Commit Graph

10184 Commits

Author SHA1 Message Date
Jason Parker
6ef4425e81 Attempt to document some of the dependencies that are needed for net-snmp
Issue 8499 - initial patch by blitzrage.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-04 17:43:24 +00:00
Russell Bryant
0d8d3b7c43 When "fetch" is in use, instead of "wget", --continue is not a valid option.
(issue #8451)


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2006-12-03 06:34:14 +00:00
Olle Johansson
9636bccb7c - Removing one of two pieces of code to handle 481 response on INVITE
- Move handling of REFER response to handle_response_refer()


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2006-12-02 21:45:58 +00:00
Olle Johansson
f89143bd13 - Disable RTP hold timers while T.38 fax transmission happens
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
   The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
   something that video phones support in the RTP stream.
   I now this is a big architectual change at this stage for 1.4, but decided it was needed
   to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample

Issue 7679 in the bug tracker. Please test.



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2006-12-02 11:32:51 +00:00
Russell Bryant
1298cf0ea6 Backport the comment containing the warning regarding the limitations on the
usage of this function.  It is thread safe, but not technically reentrant.


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2006-12-02 03:50:58 +00:00
Kevin P. Fleming
a2f6499c0a Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines

if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)

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2006-12-01 23:37:28 +00:00
Russell Bryant
fee1970293 FreeBSD 6.1 does not include wget by default. However, it has fetch which will
work just fine for our purposes of downloading the sounds packages.  So, check
for both wget and fetch and the configure script and use what was found to
download them.  If neither one was found, and sound packages are selected that
must be downloaded, the install process will print out an informative error
message indicating the situation.

Also, fix a couple places where "make" was hard coded into some output messages
by replacing them with the $(MAKE) variable.

(issue #8451, initial patch by pabelanger, with additional modifications by me)


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2006-12-01 23:16:28 +00:00
Jason Parker
8cbe6025b6 Merged revisions 48183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines

Fix a small typo - issue 8848, reported by pabelanger

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2006-12-01 20:25:51 +00:00
Jason Parker
93eb47fdbe erm, hmm...I typod a merge...that went...strangely
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2006-12-01 20:24:32 +00:00
Jason Parker
0ba598baea Merged revisions 48182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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2006-12-01 20:19:44 +00:00
Tilghman Lesher
a78cc926d4 Double-unlock error (reported by blitzrage on IRC)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-01 19:38:59 +00:00
Olle Johansson
98d3fb64ed - Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states
- Remove support for T.38 early media, since it's impossible.

(Two patches in one - extra friday evening offer due to being off line from svn today... :-)


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2006-12-01 17:41:56 +00:00
Joshua Colp
b2b70adede Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
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2006-11-30 21:18:24 +00:00
Olle Johansson
76e35bac11 Issue 8319 - change noncecount before using it.
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2006-11-30 20:51:04 +00:00
Joshua Colp
b4edf92931 Blocked revisions 48161 via svnmerge
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r48161 | file | 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines

Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel driver. (issue #8390 reported by hselasky)

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2006-11-30 20:28:19 +00:00
Joshua Colp
422c97eab7 Merged revisions 48157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines

Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex)

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2006-11-30 20:07:55 +00:00
Joshua Colp
0bed57a7f1 Merged revisions 48154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 lines

Do not listen for DTMF on the bridge that comes into existence when ParkedCall is executed. This means native bridging can now occur for this. (issue #8406 reported by kebl0155)

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2006-11-30 19:05:14 +00:00
Joshua Colp
191c129b4f Merged revisions 48151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines

Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader)

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2006-11-30 18:47:40 +00:00
Joshua Colp
1771ee4f05 Blocked revisions 48146 via svnmerge
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r48146 | file | 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines

Remember the pointer to the allocated block of memory so that we can free it and not cause a memory leak. (issue #8449 reported by arkadia)

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2006-11-30 18:19:55 +00:00
Joshua Colp
802c3c3ecf Merged revisions 48142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines

Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)

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2006-11-30 17:57:35 +00:00
Olle Johansson
e49855c2bd Explain status reports and
make codefreeze more happy :-)


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2006-11-30 14:29:39 +00:00
Olle Johansson
073e937757 Clean up bad dialogs properly. Caused by GS 487 adapter without CSEQ on separate line
in the REGISTER request. Imported from 1.2.


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2006-11-30 09:12:30 +00:00
Joshua Colp
81c438713b Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 reported by slimey)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-29 21:05:17 +00:00
Olle Johansson
a68edf400f Explain the use device status system implemented in SIP for subscriptions,
queues and manager a bit better.

Like in 1.2, you will get more detailed information if you set a call 
limit for a device. When the call limit is reached, the status system will
report a device as busy.

For queues, setting a call limit per SIP device is propably a requirement.

In most cases, it will work much better if you only use type=peer and not
type=friend. We might decide to backport the new setting from trunk to
apply all call limits to the peer part of a friend only.



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2006-11-29 19:56:56 +00:00
Joshua Colp
558639ff6a Merged revisions 48106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines

If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3)

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2006-11-29 16:50:33 +00:00
Olle Johansson
3fe8e34039 Clarify RTP timers. Sorry, grandma.
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2006-11-29 08:03:36 +00:00
Joshua Colp
53b04eea95 Don't crash if the mailstream was not created.
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2006-11-29 04:26:53 +00:00
Jason Parker
1b872bbcfa Export several more variables in top level Makefile. Inspired by issue 8438.
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2006-11-28 18:26:53 +00:00
Joshua Colp
68ad03e888 Merged revisions 48087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines

According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3)

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2006-11-28 16:57:16 +00:00
Joshua Colp
ec4cae4068 Merged revisions 48053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 lines

Use the proper function to get the new message count instead of always using the filesystem. (issue #8421 reported by slimey)

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2006-11-27 18:06:50 +00:00
Tilghman Lesher
76d002db24 Merged revisions 48045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 Nov 2006) | 2 lines

Random MOH wasn't really random (bug 8381)

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2006-11-27 17:20:37 +00:00
Russell Bryant
a569bfd1f2 Remove a couple of unused variables (issue #8380, casper)
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2006-11-27 17:17:40 +00:00
Joshua Colp
fe50f34bab Merged revisions 48037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 lines

Do not reference the freed outgoing structure in the debug message. (issue #8425 reported by arkadia)

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2006-11-27 15:32:19 +00:00
Olle Johansson
9436ecaad7 Change logging message
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2006-11-27 06:41:33 +00:00
Steve Murphy
be037061e8 might as well also document the raw values of the flag vars
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2006-11-26 00:26:16 +00:00
Steve Murphy
f4b1302b95 A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it.
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2006-11-26 00:01:34 +00:00
Olle Johansson
5a3dcaff3c Not having a HINT is not an ERROR. In 1.4 and future releases, you can disable
subscription support totally or per peer in sip.conf with allowsubscribe = yes | no



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2006-11-25 09:28:28 +00:00
Steve Murphy
e6ce5ce979 bug 8189 posted this fix for main/translate.c for PLC
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2006-11-24 17:17:07 +00:00
Christian Richter
55229f98b9 Merged revisions 47968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line

fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm
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2006-11-24 15:46:13 +00:00
Olle Johansson
c3a7e9d7c2 Don't allocate unused variable.
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2006-11-23 11:01:47 +00:00
Joshua Colp
090c412c27 Video will never reach Packet2Packet bridging and can do more harm then good.
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2006-11-22 21:47:43 +00:00
Steve Murphy
a248c7308d blocking 47910 from 1.4; the time-limits on Dial are working OK
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-22 02:37:26 +00:00
Joshua Colp
8c0f4c8711 If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330 reported by voipgate)
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2006-11-21 17:32:27 +00:00
Olle Johansson
5c0be47e69 Apparently Exosip sends a 101 after a 100 provisional response. Let's not treat that as
early media. (discovered at the AVTF meeting in Paris).


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2006-11-21 15:20:03 +00:00
Tilghman Lesher
f4133080ca Oops, merge missed release of odbc object
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2006-11-20 20:01:58 +00:00
Tilghman Lesher
04c4fbb3b3 Merged revisions 47862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) | 2 lines

Failing to trap -1 error from mmap causes segfault (Issue 8385)

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2006-11-20 20:00:54 +00:00
Joshua Colp
0e684abd82 Merged revisions 47859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 lines

Don't forget to byte swap if we are exiting the smoother feed early. (issue #8287 reported by arturs)

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2006-11-20 19:51:36 +00:00
Joshua Colp
9d45f7420e Blocked revisions 47855 via svnmerge
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r47855 | file | 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines

Free history items at the end of use of the temporary SIP pvt structure. (issue #8383 reported by benh)

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2006-11-20 16:17:47 +00:00
Joshua Colp
b42279c142 Only remove/destroy the RTCP I/O item if it exists.
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2006-11-20 15:58:50 +00:00
Joshua Colp
335630b10c Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings)
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2006-11-20 15:51:37 +00:00