cab4 According to POSIX, the capital L modifier applies only to floating point types.
Fixes a crash on Solaris.
(closes issue #16572)
 Reported by: crjw
 Patches: 
       frame_changes.patch uploaded by crjw (license 963)
       Plus several others found and fixed by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-10 19:37:30 +00:00
Richard Mudgett
fcf4f438f1 Fix using the wrong pointer type in do_idle_thread().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-07 21:24:06 +00:00
David Vossel
c8bb5daf1e Merged revisions 238411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines
  
  fixes crash in "scheduled_destroy" in chan_iax
  
  A signed short was used to represent a callnumber.  This is makes
  it possible to attempt to access the iaxs array with a negative
  index.
  
  (closes issue #16565)
  Reported by: jensvb
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-07 20:15:27 +00:00
David Vossel
084e235a8c Change in sip show channels display format allowing more digits for CID
(closes issue #16459)
Reported by: Rzadzins
Patches:
      chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-07 20:00:31 +00:00
Tilghman Lesher
82f998dcd4 Whoa, duplicate setting (dead code).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-06 06:53:23 +00:00
Richard Mudgett
c5cfc2a867 Removed unused parameters from analog_available() and sig_pri_available().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05 18:57:29 +00:00
Tilghman Lesher
91a45e4d3e Merged revisions 237318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines
  
  It's also possible for the Local channel to directly execute an Application.
  Reviewboard: https://reviewboard.asterisk.org/r/452/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-04 16:20:03 +00:00
Olle Johansson
8160e75638 - Disable res_pktccops by default
- Add dependency in chan_mgcp that was missing
- Add a small amount of doc to the source code


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-04 07:55:30 +00:00
Tilghman Lesher
0078b3bc5c global_contact_ha was renamed in trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-02 16:35:35 +00:00
Olle Johansson
6da31b48d7 Merged revisions 237135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines

Release memory of the contact acl before unloading module

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-02 09:54:22 +00:00
Tilghman Lesher
7a01655732 Merged revisions 236981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines
  
  Don't queue frames to channels that have no means to process them.
  (closes issue #15609)
   Reported by: aragon
   Patches: 
         20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14)
   Tested by: aragon
   
  Review: https://reviewboard.asterisk.org/r/452/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-30 21:59:18 +00:00
Jeff Peeler
1a219ad725 Fix compiling with LOW_MEMORY.
Modified handle_verbose to be LOW_MEMORY aware, removed old RTP related code
in chan_sip.

(closes issue #16381)
Reported by: michael_iedema
Patches: 
      ast_complete_source_filename.patch uploaded by michael iedema (license 942)
      modified by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-30 20:34:41 +00:00
Tilghman Lesher
5d2b47ffb8 Shut down the SIP session timers more gracefully, in order to prevent a possible crash.
(closes issue #16452)
 Reported by: corruptor
 Patches: 
       20091221__issue16452.diff.txt uploaded by tilghman (license 14)
 Tested by: corruptor


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-29 23:05:45 +00:00
David Vossel
6ecaa764d3 fixes iax "can't compress subclass 4294967295" error
(closes issue #16456)
Reported by: dvossel
Tested by: dvossel



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-22 21:53:46 +00:00
David Vossel
b74201f2b6 Merged revisions 236062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines
  
  fixes issue with p->method incorrectly set to ACK
  
  It is possible for a second ACK to come in for a retransmitted message.
  If an ack does not match an unacked message in our queue, restore the previous
  p->method as this ACK is completely ignored.
  
  (closes issue #16295)
  Reported by: omolenkamp
  Patches:
        issue16295_v2.diff uploaded by dvossel (license 671)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-22 17:00:08 +00:00
Kevin P. Fleming
ef9be94b35 Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-21 18:51:17 +00:00
Joshua Colp
ff0f861383 Remove some old code for going to the 'fax' extension when a T.38 switchover occurs. This would have
already happened when we detected the CNG tone so this was basically a noop.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-17 23:21:07 +00:00
Jeff Peeler
50b7338d02 Fix call forwarding for analog phones.
(closes issue #16440)
Reported by: mmichelson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-17 00:52:03 +00:00
David Vossel
181f617fd7 reverse minor sip registration regression
A registration regression caused by a code tweak in (issue #14331)
and a bug fix in (issue #15539) caused some sip registration
config entries to be constructed incorrectly.  Origially
issue #14331 contained the code tweak as well as a bug fix, but since
the issue was reported as a tweak the bug fix portion was moved into
issue #15539.  Both the tweak and the bug fix contained minor incorrect
logic that resulted in some SIP registrations to fail.

(issue #14331)
(issue #15539)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15 18:43:06 +00:00
Olle Johansson
2d49d547ed Merged revisions 234492 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 lines

Stop sending 183's after call hangup.

There where still cases where the 183 keep-alive mechanism would not stop
sending 183's even though the Asterisk server had sent a final reply to
the invite.

EDVX-28

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-14 10:46:20 +00:00
Tilghman Lesher
84678fc77d Merged revisions 234095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009) | 9 lines
  
  When we receive no response at all to our INVITE, allow the channel to be destroyed.
  (closes issue #15627)
   Reported by: falves11
   Patches:
         20091209__issue15627__1.6.0.diff.txt uploaded by tilghman (license 14)
         20091209__issue15627__1.4.diff.txt uploaded by tilghman (license 14)
   Tested by: falves11
  Review: https://reviewboard.asterisk.org/r/446/
  (closes issue #15716)
  Reported by: dant
  (closes issue #16270)
  Reported by: corruptor
  (closes issue #15356)
  Reported by: falves11
  (issue #16382)
  Reported by: lftsy
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-10 16:24:26 +00:00
David Vossel
86dc66625c Merged revisions 233471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines
  
  fixes missing Contact header angle brackets
  
  (closes issue #16298)
  Reported by: mgernoth
  Patches:
        reg_parse_issue_1.4.diff uploaded by dvossel (license 671)
  Tested by: dvossel
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-07 18:08:46 +00:00
Matthew Nicholson
3069bab67c Do not reject SDP packets describing only non audio streams.
(closes issue #16387)
Reported by: zalex1953
Patches:
      media-level-c-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, zalex1953


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-07 16:14:42 +00:00
Michiel van Baak
74c5417696 Let's unlock the lines list after the AST_LIST_TRAVERSE instead of inside it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04 16:55:07 +00:00
Michiel van Baak
c1dfce9753 Only assign line and device in handle_transfer_button when we have a subchannel.
(closes issue #16040)
Reported by: ebroad


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04 16:18:57 +00:00
Mark Michelson
74b388ea4a Do not change the exten string field or rebuild the contact header
on an inbound sip_pvt if the outbound call is redirected.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 17:18:07 +00:00
Joshua Colp
6a7f37e07d Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response.
(closes issue #16186)
Reported by: atis
Patches:
      sip_t38_response_415.patch uploaded by atis (license 242)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 16:40:14 +00:00
Joshua Colp
23781604aa Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario
causing code to execute during reload that should not.

(issue AST-263)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 14:54:28 +00:00
Jeff Peeler
08ec550c16 Merged revisions 232090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines
  
  Do not modify the gain settings on data calls.
  
  (The digital flag actually represents a data call.)
  
  (closes issue #15972)
  Reported by: udosw
  Patches: 
        transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 00:45:18 +00:00
Tilghman Lesher
f59fe83c56 More 32->64 bit codec conversions.
In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field.  Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-01 20:27:37 +00:00
Kevin P. Fleming
5ba2b689b2 Another round of UDPTL stack fixes/improvements:
1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
   session, so that log/error/debug messages generated by the UDPTL stack can
   be 'connected' to the endpoint that caused them to be generated.

2) Improve comments (and process) of calculating the far end's maximum IFP size
   when redundancy mode is in use for error correction.

3) When an IFP larger than the calculated 'far max IFP' size is presented for
   writing, truncate it rather than putting in the buffer and allowing the buffer
   to overflow; this will cause the ends to retrain to a lower bit rate that
   produces IFPs of an appropriate size if possible, and if not possible, the
   FAX transfer will fail completely. In these cases, it is due to the one endpoint
   supplying a T38FaxMaxDatagram value that is improperly calculated and is
   too low to be of use; we have configuration options available to override
   this behavior.

4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
   needed.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30 21:47:42 +00:00
Joshua Colp
c899add7f9 When receiving SDP that matches the version of the last one do not treat it as a fatal error.
(closes issue #16238)
Reported by: seandarcy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30 20:44:30 +00:00
Joshua Colp
28b009b266 Fix a bug where an immediate masquerade would cause a queued unhold frame to get lost. Now we just
indicate unhold directly after the masquerade is complete.

(issue ABE-2011)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30 16:29:29 +00:00
Jeff Peeler
ac9a4cecf3 Fix problem on digital channels due to digital flag not getting set
Changed areas in sig_pri to set the digital flag using a callback that will
also set the corresponding flag in chan_dahdi. Modified dahdi_request slightly
so that if a bearer is marked as digital, that information is available when
creating the new channel.

(closes issue #16151)
Reported by: alecdavis
Patch based on bug_16151.diff.txt uploaded by alecdavis (license 585)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-24 17:12:25 +00:00
Joshua Colp
60e10aba46 Change fax detection in chan_sip so it behaves as one would expect.
Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-23 15:45:45 +00:00
Kevin P. Fleming
fe1ebc8d46 Merged revisions 230839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line
  
  Correct fix for issue #16268... the reporter's original patch was very close to correct.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-23 15:34:16 +00:00
Kevin P. Fleming
1f759eddfa Merged revisions 230772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines
  
  Ensure that SDP parsing does not ignore the last line of the SDP.
  
  (closes issue #16268)
  Reported by: sgimeno
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-23 14:15:48 +00:00
David Vossel
417657f4ed fixes iax2 show cache locking error, thanks alecdavis!
(closes issue #16094)
Reported by: alecdavis
Patches:
      bug16094.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, dvossel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-20 22:35:54 +00:00