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2149 Commits

Author SHA1 Message Date
Luigi Rizzo
bfc782f4e9 Don't use a field size for the last argument of printf format, 
because in this case the string is left-aligned and it is not
truncated anyways.

Omitting the field size prevents the generation of trailing whitespace,
which makes the string fit in smaller windows.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-20 15:06:54 +00:00
Luigi Rizzo
b2fec9ad16 Extend the 'network settings' section with indication on the 
localnet settings (requires the change in SVN 76034), and also
give an indication on whether/why/how the remapping of addresses
in SIP message is done or not.

I think this is especially useful for debugging the configuration,
as the address remapping depends on a combination of at least 3
parameters (localnet, externhost, externip) and successful DNS lookup.

An example of the output of this section is below:

	Network Settings:
	---------------------------
	  SIP address remapping:  Enabled using externhost
	  Externhost:             foo.dyndns.net
	  Externip:               80.64.128.23:0
	  Externrefresh:          10
	  Internal IP:            12.34.56.78:5060
	  Localnet:               192.168.0.0/255.255.0.0
				  10.0.0.0/255.0.0.0

I leave to the community the judgement if the above info is a
useful addition for 1.4. It is not a bugfix, but it is neither a
new feature, only a useful diagnostic tool.

Note that I would like to move there also the bindaddress/port
information, in the usual addr:port format e.g.

          Bindaddress:            0.0.0.0:5060

so that network information is all in one place.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-20 14:54:01 +00:00
Steve Murphy
0e969271ae After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-19 23:24:27 +00:00
Luigi Rizzo
d60c5ee296 print more of the network settings (externip, externhost etc.) 
in the "sip show settings" cli output. I have put these in a
separate section, probably even bindaddr and SIP port should go
there.

There are more things to add here e.g. localnet and so on.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-19 08:07:04 +00:00
Luigi Rizzo
192ac53c3f document the use of externip, externhost and other nat-related options, 
as well as the handling of the sip socket.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-19 08:00:03 +00:00
Luigi Rizzo
fddd5b557c ast_sip_ouraddrfor() never fails, so make it void 
and remove the code that would never be called.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-19 07:51:34 +00:00
Luigi Rizzo
00d9a3e7a0 portability fix: use %f instead of %lf when printing double. 
The l is useless.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-19 07:41:45 +00:00
Tilghman Lesher
81bc1d7af5 Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 19:47:20 +00:00
Joshua Colp
a23feea9d2 Merged revisions 75623 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2 lines

Few more places that needs to check for onhold state.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 15:45:18 +00:00
Joshua Colp
d90bddfa6c Merged revisions 75621 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5 lines

(closes issue #10165)
Reported by: elandivar

It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 15:42:11 +00:00
Steve Murphy
5ac24b25d3 This corrects the problem with flags and %lld formats on 64-bit machines, where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 14:35:07 +00:00
Joshua Colp
4003b31fc5 Minor code tweaks. Variables were being checked wrong in some situations and didn't need to be checked in others. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-18 13:59:01 +00:00
Steve Murphy
8a7732f067 via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-17 19:40:29 +00:00
Steve Murphy
6bc0a4929c Merged revisions 74955 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1 line

This patch resolves 10143; thanks to irroot for the patch; looked acceptable. Let the community decide if it messes things up
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-12 20:46:32 +00:00
Olle Johansson
a1b9cbcd31 Implementation of a feature that will disable "missed calls" counters on SIP phones. 
If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.

Thanks to Ramon and Frank for feedback on this feature.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-09 08:27:37 +00:00
Tilghman Lesher
ba857cc8a9 Merged revisions 73985 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007) | 2 lines

Doxygen formatting fixes; fixes errors while 'make progdocs'.  (Closes issue #10104)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-09 04:09:16 +00:00
Olle Johansson
74e8ab14fc Merged revisions 73849 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2 lines

While tracking down a bug, I need some more history. Dumphistory is very useful, indeed.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-08 09:49:21 +00:00
Russell Bryant
1da115c8d9 Merged revisions 73769 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73769 | russell | 2007-07-06 18:02:58 -0500 (Fri, 06 Jul 2007) | 12 lines

Merged revisions 73768 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines

If a sip_pvt struct has already registered an extension state callback,
remove the old one before adding a new one.  If this isn't done, Asterisk
will crash.  (issue #10120)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-06 23:05:24 +00:00
Russell Bryant
a0c37d2548 Merged revisions 73679 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73679 | russell | 2007-07-06 10:57:25 -0500 (Fri, 06 Jul 2007) | 15 lines

Merged revisions 73678 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines

(closes issue #10125)
Reported by: makoto
Patches submitted by: makoto

This fixes a crash in chan_sip that happens when the bindaddr setting is not
valid on Asterisk startup, gets fixed, and then a reload gets issued.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-06 16:00:03 +00:00
Russell Bryant
134a556c9f Merged revisions 73598 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73598 | russell | 2007-07-05 18:59:22 -0500 (Thu, 05 Jul 2007) | 3 lines

Fix a crash in chan_sip.  Don't try to stop the monitor thread if it was never
started.  (closes issue #10124, reported by gzero, fixed by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-05 23:59:50 +00:00
Kevin P. Fleming
cc19ba80f5 Merged revisions 73548 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73548 | kpfleming | 2007-07-05 17:20:44 -0500 (Thu, 05 Jul 2007) | 10 lines

Merged revisions 73547 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) | 2 lines

we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support it for G.729

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-05 22:29:37 +00:00
Joshua Colp
0fc25ac3ee Merged revisions 73467 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73467 | file | 2007-07-05 16:18:02 -0300 (Thu, 05 Jul 2007) | 10 lines

Merged revisions 73466 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines

Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-05 19:20:12 +00:00
Jason Parker
daec10d187 Fix building with -Wdeclaration-after-statement, here too 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-28 19:35:23 +00:00
Joshua Colp
62084eb2a4 Add SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables when a transfer takes place. (issue #8378 reported by jcovert) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-27 23:13:09 +00:00
Joshua Colp
1961b57705 Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-26 23:31:23 +00:00
Joshua Colp
d77301b8cd Tweak CLI command completion and some help text. (issue #10049 reported by IgorG) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-25 15:35:10 +00:00
Joshua Colp
76455dda03 Merged revisions 71430 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71430 | file | 2007-06-24 21:10:06 -0400 (Sun, 24 Jun 2007) | 10 lines

Merged revisions 71414 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 lines

Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-25 01:11:47 +00:00
Joshua Colp
18f4920227 Merged revisions 70552 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70552 | file | 2007-06-20 18:22:20 -0400 (Wed, 20 Jun 2007) | 10 lines

Merged revisions 70551 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 lines

Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 22:24:47 +00:00
Russell Bryant
238b7a54cc Merged revisions 69944 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) | 10 lines

Fix a crash that could occur when handing device state changes.
When the state of a device changes, the device state thread tells the extension
state handling code that it changed.  Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent.  However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use
 the sip_pvt lock wrappers by eliel)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-19 15:27:16 +00:00
Tilghman Lesher
a67890d7a9 Merged revisions 69796 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007) | 2 lines

Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 19:52:56 +00:00
Joshua Colp
9ed0563f17 Merged revisions 69794 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2 lines

Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 19:02:45 +00:00
Joshua Colp
59bc48bd05 Merged revisions 69775 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69775 | file | 2007-06-18 14:18:12 -0400 (Mon, 18 Jun 2007) | 10 lines

Merged revisions 69765 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 lines

Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 18:19:54 +00:00
Joshua Colp
1dbfbe6d71 Merged revisions 69668 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2 lines

Don't defer the BYE till later on a transfer when the transfer itself goes kaboom and has no hope of working.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 16:06:17 +00:00
Joshua Colp
3e4980da79 Merged revisions 69661 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2 lines

Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 15:48:05 +00:00
Joshua Colp
981a94f023 Merged revisions 69625 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2 lines

Fix issue where it would be possible for the negotiated codecs to get set back to nothing. (issue #9992 reported by yehavi)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-18 13:57:33 +00:00
Russell Bryant
055d82cbce Add a massive set of changes for converting to use the ast_debug() macro. 
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-14 19:39:12 +00:00
Russell Bryant
8e11d6c147 Merged revisions 69183 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) | 9 lines

Move the logic for destroying a call when no response is received to a BYE
outside of the block that checks for FLAG_FATAL to be set.  This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted.  A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered.  This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235, patch from jcmoore)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-13 20:03:03 +00:00
Jason Parker
63535ada60 Fixes for ast_strlen_zero() janitor project. 
Issue 9968, patch by eliel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-13 17:06:53 +00:00
Russell Bryant
156d6338b2 Merged revisions 69071 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) | 3 lines

Clarify a bit of logic.  This doesn't change behavior in any way, but it is
helpful when following the logic to debug problems like 9235.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-13 16:59:42 +00:00
Russell Bryant
1d57ccb6f7 Fix a bunch of doxygen errors and document more things 
(issue #9842, snuffy)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-07 23:07:25 +00:00
Olle Johansson
1f9d98016e - Doxygen updates 
- Adding docs on flags to be able to clean up a bit


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-07 19:45:32 +00:00
Joshua Colp
f2cc861bcf Merged revisions 67941 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67941 | file | 2007-06-06 20:10:48 -0400 (Wed, 06 Jun 2007) | 10 lines

Merged revisions 67938 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines

Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-07 00:12:21 +00:00
Russell Bryant
6aec360466 Remove our little joke that was making fun of email disclaimers which nobody 
else seemed to think was very funny.  Oh well ... :)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 22:27:18 +00:00
Tilghman Lesher
9d05ff8ed5 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 21:20:11 +00:00
Russell Bryant
033a3df22a Merged revisions 67862 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) | 4 lines

Fix a crash when doing call pickups with SIP phones.  The code unlocked the
channel when it should not have.
(issue #9652, reported by corruptor, fixed by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 21:16:18 +00:00
Joshua Colp
e3492b9511 Merged revisions 67068 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 lines

Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 19:32:08 +00:00
Russell Bryant
93f2be2675 Fix a couple of places where "tos" was used instead of "cos". 
(issue #9540, patch by IgorG)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 15:14:52 +00:00
Russell Bryant
3d2b58751f To satisfy some legal concerns, add an option for chan_sip to include a 
disclaimer along with SIP messages in the header, X-Disclaimer.  This is off by
default.  Also, the text of the disclaimer can be customized in sip.conf.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-31 19:41:03 +00:00
Joshua Colp
9f79587d47 Merged revisions 66768 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines

Merged revisions 66764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines

It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-31 16:18:14 +00:00
Olle Johansson
0b2db74e5a Issue #9842 - Doxygen updates by snuffy. Thanks! 
(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-31 10:26:55 +00:00