Olle Johansson
90bad9d2f5
Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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2007-05-16 07:35:56 +00:00
Matthew Fredrickson
9b5454232c
XXX-XXX-XXX appears to be the standard ANSI pointcode format
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2007-05-15 20:45:20 +00:00
Matthew Fredrickson
79f553a4ab
Only print the SS7 UP once. Not every time we get the test messages on the
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line.
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2007-05-14 22:25:56 +00:00
Olle Johansson
fa2622cb1d
Merged revisions 64324 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines
Change -2 to XMIT_ERROR to clarify a bit more
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2007-05-14 19:35:58 +00:00
Russell Bryant
ee7217c011
Merged revisions 64306 via svnmerge from
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r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
will trigger an error and cause sounds to stop, which in this case, is ringing.
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2007-05-14 19:21:31 +00:00
Joshua Colp
4fbb449444
If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
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2007-05-14 18:21:30 +00:00
Steve Murphy
a76ff39c0e
Merged revisions 64193 via svnmerge from
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r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 line
As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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2007-05-14 14:13:45 +00:00
Olle Johansson
fc057b1890
Improve handling network errors on transmission to hosts that don't reply or are unreachable
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With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.
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2007-05-13 19:20:36 +00:00
Joshua Colp
b71e691b3e
Merged revisions 64114 via svnmerge from
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r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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2007-05-12 22:28:09 +00:00
Joshua Colp
38e951cfda
Merged revisions 64086 via svnmerge from
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r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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2007-05-12 21:12:18 +00:00
Joshua Colp
82a30356da
Merged revisions 64044 via svnmerge from
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r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines
Ensure the onhold flag is set no matter what when being put on hold.
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2007-05-12 16:33:34 +00:00
Jason Parker
60f6a72123
Add/fix support for Redial, Speeddial, and Messages buttons.
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Combined effort by DEA and mvanbaak.
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2007-05-11 22:52:36 +00:00
Jason Parker
2768946cf5
Merged revisions 63830 via svnmerge from
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r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, 10 May 2007) | 12 lines
Merged revisions 63828 via svnmerge from
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r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines
Fix an issue with trying to kill a thread before it gets created.
Issue 9709, patch by nic_bellamy.
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2007-05-10 23:16:33 +00:00
Olle Johansson
aa320037d2
Merged revisions 63749 via svnmerge from
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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines
Merged revisions 63748 via svnmerge from
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r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines
Do not allocate SIP pvt's for PEERs we can not reach.
This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.
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2007-05-10 20:51:59 +00:00
Matthew Fredrickson
e2ca869abd
Merged revisions 63654 via svnmerge from
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r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, 09 May 2007) | 10 lines
Merged revisions 63653 via svnmerge from
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r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines
Make sure we only create a DSP if it's requested on SUB_REAL
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2007-05-09 17:35:54 +00:00
Joshua Colp
7e10164e20
Merged revisions 63611 via svnmerge from
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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines
Merged revisions 63610 via svnmerge from
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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2007-05-09 16:56:45 +00:00
Olle Johansson
c358b18a5a
Merged revisions 63532 via svnmerge from
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r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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2007-05-09 13:07:44 +00:00
Russell Bryant
314c874d7d
I noted this on the dev list but got no response, so I just did it myself.
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Lock the call features when being used in chan_sip.
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2007-05-08 16:41:35 +00:00
Olle Johansson
a39f95b94f
Adding external referenses for doxygen
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See http://www.asterisk.org/doxygen/trunk/extref.html
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2007-05-07 18:25:56 +00:00
Olle Johansson
33214f76a7
Adding external reference
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2007-05-07 18:18:43 +00:00
Olle Johansson
238bcd6c45
Doxyfication... There's a shortage of comments in this file...
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2007-05-07 18:17:18 +00:00
Joshua Colp
28f4727e75
Lock iax2 pvt structure when passing off to the AMI function, and make sure it exists. (issue #9674 reported by arabe)
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2007-05-06 20:09:18 +00:00
Olle Johansson
d326d84ae0
- Adding some missing spaces
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- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily
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2007-05-05 08:05:38 +00:00
Steve Murphy
02337303ef
a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
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2007-05-04 17:49:20 +00:00
Steve Murphy
3ee0077f04
Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
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2007-05-04 16:37:23 +00:00
Olle Johansson
1b15d8852d
Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
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is needed to follow the call through the PBX.
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2007-05-04 13:56:25 +00:00
Joshua Colp
81cade7a4c
Merged revisions 62989 via svnmerge from
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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines
Merged revisions 62987 via svnmerge from
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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines
When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)
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2007-05-03 16:45:39 +00:00
Tilghman Lesher
121561076e
Merged revisions 62692 via svnmerge from
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r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines
Merged revisions 62691 via svnmerge from
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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.
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2007-05-02 17:49:36 +00:00
Steve Murphy
fe7068a51b
Merged revisions 62689 via svnmerge from
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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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2007-05-02 17:24:03 +00:00
Russell Bryant
3d409eb793
Update the device state functionality of chan_local such that it will return
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NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048 , patch from tim_ringenbach)
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2007-05-02 15:46:49 +00:00
Olle Johansson
e1ec3f917c
Add a small message that we're doing something. On my systems, there's a long
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dead period with a non-responsive CLI after I issue "load chan_sip.so"
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2007-05-02 12:12:02 +00:00
Olle Johansson
1d51b2e161
More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
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But first, some serious SIP testing :-)
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2007-05-02 12:00:03 +00:00
Olle Johansson
8fee67c83b
Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly
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2007-05-02 09:41:03 +00:00
Olle Johansson
daefa6a8b4
Merged revisions 62624 via svnmerge from
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r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines
Don't unlock a channel that we already know does not exist (propably isue 8228)
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2007-05-02 09:35:14 +00:00
Russell Bryant
b419fc1134
Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540 , patch submitted by IgorG)
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2007-04-30 16:16:26 +00:00
Russell Bryant
a91f9b138d
Merged revisions 62419 via svnmerge from
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r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines
Merged revisions 62417 via svnmerge from
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588 , reported and patched by softins)
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2007-04-30 15:59:27 +00:00
Russell Bryant
5cb08adc7a
Don't crash when invalid arguments are provided to the CHANNEL() function
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for a SIP channel.
(issue #9619 , reported by jtodd, original patch by Corydon76, committed patch
slightly modified by me)
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2007-04-30 15:37:23 +00:00
Russell Bryant
0efe511879
Merged revisions 62331 via svnmerge from
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r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines
Fix a bug that made the "language" setting in zapata.conf not
functional. (issue #9626 , reported and fixed by sergee)
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2007-04-29 05:51:18 +00:00
Russell Bryant
7d2102c081
Reformat some of iax2.h and convert comments to doxygen format
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2007-04-28 21:26:00 +00:00
Russell Bryant
b6b1bf3213
Merge changes from team/russell/events
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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2007-04-28 21:01:44 +00:00
Russell Bryant
c2468b4c32
Merged revisions 62218 via svnmerge from
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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines
Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615 , #9616 , reported by jiddings and BigJimmy, and fixed by me)
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2007-04-27 21:11:46 +00:00
Olle Johansson
240bd841b0
Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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2007-04-27 14:40:28 +00:00
Olle Johansson
f9c592e50c
Merged revisions 62137 via svnmerge from
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r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines
Merged revisions 62126 via svnmerge from
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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
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2007-04-27 14:37:10 +00:00
Joshua Colp
14d8979b2f
Merged revisions 62038 via svnmerge from
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r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines
Merged revisions 62037 via svnmerge from
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
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2007-04-26 16:35:14 +00:00
Kevin P. Fleming
ee95074173
Merged revisions 61914 via svnmerge from
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r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61913 via svnmerge from
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286 )
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2007-04-25 22:34:58 +00:00
Russell Bryant
dc7514a746
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r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines
If the callerid= option is specified, but empty, clear any previous data.
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2007-04-25 22:01:37 +00:00
Russell Bryant
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r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines
Ensure that callerid settings are reset on a reload.
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2007-04-25 21:15:19 +00:00
Russell Bryant
f5d4a16cda
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r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587 , reported and patched by xrg)
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2007-04-25 16:23:00 +00:00
Dwayne M. Hubbard
f3ab33014a
removed #if 0 block from chan_zap restart_monitor()
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2007-04-24 19:08:28 +00:00
Joshua Colp
721f85d084
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r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines
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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines
Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)
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2007-04-24 16:10:10 +00:00
Russell Bryant
b94378b5e6
Merge changes from team/russell/iax2_osp
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This set of changes adds OSP support to chan_iax2. However, I have modified
the patch a bit from what was submitted. You now use the CHANNEL() function
to get and set the OSP token for IAX2.
(issue #8531 , reported by and original patch by homesick, patch updated by me)
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2007-04-20 21:12:53 +00:00
Olle Johansson
49af71c100
Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-)
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This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason.
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2007-04-20 08:41:24 +00:00
Dwayne M. Hubbard
34469a8707
added CLI 'sip unregister <peer>' for issue 9326. thanks eliel
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2007-04-13 21:23:10 +00:00
Joshua Colp
4f04ff8597
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r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2 lines
For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg)
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2007-04-13 17:21:53 +00:00
Steve Murphy
901413c76c
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r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 line
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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2007-04-13 17:11:53 +00:00
Joshua Colp
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r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2 lines
Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa)
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2007-04-13 16:35:33 +00:00
Joshua Colp
c4c2def716
Don't treat a host lookup as failed if sipregs is not in use when doing a realtime lookup. (issue #9255 reported by sergee)
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2007-04-12 19:32:00 +00:00
Russell Bryant
3c0b24bda8
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r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines
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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines
If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all. It is an optional header, anyway. Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason. (issue #9488 , reported by makoto, fixed by me)
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Nadi Sarrar
9978dc647b
Merged revisions 61342,61372-61373,61443 via svnmerge from
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r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2 lines
AOCD's are now exported to asterisk channel variables.
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r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr 2007) | 2 lines
Ignore facility messages in case we don't have a corresponding channel object.
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r61373 | nadi | 2007-04-11 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines
Export AOCD variables on misdn_hangup.
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r61443 | nadi | 2007-04-11 17:39:14 +0200 (Mi, 11 Apr 2007) | 2 lines
Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..
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2007-04-11 15:48:54 +00:00
Russell Bryant
6b033eea04
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r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines
Merged revisions 61426 via svnmerge from
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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines
Fix a bug with switching between host=dynamic and using specific hosts for
peers. The code would only reset the peer's address when it is dynamic if
it was a new peer structure. Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515 , reported by caio1982, fixed by me)
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2007-04-11 15:13:12 +00:00
Russell Bryant
e34c67d308
Merged revisions 61377 via svnmerge from
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r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines
Merged revisions 61376 via svnmerge from
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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
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2007-04-11 14:13:08 +00:00
Joshua Colp
9fff461080
Remove duplicate prototype declaration. (issue #9517 reported by junky)
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2007-04-11 14:01:53 +00:00
Doug Bailey
7c9e46507b
Added zapata.conf parameter "cid_rxgain" to allow the user to adjust the gain bump used during CID acquisition.
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2007-04-10 22:02:53 +00:00
Steve Murphy
28b5fb02bd
updated ast_channel_alloc() call to include the 4 extra args everyone got. Not much info there, as the config file evidently does not allow amaflags, or accountcode settings; and the pvt's exten doesn't sound like what we need in the cdr, either.
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2007-04-10 16:07:10 +00:00
Nadi Sarrar
0220ec0511
Merged revisions 61183 via svnmerge from
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r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di, 10 Apr 2007) | 10 lines
Merged revisions 61170 via svnmerge from
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r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 lines
msns config parameter defaults to '*'
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2007-04-10 12:47:57 +00:00
Steve Murphy
ecaf781933
Merged revisions 60989 via svnmerge from
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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2007-04-10 05:41:34 +00:00
Olle Johansson
b52f774850
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r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines
Merged revisions 61038 via svnmerge from
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r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 lines
- Don't send ActionID before Response: header.
- Don't use a blank in an AMI header
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2007-04-09 20:01:28 +00:00
Olle Johansson
4aef0155d6
use "ChannelType" in events to indicate which channel driver that generates the event. This replaces
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"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"
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2007-04-09 18:22:43 +00:00
Joshua Colp
a4bef3bb3a
Make RTP session ID and session version generation random. (issue #9456 reported by tjardick)
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2007-04-09 12:33:49 +00:00
Tilghman Lesher
c6c64af72d
Merged revisions 60847 via svnmerge from
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r60847 | tilghman | 2007-04-08 21:42:48 -0500 (Sun, 08 Apr 2007) | 10 lines
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r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines
Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid.
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2007-04-09 02:44:37 +00:00
Joshua Colp
ed75ded048
Add counter for sip show registry CLI command. (issue #9352 reported by junky)
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2007-04-09 00:47:06 +00:00
Olle Johansson
5bc2aa8ab1
Use the same parameter to the two "Registry" AMI events - ChannelDriver
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2007-04-06 19:26:01 +00:00
Matthew Fredrickson
bc0310e94e
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r60485 | mattf | 2007-04-06 13:21:52 -0500 (Fri, 06 Apr 2007) | 2 lines
Make sure we check the faxdetect option before doing fax processing
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2007-04-06 18:26:24 +00:00
Matthew Fredrickson
1031bbba01
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r60459 | mattf | 2007-04-06 12:32:31 -0500 (Fri, 06 Apr 2007) | 10 lines
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r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 lines
There should only be one code path for doing DTMF conditionals on channels. This fixes it.
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2007-04-06 18:25:38 +00:00
Joshua Colp
95a7dc0509
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r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, 05 Apr 2007) | 10 lines
Merged revisions 60213 via svnmerge from
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r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines
Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa)
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2007-04-05 12:57:35 +00:00
Russell Bryant
d1588ce2d5
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r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) | 3 lines
Add a Content-Length of 0 to the response built by transmit_response_with_unsupported().
(issue #9454 , reported by makoto, fixed by me)
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2007-04-04 16:50:31 +00:00
Russell Bryant
bb53ef9d32
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r60088 | russell | 2007-04-04 11:39:04 -0500 (Wed, 04 Apr 2007) | 12 lines
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r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines
Fix the return value of handle_common_options() so that it always properly
indicates whether it handled the option or not.
(issue #9455 , reported by Netview, fixed by me)
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2007-04-04 16:40:01 +00:00
Russell Bryant
b2ddaaf033
Add support for RTP packetization in chan_jingle and chan_gtalk.
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(issue #9416 , phsultan)
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2007-04-03 22:33:03 +00:00
Russell Bryant
11ab6db24b
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r59939 | russell | 2007-04-03 14:16:53 -0500 (Tue, 03 Apr 2007) | 12 lines
Merged revisions 59938 via svnmerge from
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r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines
Don't attempt to report configuration errors in build_user(). oej pointed out
that for a "friend" entry, this won't work, because all user options are valid
for peers, but not the other way around.
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2007-04-03 19:17:55 +00:00
Russell Bryant
3f14c8b6fc
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r59936 | russell | 2007-04-03 13:55:57 -0500 (Tue, 03 Apr 2007) | 11 lines
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r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines
Make chan_sip report when it encounters an unknown option.
(issue #9440 , reported by nightcrawler)
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2007-04-03 18:57:52 +00:00
Russell Bryant
b908f9717a
Remove a duplicate function prototype. (issue #9444 , junky)
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2007-04-03 18:34:14 +00:00
Nadi Sarrar
4d1edcd4fd
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r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di, 03 Apr 2007) | 15 lines
Merged revisions 59788,59803 via svnmerge from
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r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines
Use the new sysfs way of mISDN 1.2 to check if a port is NT or not.
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r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines
ptp is the 5th bit, not the 4th.
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2007-04-03 11:15:45 +00:00
Russell Bryant
592870b6ce
This hashing code is still causing some random crashes on my system, and
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probably others, too. I don't really have time to work on it at the moment,
so I am just going to revert it for now.
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2007-04-02 17:40:26 +00:00
Matthew Fredrickson
d6e7b047f0
Update to support multiple CIC groups and DPCs per linkset.
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2007-03-30 20:54:42 +00:00
Russell Bryant
5d4e6a480a
Fix an issue with hashing iax2 pvt structures that caused random crashes on
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systems under heavy load such as IAXtel.
(possibly related to issue #9403 )
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2007-03-29 18:31:35 +00:00
Russell Bryant
5a1c6826ce
Merged revisions 59341 via svnmerge from
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r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) | 8 lines
When the IAX2 read callback gets called, return NULL instead of a "null frame".
This will cause Asterisk to hangup the call instead of keep trying whatever it
was doing. Under normal conditions, this function would *never* be called.
However, the author of this patch says an error will occur that will cause it
to get called every 100 thousand calls or so. When this does happen, it puts
the channel in a loop that eventually brings down the system. So, hangup up
the call is certainly a better alternative. (issue #8286 , john)
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2007-03-29 16:56:35 +00:00
Dwayne M. Hubbard
6626c31f35
added filtering options to 'zap show channels' to implement functionality described in issue 6520
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2007-03-28 00:09:25 +00:00
Joshua Colp
04f1955781
Use better english. Renegotiate! Repeat after me: renegotiate.
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2007-03-27 20:11:11 +00:00
Russell Bryant
93e2d66f13
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r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) | 3 lines
Fix the check that ensures that the CHANNEL function's first argument is "rtpqos".
Thanks, Corydon. :)
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2007-03-27 18:18:36 +00:00
Russell Bryant
b5adacebf1
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r59259 | russell | 2007-03-27 13:05:46 -0500 (Tue, 27 Mar 2007) | 12 lines
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r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | 4 lines
Fix the use of the "sourceaddress" option when "bindaddr" is set to 0.0.0.0
instead of having each interface explicitly listed.
(issue #7874 , patch by stevens)
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2007-03-27 18:08:59 +00:00
Russell Bryant
7c884d76ea
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r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines
Convert the RTPQOS function to just be additional parameter of the CHANNEL
function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
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2007-03-27 16:25:02 +00:00
Tilghman Lesher
0e0600a446
Merged revisions 59227 via svnmerge from
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r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines
Change this to a single dp function to make oej happy.
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2007-03-26 21:44:59 +00:00
Russell Bryant
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r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | 1 line
Rename the new dialplan functions to match the variable name
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2007-03-26 17:57:50 +00:00
Russell Bryant
08e3a9bdc8
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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370 , patch by Corydon76, with some testing by blitzrage)
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2007-03-26 17:51:27 +00:00
Matthew Fredrickson
e4a7d1b35f
Fix bug in which parameter type we are passing. This shouldn't be a problem
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since both types are the same underneath.
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2007-03-26 16:48:29 +00:00
Matthew Fredrickson
df42d78569
Small API related SS7 updates.
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2007-03-26 16:46:07 +00:00
Nadi Sarrar
24d8595d00
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r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
(the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).
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2007-03-26 15:59:56 +00:00
Joshua Colp
cc22f60f30
Merged revisions 59195 via svnmerge from
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r59195 | file | 2007-03-23 21:39:44 -0400 (Fri, 23 Mar 2007) | 10 lines
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r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines
Only try to handle a response if it has a response code. (ASA-2007-011)
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2007-03-24 01:42:11 +00:00
Kevin P. Fleming
e8e9e5e23c
Merged revisions 59182 via svnmerge from
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r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007) | 2 lines
don't allow string input to overrun the buffer to hold it (ASA-2007-010)
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2007-03-22 23:41:37 +00:00
Kevin P. Fleming
a6d4f89769
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r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22 Mar 2007) | 2 lines
remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)
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2007-03-22 23:39:26 +00:00
Joshua Colp
af9c17025f
Minor tweak. Only queue up an unhold control frame if we are actually on hold. This would have shown itself when a call was initially being setup and the SDP data was being parsed in.
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2007-03-21 03:33:57 +00:00
Joshua Colp
1d5be2d1c7
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r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2 lines
Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown)
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2007-03-21 03:27:58 +00:00
Joshua Colp
2874a4317d
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r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar 2007) | 2 lines
Copy len variable as well, should fix remaining IAX2 DTMF issues.
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2007-03-20 20:44:58 +00:00
Olle Johansson
dddb57b242
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r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3 lines
Issue #9313 , Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)
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2007-03-18 20:39:37 +00:00
Joshua Colp
4787adb4e9
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r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2 lines
Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado)
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2007-03-15 18:21:03 +00:00
Russell Bryant
1bc728ed4b
Merged revisions 58906 via svnmerge from
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r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | 4 lines
Some people like to put "limitonpeer" instead of "limitonpeers" in their
configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172 )
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2007-03-14 19:19:00 +00:00
Olle Johansson
f9c3f60ab9
Merged revisions 58848 via svnmerge from
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r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, 13 Mar 2007) | 10 lines
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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines
Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)
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2007-03-14 16:59:35 +00:00
Olle Johansson
b4490af8ac
Merged revisions 58845 via svnmerge from
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r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3 lines
Don't hangup the call on OK or errors on MESSAGE and INFO
inside of a dialog (like video update requests).
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2007-03-13 10:14:13 +00:00
Olle Johansson
32f9227ca4
Merged revisions 58843 via svnmerge from
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r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2 lines
Issue #9251 - Clear From URI from user attributes (tgrman)
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2007-03-13 09:15:17 +00:00
Olle Johansson
d008232b27
Change URL to OpenH323 (thanks, Tzafrir!)
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2007-03-12 09:37:13 +00:00
Joshua Colp
ea226e9d77
Merged revisions 58779 via svnmerge from
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r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
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2007-03-12 00:54:13 +00:00
Russell Bryant
5dc7f1d898
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r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | 6 lines
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
frame was not properly initialized.
- Interpolating a frame when the jitterbuffer is in use
- decrypting a frame when IAX2 encryption is on
- frames in an IAX2 trunk
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2007-03-10 18:15:41 +00:00
Joshua Colp
5629ee0292
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r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, 09 Mar 2007) | 10 lines
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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
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2007-03-09 20:53:37 +00:00
Russell Bryant
d04a9e0bc0
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r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | 3 lines
Add a missing break statement so that handling the above event does not
incorrectly destroy the channel. (issue #9242 , andrew)
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2007-03-08 22:08:28 +00:00
Russell Bryant
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r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | 6 lines
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
(issue #7256 , tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
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2007-03-08 01:06:00 +00:00
Russell Bryant
f86aee41f7
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r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines
(This bug was reported to me by Kinsey Moore)
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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2007-03-07 18:20:51 +00:00
Joshua Colp
e7da006562
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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines
Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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2007-03-07 17:55:11 +00:00
Steve Murphy
5bb1d16f47
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r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, 06 Mar 2007) | 9 lines
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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2007-03-06 23:19:59 +00:00
Olle Johansson
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r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, 06 Mar 2007) | 10 lines
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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines
Change error message to proper message
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2007-03-06 20:45:29 +00:00
Russell Bryant
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r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines
Return an error of transmit_response is called without a session.
(issue #9002 )
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2007-03-06 18:02:35 +00:00
Joshua Colp
b7e47198da
Add zap show version CLI command. This pulls the version/echo canceller in use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai)
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2007-03-05 20:13:51 +00:00
Joshua Colp
2adf4fa4e0
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r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 lines
Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)
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2007-03-05 19:20:45 +00:00
Joshua Colp
e23433e50e
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r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines
If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.
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2007-03-02 17:09:18 +00:00
Joshua Colp
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines
Merged revisions 57317 via svnmerge from
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines
Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)
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2007-03-01 22:23:26 +00:00
Joshua Colp
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r57093 | file | 2007-02-28 13:57:52 -0500 (Wed, 28 Feb 2007) | 10 lines
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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines
Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)
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2007-02-28 19:00:24 +00:00
Tilghman Lesher
a3da18c244
Issue 7789 - some telcos want the TON set based on the number, but without the NANP prefix removed
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2007-02-27 00:11:32 +00:00
Russell Bryant
b3776415aa
Merged revisions 56888 via svnmerge from
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r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | 4 lines
Restore the behavior of Asterisk 1.2 where if a device was not specified in
alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139 )
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2007-02-26 20:43:18 +00:00
Russell Bryant
c617945b5c
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r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) | 2 lines
Fix a crash in my last change to iax2_indicate(). (issue #9150 )
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2007-02-26 20:04:53 +00:00
Russell Bryant
82179d1202
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) | 3 lines
Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
(Problem brought up by Ben Smithurst on the asterisk-dev list)
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2007-02-26 16:54:17 +00:00
Russell Bryant
5873c9c90f
There is no need to look in the iaxs array for the pvt struct when we already
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have a pointer to it.
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2007-02-26 00:37:03 +00:00
Olle Johansson
75d387acbc
Doxygen additions, corrections
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2007-02-24 20:29:41 +00:00
Olle Johansson
e916cf45da
Doxygen updates and corrections
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2007-02-24 19:49:11 +00:00
Olle Johansson
bc01e39174
Creating new doxygen macro "\extref" to create page that lists
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external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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2007-02-24 19:27:50 +00:00
Jason Parker
97ab07a9e8
Allow a Skinny device to monitor a dialplan hint (w00t!).
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See skinny.conf.sample for configuration example.
Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.
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2007-02-24 02:23:43 +00:00
Jason Parker
a23096985b
Support devicestate requests.
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Now you should be able to subscribe to a Skinny device/line.
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2007-02-24 02:06:44 +00:00
Jason Parker
639aee2ec7
Merged revisions 56569 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines
Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected
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2007-02-24 02:04:34 +00:00
Russell Bryant
b85a5ffbe3
Make the hashing function calculate something that makes more sense.
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(Thanks to bmd on #asterisk-dev for pointing out my pointless math).
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2007-02-23 22:26:00 +00:00
Russell Bryant
9fb1404279
Merge team/russell/iax2_performance.
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There is not a large amount of code here and the changes are not very invasive.
However, they should significantly improve performance of chan_iax2 under load.
IAX2 media frames only carry the *source* call number. So, when one arrives,
the correct session that it is a part of has to be matched on IP address, port
number, and call number, instead of just a call number. Had these frames
carried the *destination* call number, this would not be an issue, because that
would be a unique identifier that would make it easy to immediately identify
the correct session.
The way that chan_iax2 did this matching was extremely inefficient. It starts
at the first available call number and traverses each call number sequentially,
locking and unlocking a mutex for each one, to try to match against it. It
would do this regardless of whether the call number was in use or not. So,
for a call with a local call number of 25000, every single incoming media
frame would require a traversal that required 25000 mutex lock and unlock
operations. (Note that the max call number is about 32k).
I have introduced a hash table of active IAX2 calls to improve this lookup
process. The hash is done on the IP address, port number, and call number.
So, for the previous example, a few lock/unlock operations may be done versus
25000 for each frame.
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2007-02-23 21:20:33 +00:00
Russell Bryant
70ce880338
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r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines
Merged revisions 56406 via svnmerge from
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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2007-02-23 20:21:35 +00:00
Joshua Colp
afc99294fa
Merged revisions 56231 via svnmerge from
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r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines
Merged revisions 56230 via svnmerge from
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r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines
Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.
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2007-02-22 18:53:22 +00:00
Olle Johansson
cb0fddc905
Merged revisions 56125 via svnmerge from
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r56125 | oej | 2007-02-22 11:33:55 +0100 (Thu, 22 Feb 2007) | 2 lines
Move message from verbose to debug
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2007-02-22 10:46:09 +00:00
Russell Bryant
a6cbe5d651
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r56055 | russell | 2007-02-21 19:24:10 -0600 (Wed, 21 Feb 2007) | 3 lines
Restructure a little bit of code to reduce nesting. There is no functionality
change here.
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2007-02-22 01:26:22 +00:00
Russell Bryant
80fce39036
Merged revisions 56011 via svnmerge from
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r56011 | russell | 2007-02-21 18:57:36 -0600 (Wed, 21 Feb 2007) | 11 lines
Merged revisions 56010 via svnmerge from
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r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines
If we receive a frame that is not in any of the negotiated formats, then drop
it. (potentially issue #8781 and SPD-12)
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2007-02-22 00:59:17 +00:00
Jason Parker
28a6129af8
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 lines
Fix locking issue, and accept "transport-accept" as a valid accept message.
This should solve issues 8970 and 8503.
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2007-02-21 20:30:54 +00:00
Joshua Colp
cdf9cab49d
Clarify in the doxygen docs abou RFC2833 compensation flag.
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2007-02-21 20:26:43 +00:00
Joshua Colp
93671917f9
Merged revisions 55914 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 lines
Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113)
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2007-02-21 17:23:42 +00:00
Olle Johansson
4534fbb7bc
Merged revisions 55834 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55834 | oej | 2007-02-21 09:32:34 +0100 (Wed, 21 Feb 2007) | 2 lines
Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg)
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2007-02-21 08:39:15 +00:00
Jason Parker
8f28800765
Merged revisions 55799 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 lines
Fix segfault when buddy couldn't be found.
Issue 7764, patch by sailer
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2007-02-21 02:04:10 +00:00
Joshua Colp
22c1925696
Merged revisions 55717 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55717 | file | 2007-02-20 18:57:03 -0500 (Tue, 20 Feb 2007) | 2 lines
Return behavior I removed. I did not remember that you could just add a localnet entry to make it work.
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2007-02-21 00:00:11 +00:00
Joshua Colp
62cb480f28
Merged revisions 55688 via svnmerge from
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r55688 | file | 2007-02-20 18:08:45 -0500 (Tue, 20 Feb 2007) | 2 lines
Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska)
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2007-02-20 23:12:55 +00:00
Joshua Colp
ab96ee237b
Merged revisions 55670 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55670 | file | 2007-02-20 17:47:00 -0500 (Tue, 20 Feb 2007) | 10 lines
Merged revisions 55669 via svnmerge from
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r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines
Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155)
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2007-02-20 22:49:53 +00:00
Jason Parker
ae47fc4541
Merged revisions 55555 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 lines
No need to cast nor free with strdupa (thanks file)
55555!
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2007-02-20 16:56:58 +00:00
Doug Bailey
c6dd460655
Merged revisions 55397 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) | 3 lines
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
Modified module unload process to avoid deadlocks on pthread cancels
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2007-02-19 15:24:46 +00:00
Joshua Colp
8ebf3659aa
Update chan_h323 to new set_rtp_peer definition.
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2007-02-18 19:09:29 +00:00
Kevin P. Fleming
74f31a51bc
add -Wundef to the --enable-dev-mode flags, so that mistyped macro names in #if expressions will be caught
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convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important)
Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated
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2007-02-18 15:03:42 +00:00
Jason Parker
0e5f3cf25e
Merged revisions 55217 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb 2007) | 4 lines
Fix an issue where callerid would not be displayed on some phones.
Issue 8995, initial patch and research done by wedhorn
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2007-02-17 17:11:07 +00:00
Joshua Colp
354c03c4e6
Update chan_jingle to new definition of set_rtp_peer.
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2007-02-17 01:37:29 +00:00
Joshua Colp
977fb01cdd
Merged revisions 55086 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55086 | file | 2007-02-16 20:16:59 -0500 (Fri, 16 Feb 2007) | 10 lines
Merged revisions 55073 via svnmerge from
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r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines
Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)
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2007-02-17 01:22:01 +00:00
Joshua Colp
4331761b5e
Merged revisions 55002 via svnmerge from
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r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, 16 Feb 2007) | 10 lines
Merged revisions 54999 via svnmerge from
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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2007-02-16 22:20:55 +00:00
Olle Johansson
cfb3f84979
Formatting, whitespace fixes
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2007-02-16 14:31:18 +00:00
Olle Johansson
ba32ee49d0
Adding Realtime Text support (T.140) to Asterisk
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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2007-02-16 13:35:44 +00:00
Olle Johansson
84d1cf37fe
Merged revisions 54787 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54787 | oej | 2007-02-16 13:06:23 +0100 (Fri, 16 Feb 2007) | 2 lines
Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted.
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2007-02-16 12:14:53 +00:00
Olle Johansson
fef74f1574
Add callgroup and pickupgroup to SIPPEER function. (thanks ramon)
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2007-02-15 15:52:35 +00:00
Olle Johansson
1f52d1cc81
Issue #7443 - amdtech - Optionally SIP registrations in another
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realtime family.
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2007-02-15 12:10:55 +00:00
Matthew Fredrickson
daa5f504ec
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r54375 | mattf | 2007-02-14 10:56:40 -0600 (Wed, 14 Feb 2007) | 10 lines
Merged revisions 54373 via svnmerge from
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r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines
When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011.
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Olle Johansson
276b570c3e
Issue #9060 - host= parameter in sip.conf stopped working
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caused by outbound proxy patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-14 16:20:48 +00:00
Olle Johansson
0653be0c33
Add port number to SIPPEER dialplan function
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-14 15:27:49 +00:00
Russell Bryant
83856d4683
Merged revisions 54204 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | 5 lines
If we fail to create the SIP socket, then return -1 from reload_config() so
that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
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2007-02-13 21:57:31 +00:00
Russell Bryant
9e99a51802
Merged revisions 54235 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54235 | russell | 2007-02-13 15:31:22 -0600 (Tue, 13 Feb 2007) | 2 lines
Remove a couple of leftover debug messages
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2007-02-13 21:33:03 +00:00
Olle Johansson
1295d40d77
Be careful with debug messages in trunk, they tend to stay around for release....
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2007-02-11 20:49:38 +00:00
Olle Johansson
6e139adc56
Small fix in outbound proxy support.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-11 20:04:49 +00:00
Olle Johansson
32495f91f0
Add support for outbound proxy for peers and [general]
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This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-11 19:42:55 +00:00
Paul Cadach
012e99016c
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53881 | pcadach | 2007-02-10 01:09:49 -0800 (Сбт, 10 Фев 2007) | 1 line
Fix VLDTMF reception
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2007-02-10 09:23:09 +00:00
Russell Bryant
5715b49c30
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
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2007-02-10 00:40:57 +00:00
Russell Bryant
ac4090fce0
add another dependency
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2007-02-10 00:20:57 +00:00
Kevin P. Fleming
e758ef09b3
Merged revisions 53779-53781 via svnmerge from
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r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007) | 2 lines
fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file
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r53780 | kpfleming | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines
add some inter-module dependencies
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r53781 | kpfleming | 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines
another dependency
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2007-02-09 23:53:51 +00:00
Jason Parker
cee4bd43dc
Rename this instance of "busy limit" to "busy level" as well
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2007-02-08 17:19:27 +00:00
Kevin P. Fleming
44c6630e4d
rename busy-limit to busy-level, since it is not a limit
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actually parse the busy-limit option from sip.conf, instead of ignoring it
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2007-02-08 16:41:23 +00:00
Joshua Colp
bd41ac0ae8
Merged revisions 53358 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53358 | file | 2007-02-07 10:43:39 -0500 (Wed, 07 Feb 2007) | 10 lines
Merged revisions 53357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines
Fix a few potential memory leaks with realtime users and peers. (issue #8999 reported by bsmithurst)
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2007-02-07 15:46:30 +00:00
Christian Richter
a96e62d9d2
Merged revisions 53324 via svnmerge from
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r53324 | crichter | 2007-02-07 10:22:44 +0100 (Mi, 07 Feb 2007) | 9 lines
Merged revisions 52843 via svnmerge from
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r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line
fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated)
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2007-02-07 09:51:06 +00:00
Olle Johansson
17af1bd4c8
Merged revisions 53143 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3 lines
Add some comments on queue system behaviour and how it affects the
SIP channel
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2007-02-05 00:30:03 +00:00
Joshua Colp
014feba426
Merged revisions 53138 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines
Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)
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2007-02-03 21:06:36 +00:00
Paul Cadach
3e5e0f5f00
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r53131 | pcadach | 2007-02-03 02:02:55 -0800 (Сбт, 03 Фев 2007) | 1 line
Remove quote from H.323 vendor string because due to compatibilities with
Nortel Meridian CS1000 reported at www.voip-info.org
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2007-02-03 10:12:20 +00:00
Joshua Colp
ce8a7c3d9c
Add onHold value to sip show inuse as well.
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2007-02-02 18:21:46 +00:00
Olle Johansson
cfe66e6b26
Patch based on this patch with small changes for trunk...
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Merged revisions 53109 via svnmerge from
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r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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2007-02-02 00:26:25 +00:00
Joshua Colp
44a9af3576
Merged revisions 53104 via svnmerge from
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r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53103 via svnmerge from
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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.
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2007-02-01 22:26:11 +00:00
Joshua Colp
bf66c620c3
Merged revisions 53097 via svnmerge from
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r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53095 via svnmerge from
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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines
Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113)
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2007-02-01 21:56:23 +00:00
Olle Johansson
544f414c0d
Merged revisions 53085 via svnmerge from
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r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 lines
- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
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2007-02-01 21:17:08 +00:00
Olle Johansson
0b84b386b9
Implementing "busy-limit".
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If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).
If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled.
This affects SIP subscriptions, call queues and manager applications.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 20:43:49 +00:00
Olle Johansson
34eaa61700
Merged revisions 53079 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2 lines
Cleaning up the devicestate callback function
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2007-02-01 20:33:59 +00:00
Olle Johansson
38b87ec4b7
Signal HOLD status to phones that subscribe for status.
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2007-02-01 19:04:47 +00:00
Joshua Colp
e88dda8ca9
Merged revisions 53064 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2 lines
Fix silly logic. We really want to write UDPTL frames out when the call is up.
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2007-02-01 17:42:08 +00:00
Paul Cadach
e5f184ee43
Oops -- Merged revisions 53057 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53057 | pcadach | 2007-02-01 03:07:41 -0800 (Чтв, 01 Фев 2007) | 1 line
chan_h323 is very stable, so let it built by default
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2007-02-01 11:16:00 +00:00
Russell Bryant
b233892198
Merged revisions 53046 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines
Merged revisions 53045 via svnmerge from
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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2007-01-31 21:35:15 +00:00
Russell Bryant
d11f8b7ccd
Merged revisions 52952 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | 5 lines
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO. This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936 )
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2007-01-30 19:36:28 +00:00
Joshua Colp
1fc144435d
Use provided variable for name instead of one in the structure since the structure was just allocated and will be NULL. (issue #8938 reported by st41ker)
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2007-01-30 15:39:09 +00:00
Russell Bryant
116952bba1
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r52763 | russell | 2007-01-29 18:15:50 -0600 (Mon, 29 Jan 2007) | 13 lines
Merged revisions 52762 via svnmerge from
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r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines
Fix the extraction of the timestamp from video frames. It was using the
mapping for a mini-frame instead of a video-frame, which caused it to
get invalid data.
(issue #8795 , mihai)
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2007-01-30 00:16:38 +00:00
Joshua Colp
8e755c25ca
Merged revisions 52370 via svnmerge from
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r52370 | file | 2007-01-26 19:08:18 -0500 (Fri, 26 Jan 2007) | 10 lines
Merged revisions 52360 via svnmerge from
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r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 lines
Make the last context entry read in the dominant one. (issue #8918 reported by pj)
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2007-01-27 00:10:19 +00:00
Joshua Colp
300f980223
Use atomic operation functions for use/ringing/hold manipulation.
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2007-01-25 18:10:18 +00:00
Joshua Colp
b8d6cbcd3f
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r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2 lines
Drop out variables I accidentally put in.
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2007-01-25 17:51:35 +00:00
Joshua Colp
afb9151e19
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r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2 lines
Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42)
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2007-01-25 17:17:56 +00:00
Joshua Colp
2396e24e65
Merged revisions 52016 via svnmerge from
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r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2 lines
Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc)
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2007-01-24 18:04:47 +00:00
Olle Johansson
273f1d78c7
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3 lines
Show capabilities *and* preference in general settings in "sip show settings"
(reported by Clona/Telio - Thanks!)
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2007-01-24 09:42:31 +00:00
Joshua Colp
ee3ab150f6
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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines
Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky)
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2007-01-23 22:59:55 +00:00
Olle Johansson
ef4db783c6
Issue #8817 - Registry corruption when packet retransmits fail. (tootai, patchy by oej)
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2007-01-23 15:36:01 +00:00
Paul Cadach
cd61f69a83
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r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) | 1 line
Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen)
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2007-01-23 06:56:26 +00:00
Joshua Colp
34df128519
Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the RFC so we might want to support it and please don't hurt me Marko ... (issue #7812 reported by drorlb)
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2007-01-23 03:15:04 +00:00
Joshua Colp
5a25c156c6
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r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2 lines
Only change audio formats on the channel if we have an audio format to change to. (issue #8535 reported by ivoc)
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2007-01-23 03:02:09 +00:00
Olle Johansson
1a5dfca2a1
Remove (to quote Rizzo) "useless" variable.
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2007-01-22 19:00:25 +00:00
Russell Bryant
3275357f20
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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2007-01-19 19:09:04 +00:00
Russell Bryant
dcca8f345f
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597 , maybe others...)
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2007-01-19 18:06:03 +00:00
Luigi Rizzo
a52fa7e5ad
sizeof() is compatible with format %d so don't be too
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picky on printf formats.
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2007-01-19 17:45:46 +00:00
Luigi Rizzo
455ea1c99e
remove variable declaration in the middle of a block
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2007-01-19 17:38:28 +00:00
Luigi Rizzo
0a8d59d88a
whoops, fix a cut&paste error...
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2007-01-19 16:35:37 +00:00
Luigi Rizzo
03fc21712a
slight change to the initialization of a structure,
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also using '\0' to make it clear we need a (char)0
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2007-01-19 16:32:05 +00:00
Luigi Rizzo
3fefb9d516
include "asterisk/zapata.h" to get the zaptel headers.
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2007-01-19 16:28:24 +00:00
Luigi Rizzo
dc31eaa025
include "asterisk/zapata.h" instead of looking
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directly for the zaptel.h and tonezone.h
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2007-01-19 16:16:19 +00:00
Dwayne M. Hubbard
18c23101a5
Merged revisions 51274 via svnmerge from
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r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007) | 3 lines
chan_zap compiles without libpri after committing 7877 patch
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2007-01-19 00:28:25 +00:00
Dwayne M. Hubbard
33243c6624
Merged revisions 51272 via svnmerge from
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r51272 | dhubbard | 2007-01-18 17:56:49 -0600 (Thu, 18 Jan 2007) | 11 lines
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r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines
issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file.
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2007-01-19 00:06:35 +00:00
Joshua Colp
1a06a58250
Merged revisions 51243 via svnmerge from
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r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2 lines
Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113)
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2007-01-18 18:39:21 +00:00
Paul Cadach
ae286e1843
Update ast_append_ha() usage
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2007-01-18 06:59:22 +00:00
Russell Bryant
45bd0c4f3f
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r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) | 4 lines
Instead of dividing the offset by 2 directly, make it more clear that the
offset is being scaled by the size of the elements in the buffer.
(Inspired by a discussing on the asterisk-dev list about this code)
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2007-01-17 23:31:40 +00:00
Russell Bryant
c56f9184e8
Merged revisions 51198 via svnmerge from
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r51198 | russell | 2007-01-17 15:18:35 -0600 (Wed, 17 Jan 2007) | 11 lines
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r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | 3 lines
Move the check for a failure of ast_channel_alloc() to before locking the
pvt structure again. Otherwise, on a failure, this will cause a deadlock.
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2007-01-17 21:20:22 +00:00
Joshua Colp
31224a6b52
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r51172 | file | 2007-01-16 19:46:29 -0500 (Tue, 16 Jan 2007) | 2 lines
Move rescheduling of lagrq/pings into the scheduler callback.
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2007-01-17 00:50:59 +00:00
Joshua Colp
04426fab2c
Add support for G729 passthrough with Sigma Designs boards. (issue #8829 reported by ywalther)
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2007-01-16 17:23:31 +00:00
Tilghman Lesher
3d91c0a0c0
IAX2 remote variables - Bug 7619
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2007-01-16 08:38:59 +00:00
Joshua Colp
bf5c607c69
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r51087 | file | 2007-01-16 00:55:23 -0500 (Tue, 16 Jan 2007) | 10 lines
Merged revisions 51085 via svnmerge from
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r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines
Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296 reported by gkloepfer)
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2007-01-16 05:56:42 +00:00
Tilghman Lesher
b3288f79fe
Feature: allow soundcard to be used in both modes (autoanswer and not),
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selectable by how it is called in the dialplan. This allows a speaker
system hooked up to the soundcard to be used for both ring notification,
as well as paging.
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2007-01-15 02:28:32 +00:00
Joshua Colp
91dfb27494
Get rid of unneeded code, fix a spelling mistake, and use registry state a bit more. (issue #8402 reported by rizzo)
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2007-01-13 04:56:25 +00:00
Joshua Colp
033d849bda
Drop trunkrealloc option and just have the maximum size be a configurable option. This is per Kevin's comments on -dev and my own thoughts after I put the previous option in.
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2007-01-13 04:04:04 +00:00
Joshua Colp
0d781f9553
Ensure error variable is set to 0 or else we might get false error messages. (issue #8798 reported by tootai, fix by anthonyl)
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2007-01-13 03:37:10 +00:00
Joshua Colp
c4b4615dcd
Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by marcodmb, branch by anthonyl)
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2007-01-13 03:26:04 +00:00
Joshua Colp
469cf8ab2a
Merged revisions 50468 via svnmerge from
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r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2 lines
Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing.
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2007-01-11 05:54:14 +00:00
Joshua Colp
4942fd94d2
Merged revisions 50466 via svnmerge from
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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines
Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson)
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2007-01-11 05:21:03 +00:00
Joshua Colp
47c515768c
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r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2 lines
Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker)
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2007-01-10 18:34:45 +00:00
Joshua Colp
f215636ef3
Allow usedistinctiveringdetection and distinctiveringaftercid to be reset during a reload. (issue #8739 reported by tzafrir)
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2007-01-10 04:21:46 +00:00
Russell Bryant
444acad95b
Covert some spaces to tabs, and put a list of defines in an enum.
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2007-01-10 02:50:06 +00:00
Olle Johansson
3246ad09a4
Based on the following patch, changed for trunk...
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r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3 lines
- handle re-invites properly in sip_hangup()
- Add some invitestate status changes just to be sure
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2007-01-09 12:25:33 +00:00
Olle Johansson
5d4f272a90
Merged revisions 50006 via svnmerge from
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r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines
Issue #8677 - Handle failure of T.38 re-invite
This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).
To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.
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2007-01-08 14:31:16 +00:00
Olle Johansson
835e2ed684
Merged revisions 49983 via svnmerge from
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r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3 lines
Issue #8524 , support multiple via header values (tardieu)
Thanks!
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2007-01-08 13:34:56 +00:00
Olle Johansson
975dba04bc
Merged revisions 49945 via svnmerge from
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r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2 lines
We only need one forward declaration
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2007-01-08 09:22:53 +00:00
Olle Johansson
7158f48a79
Merged revisions 49925 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2 lines
Issue 8735: Terminate state when extension is unavailable for subscription
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2007-01-08 08:57:35 +00:00
Joshua Colp
585e0e3e09
Merged revisions 49890 via svnmerge from
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r49890 | file | 2007-01-08 00:11:54 -0500 (Mon, 08 Jan 2007) | 10 lines
Merged revisions 49889 via svnmerge from
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r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 lines
Ensure we use the default refresh value of 60 if the remote server does not send one. (issue #8746 reported by maethor)
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2007-01-08 05:13:24 +00:00
Tilghman Lesher
4866a44535
Merged revisions 49831 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007) | 2 lines
Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747)
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2007-01-07 21:26:43 +00:00
Kevin P. Fleming
37182c873e
finish const-ifying ast_func_read()
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2007-01-06 00:13:33 +00:00
Kevin P. Fleming
cd73a483f1
const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way here
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2007-01-05 23:32:42 +00:00
Jason Parker
cdd63e685d
Merged revisions 49705 via svnmerge from
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r49705 | qwell | 2007-01-05 17:16:16 -0600 (Fri, 05 Jan 2007) | 4 lines
Make codec_zap and chan_zap also depend on zaptel.
This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel.
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2007-01-05 23:17:45 +00:00
Kevin P. Fleming
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r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007) | 2 lines
reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases
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2007-01-05 22:43:18 +00:00
Kevin P. Fleming
a8dc50812f
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r49636 | kpfleming | 2007-01-05 11:09:00 -0600 (Fri, 05 Jan 2007) | 10 lines
Merged revisions 49635 via svnmerge from
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r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines
ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly
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2007-01-05 17:10:59 +00:00
Kevin P. Fleming
6efd946f68
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r49600 | kpfleming | 2007-01-04 18:01:40 -0600 (Thu, 04 Jan 2007) | 2 lines
revert the dynamic_list insertion change... that was not the right thing to do
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2007-01-05 00:02:48 +00:00
Kevin P. Fleming
55cc7f9a03
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r49581 | kpfleming | 2007-01-04 17:50:15 -0600 (Thu, 04 Jan 2007) | 3 lines
create the IAX2 processing threads as background threads so they will use smaller stacks
when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it
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2007-01-04 23:54:09 +00:00
Joshua Colp
5a140436f3
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r49568 | file | 2007-01-04 18:00:50 -0500 (Thu, 04 Jan 2007) | 2 lines
It's possible for the iax2 pvt to disappear, so if it has... don't bother looking for dpentries.
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2007-01-04 23:02:32 +00:00
Kevin P. Fleming
c5c17a1434
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r49536 | kpfleming | 2007-01-04 15:58:42 -0600 (Thu, 04 Jan 2007) | 2 lines
don't mark these allocations as 'cache' allocations when caching has been disabled
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2007-01-04 21:59:06 +00:00
Kevin P. Fleming
7f2e9cd62c
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r49523 | kpfleming | 2007-01-04 15:06:02 -0600 (Thu, 04 Jan 2007) | 2 lines
if we're going to decrement the frame count when we free a frame, we should inrement it when we create one :-)
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2007-01-04 21:06:35 +00:00
Kevin P. Fleming
d6d713c84f
Merged revisions 49465 via svnmerge from
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r49465 | kpfleming | 2007-01-04 12:31:55 -0600 (Thu, 04 Jan 2007) | 2 lines
only do IAX2 frame caching for voice and video frames
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2007-01-04 18:32:23 +00:00
Kevin P. Fleming
ec2ae45651
Merged revisions 49457,49460-49461 via svnmerge from
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r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines
make building of codec_gsm against the system GSM library actually work
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r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines
don't define this type either if LOW_MEMORY is enabled
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r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame header caching in the core if LOW_MEMORY is defined
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2007-01-04 18:19:55 +00:00
Matt O'Gorman
c6b43cf285
Merged revisions 49458 via svnmerge from
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r49458 | kpfleming | 2007-01-04 12:06:51 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame caching in LOW_MEMORY mode
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2007-01-04 18:17:24 +00:00
Christian Richter
1fe0e3d192
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r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines
Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line
changed a few debugs to higher debug levels
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r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line
added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that.
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r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict.
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r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line
when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults.
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r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line
when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines
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r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line
added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE.
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r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines
* Added check for bridging in misdn_call to avoid setting echocancellation
when 2 mISDN channels are involved and when bridging is set. That lead
to a kernel panic before under different situations, because we switched
about 2 times between hardware bridging and echocancelation
* readded MISDN_URATE variable which got lost before, this should make app_v110
work again
* fixed typo
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2007-01-03 11:15:02 +00:00
Christian Richter
f4cf6c6339
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r47989 | crichter | 2006-11-24 16:46:13 +0100 (Fr, 24 Nov 2006) | 9 lines
Merged revisions 47968 via svnmerge from
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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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2007-01-03 11:03:57 +00:00
Joshua Colp
1825b70a4c
Merged revisions 49259 via svnmerge from
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r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2 lines
Check pvt structure presence before passing to send_command. This gets rid of the irritating message about a packet without pvt structure. This happens because the scheduled item is getting cancelled at almost the exact moment it is getting executed.
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2007-01-03 01:21:41 +00:00
Olle Johansson
f7ee0abcd2
Removing propably accidentally added debug messages sent to verbose channel
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2007-01-02 20:02:37 +00:00
Olle Johansson
0d340b7d02
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r49212 | oej | 2007-01-02 20:58:45 +0100 (Tue, 02 Jan 2007) | 2 lines
Small cleanup of add_t38sdp - it's always enabled at that point in the code
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2007-01-02 20:01:34 +00:00
Kevin P. Fleming
29058da351
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r49165 | kpfleming | 2007-01-02 07:59:44 -0600 (Tue, 02 Jan 2007) | 2 lines
remove comment that is unrelated to this function
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2007-01-02 14:01:20 +00:00
Kevin P. Fleming
cb97e0c353
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r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) | 2 lines
check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version)
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2007-01-01 23:43:43 +00:00
Olle Johansson
9074555e37
- Add error handling to ast_parse_allow_disallow
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- Use this in chan_sip configuration parsing
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2007-01-01 19:48:31 +00:00
Olle Johansson
f83b845f08
- Implement error handling in ast_append_ha
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- Use this in chan_sip
- Document ha functions in acl.c
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2007-01-01 19:20:46 +00:00
Joshua Colp
7eb0f10f34
count is no longer used in the iaxq structure really so let's just make this a statically declared linked list.
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2006-12-31 19:15:23 +00:00
Olle Johansson
7674260e75
Add version number to useragent string - Issue #8700 , blanchet - THANKS!
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2006-12-31 09:31:23 +00:00
Joshua Colp
ae649fef27
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r49063 | file | 2006-12-29 22:37:22 -0500 (Fri, 29 Dec 2006) | 2 lines
Initialize the packet queue in load_module instead of just declaring the list with the default value. (issue #8695 reported by ssokol)
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2006-12-30 03:40:42 +00:00
Russell Bryant
1f8001e816
Convert various comments to doxygen format.
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2006-12-29 13:25:07 +00:00
Olle Johansson
fc86ea2eed
Removing extra output
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2006-12-29 11:02:28 +00:00
Olle Johansson
b54feaffec
Only include include files once (imported from 1.4)
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2006-12-27 21:27:57 +00:00
Olle Johansson
906a51a1af
Only set "rfc2833compensate" option once
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2006-12-27 20:30:03 +00:00
Olle Johansson
8794577966
Only handle T38 options once
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2006-12-27 20:26:30 +00:00
Olle Johansson
820d82470a
-Remove "localmask" setting (deprecated in earlier version)
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- Remove "musiconhold" and "musicclass" settings (also deprecated earlier)
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2006-12-27 19:57:06 +00:00
Kevin P. Fleming
85b6138d83
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r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006) | 2 lines
make the option actually match the documentation
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2006-12-27 18:34:24 +00:00
Kevin P. Fleming
d68c7c8ce6
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r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006) | 2 lines
allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks
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2006-12-27 18:33:44 +00:00
Olle Johansson
29ed493b40
Be politically correct
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2006-12-27 18:02:10 +00:00
Olle Johansson
da7a35a1cc
Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
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2006-12-27 16:56:11 +00:00
Olle Johansson
3713ce7ed1
Cleanup of handle_common_options
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2006-12-27 16:35:41 +00:00
Olle Johansson
a12dc35eb3
Reset invitestate when sending new invite
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2006-12-27 16:26:02 +00:00
Olle Johansson
d69e9086cc
Issue #8600 - bogus SDP Content Length in T.38 re-invite
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2006-12-27 16:01:07 +00:00
Joshua Colp
7f61b822c1
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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2006-12-26 04:34:07 +00:00
Luigi Rizzo
09f75aa6dc
rename the structs struct tone_zone_sound and struct tone_zone
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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2006-12-25 06:38:09 +00:00
Russell Bryant
ad8a480085
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r48948 | russell | 2006-12-24 16:19:37 -0500 (Sun, 24 Dec 2006) | 3 lines
Fix a typo in an error message that indicated that the MGCP channel type could
not be registered, instead of the correct type, OSS.
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2006-12-24 21:21:21 +00:00
Russell Bryant
539514d793
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r48944 | russell | 2006-12-24 02:25:38 -0500 (Sun, 24 Dec 2006) | 11 lines
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r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | 3 lines
Check for the proper return value on an error in a call to mmap().
This was reported by Andy Wang on the asterisk-dev list. Thanks!
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2006-12-24 07:27:29 +00:00
Russell Bryant
e0f55d2385
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r48940 | russell | 2006-12-24 01:49:31 -0500 (Sun, 24 Dec 2006) | 11 lines
Merged revisions 48939 via svnmerge from
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r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines
Remove a couple of misplaced dots in log messages. This was reported by
Andrea Spadaccini on the asterisk-dev mailing list.
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2006-12-24 06:51:37 +00:00
Jason Parker
a01ae565e9
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r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec 2006) | 2 lines
Note to self: Run make before committing...
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2006-12-22 21:40:56 +00:00
Jason Parker
9883943ec5
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r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec 2006) | 2 lines
Fix for issue 7774 - patch by alamantia
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2006-12-22 20:43:38 +00:00
Steve Murphy
7b338004bf
a quick fix to app_sms.c to get rid of cursed compiler warnings so I can compile under --enable-dev-mode
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2006-12-21 19:44:20 +00:00
Steve Murphy
9327720c37
As per bug 7978, this version introduces the jittertargetextra option in config files
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2006-12-21 00:24:08 +00:00
Kevin P. Fleming
fa5931d0c3
Merged revisions 48586 via svnmerge from
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r48586 | kpfleming | 2006-12-19 15:28:04 -0600 (Tue, 19 Dec 2006) | 2 lines
suppress compiler warnings in this module until it can be improved
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2006-12-19 21:28:43 +00:00
Joshua Colp
381d49c9af
Clean up find_idle_thread function and use atomic operations for dynamic thread count.
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2006-12-18 17:30:53 +00:00
Joshua Colp
c17181aae3
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r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2 lines
Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia)
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2006-12-18 17:18:18 +00:00
Luigi Rizzo
d5eef300d8
define a mask SIP_INSECURE sam as for other sets of flags.
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2006-12-17 12:47:08 +00:00
Joshua Colp
6bdd1a3388
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r48504 | file | 2006-12-15 14:38:51 -0500 (Fri, 15 Dec 2006) | 2 lines
Hold call structure lock in places where a qualify or peer action can destroy it.
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2006-12-15 19:40:29 +00:00
Joshua Colp
12565be69e
Merged revisions 48502 via svnmerge from
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r48502 | file | 2006-12-15 14:24:15 -0500 (Fri, 15 Dec 2006) | 2 lines
Lock network retransmission queue in all places that it is used.
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2006-12-15 19:26:45 +00:00
Luigi Rizzo
7db228ac95
replace ast_safe_string_alloc() with asprintf()
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2006-12-15 15:40:57 +00:00
Luigi Rizzo
a736605bf4
replace ast_safe_string_alloc() with asprintf()
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2006-12-15 15:39:29 +00:00
Olle Johansson
168deb2366
Issue #8592 - treat 504 as congestion (imported from 1.2/1.4)
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2006-12-15 10:58:36 +00:00
Olle Johansson
bf2172faac
Update to latest IANA specs
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2006-12-15 10:54:06 +00:00
Joshua Colp
315f8bb1b2
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r48478 | file | 2006-12-15 01:28:05 -0500 (Fri, 15 Dec 2006) | 2 lines
Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted.
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2006-12-15 06:31:26 +00:00
Russell Bryant
52396dff26
revert check for a zaptel transcoder related definition that was done in the
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wrong module.
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2006-12-13 00:56:30 +00:00
Russell Bryant
666d526aad
Fix various spelling mistakes in comments.
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2006-12-12 22:32:20 +00:00
Russell Bryant
4490b5710d
Make chan_zap inform you that your version of zaptel is too old instead of
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just failing to compile.
It seems like the proper way to do this would be in the configure script.
However, that wouldn't help existing checkouts unless we forced the configure
script to be executed after any code was changed.
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2006-12-12 22:27:22 +00:00
Steve Murphy
23e606f70a
Merged revisions 48372 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48372 | murf | 2006-12-09 20:04:18 -0700 (Sat, 09 Dec 2006) | 9 lines
Merged revisions 48371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line
This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels).
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2006-12-10 03:14:27 +00:00
Russell Bryant
a9415d4b8e
convert the thread IO state and type to use enums.
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2006-12-09 16:44:41 +00:00
Russell Bryant
86ad1435c1
Merged revisions 48363 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) | 8 lines
Use locking when accessing the registrations list. This list is not actually
used very often, so the likelihood of there being a problem is pretty small,
but still possible. For example, if the CLI command to list the registrations
was called at the same time that a reload was occurring and the registrations
list was getting destroyed and rebuilt, a crash could occur.
In passing, go ahead and convert this list to use the linked list macros.
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2006-12-09 16:04:06 +00:00
Russell Bryant
62c5be4fda
chan_iax2 has an extremely large function, socket_process(), to handle incoming
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frames. The function, before this commit, was roughly 1400 lines long. So, I
am working on breaking this up into functions so that the code is easier to
follow and debug. Also, I will be committing these changes in chunks as I do
them to ease tracking down any potentially introduced problems.
Break out roughly 150 lines from socket_process() and introduce a new function,
socket_process_meta() which handles the parsing of an incoming meta frame.
Also, restructure some of this code a bit to reduce the deep nesting that was
in this code.
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2006-12-09 07:10:55 +00:00
Russell Bryant
6592c25a69
- Fix a few spelling mistakes
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- Use sizeof() to pass an array size to a function
- Use a single bit for a variable in the chan_iax2_pvt stuct since that is all
it needs.
- Add some comments about the iaxs, iaxl, and lastused arrays.
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2006-12-09 05:04:38 +00:00
Olle Johansson
2b42422940
Handle multiple 487's correctly
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2006-12-06 16:54:27 +00:00
Olle Johansson
c39f887ac8
Don't send Contact in SIP Messages (imported from 1.2/1.4).
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Reported by Gunnar at Omnitor.
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2006-12-06 12:34:58 +00:00
Russell Bryant
ec8591d04c
Constify a bunch of usage strings for CLI commands.
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2006-12-06 07:35:31 +00:00
Russell Bryant
a44e55a3f3
Constify a bunch of the usage strings for CLI commands.
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2006-12-06 07:23:32 +00:00
Russell Bryant
b761fb83cd
Instead of creating an unused instance of an unnamed enum, give it a name.
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2006-12-06 07:16:09 +00:00
Olle Johansson
22142e1aa0
Cleaning up handle_response a bit. (Imported from 1.4)
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2006-12-02 22:03:14 +00:00
Olle Johansson
df8c773783
Send CANCEL to call with early media (PROGRESS INBAND).
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This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
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2006-12-02 20:30:58 +00:00
Olle Johansson
fc6de9e982
Invitestate updates
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2006-12-02 19:11:02 +00:00
Olle Johansson
c23bc46089
- Disable RTP timeouts during T.38 transmission
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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2006-12-02 12:05:40 +00:00
Olle Johansson
4ce5b7c080
- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4)
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- Small fixes to limitonpeer
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2006-12-01 18:16:16 +00:00
Joshua Colp
869101028b
Merged revisions 48168 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
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 21:22:01 +00:00
Olle Johansson
813f8d26fe
Issue #8319 (imported from 1.2, 1.4) - Increment nonce-count properly (noriyuki)
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2006-11-30 20:55:43 +00:00
Joshua Colp
a77935b674
Merged revisions 48158 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48158 | file | 2006-11-30 15:07:55 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48157 via svnmerge from
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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 20:09:46 +00:00
Olle Johansson
389ed67b41
Merging patch from 1.2/1.4. I think this was originally spotted by
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Luigi, but hit me in the back today.
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2006-11-30 09:23:25 +00:00
Joshua Colp
bf9a5e1dee
I am pretty sure that oej only meant to change the variable name in the source, not the configuration option name so let's turn it back to srvlookup instead of global_srvlookup. (issue #8442 reported by jtodd)
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2006-11-30 03:29:42 +00:00
Russell Bryant
c16c606cff
Add a comment to note near some code that performs a very expensive operation
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that occurs for every incoming media frame.
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2006-11-28 23:13:37 +00:00
Joshua Colp
e021f5fbb9
Merged revisions 48088 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48088 | file | 2006-11-28 11:57:16 -0500 (Tue, 28 Nov 2006) | 10 lines
Merged revisions 48087 via svnmerge from
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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 16:59:20 +00:00
Olle Johansson
42fb134171
Change error message (imported from 1.4)
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2006-11-27 06:44:40 +00:00
Matthew Fredrickson
2102534748
Little fix so we use the right message
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2006-11-25 21:50:42 +00:00
Matthew Fredrickson
94ec7877b5
Make compiler happier
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2006-11-25 21:46:58 +00:00
Matthew Fredrickson
8b8187c6f8
bug fix
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2006-11-25 21:35:23 +00:00
Matthew Fredrickson
1ded1fd55a
Make sure we don't send a group reset on a group larger than 32 CICs
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2006-11-25 21:32:33 +00:00
Matthew Fredrickson
cb06f87d11
Add ss7 show linkset command
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2006-11-25 20:54:38 +00:00
Matthew Fredrickson
4dfd954ac3
Updates to show linkset command
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2006-11-25 20:30:04 +00:00
Olle Johansson
e46e42ceec
Changing ERROR to lesser level. Imported from 1.2/1.4
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2006-11-25 09:48:30 +00:00
Olle Johansson
a4cc4d03c9
Remove unused memory allocation
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2006-11-23 11:04:25 +00:00
Olle Johansson
db08a0f346
Don't over-deprecate... :-)
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2006-11-22 11:28:12 +00:00
Olle Johansson
4221020191
Treat 101 as 100, not 183 session progress
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2006-11-21 15:25:38 +00:00
Olle Johansson
230376c7c0
Properly reset schedule items (rizzo)
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2006-11-20 11:36:45 +00:00
Luigi Rizzo
f14aa70947
Merged revisions 47823 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5 lines
fix bug 7450 - Parsing fails if From header contains angle brackets
(the bug was only in a corner case where the < was right after the
opening quote, and the fix is trivial).
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2006-11-18 18:03:42 +00:00
Luigi Rizzo
34ec231da7
prevent the sound thread from consuming all the available CPU
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doing busy-wait on the output audio device.
As it is set now, it tries to push a frame every 10ms,
which is still too frequent but avoids deep restructuring
of the code (which i should do, though).
Note, this is only for ring tones, regular audio coming
from the network is still delivered as soon as it is
available.
Eventually this could well end up in the 1.4 branch, but
since i am probably the only user of chan_oss there isn't
much urgency to do that.
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2006-11-18 08:19:41 +00:00
Matthew Fredrickson
03ee553343
Add some sense of link state. Don't make calls if link is down. Only reset if we're bringing the
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link up for the first time.
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2006-11-17 18:26:33 +00:00
Luigi Rizzo
b55e30b075
remove an unused function
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2006-11-17 11:31:55 +00:00
Luigi Rizzo
779f4c0e0a
use the regexp cli support on some of the command
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2006-11-17 11:15:54 +00:00
Matthew Fredrickson
57d4919427
Make sure we choose the last channel as the dchannel if it's not E1 (for BRI). ( #8077 ) Thanks Tzafrir.
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2006-11-17 01:05:28 +00:00
Luigi Rizzo
24c241c5a9
convert two entries to new style
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2006-11-16 23:05:50 +00:00
Jason Parker
65fb29bc03
Remove the interim variable for range modifications, and set it on the structure directly.
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Also move the default checking to where it gets set initially.
Fixes suggested by file.
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2006-11-16 22:51:00 +00:00
Luigi Rizzo
2c9643d49f
convert some handlers to new style.
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2006-11-16 22:44:29 +00:00
Jason Parker
bfd630682e
Add ability to modify range for dring matching.
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Issue #8369 , patch by ssuehring, modified slightly by me.
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2006-11-16 22:32:23 +00:00
Luigi Rizzo
a63d971afc
fix indentation
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2006-11-16 22:03:27 +00:00
Joshua Colp
cce2057f08
Merged revisions 47764 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47764 | file | 2006-11-16 16:11:06 -0500 (Thu, 16 Nov 2006) | 2 lines
Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath)
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2006-11-16 21:13:17 +00:00
Olle Johansson
a6f5adefa1
Make it possible to enable/disable onhold tracking, in order to make life easier
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for realtime users.
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2006-11-16 19:29:28 +00:00
Joshua Colp
f781673614
Merged revisions 47751 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47751 | file | 2006-11-16 13:29:12 -0500 (Thu, 16 Nov 2006) | 10 lines
Merged revisions 47750 via svnmerge from
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r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 lines
Because of the way chan_local is written we should be extra careful and make sure our callback functions have a tech_pvt. (issue #8275 reported by mflorell)
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2006-11-16 18:32:27 +00:00
Olle Johansson
638d5e1fc8
Merging implementation of invite states from my "invitestate" branch for 1.2. This is a bit more
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clean platform for the handling of BYE/CANCEL than what we had. It might also need to changes
in other parts of the code, since we know the state of the INVITE transaction.
Please observe that this is is not dialog states at all, this is INVITE transaction states.
Hello Michael Proctor, and thank you! :-)
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2006-11-16 16:02:41 +00:00
Olle Johansson
a427a2a89a
- CANCEL never uses authentication
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- Add docs on canreinvite
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2006-11-16 15:12:30 +00:00
Joshua Colp
176c7ca47c
Merged revisions 47712 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47712 | file | 2006-11-15 17:31:17 -0500 (Wed, 15 Nov 2006) | 10 lines
Merged revisions 47711 via svnmerge from
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r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 lines
Make sure that the pvt structure exists before trying to do fixup on Local channels. (issue #7937 reported by mada123, fix by alamantia with mods by me)
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2006-11-15 22:32:51 +00:00
Olle Johansson
5231f5aff8
Hunting the initreq change for an ACK
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2006-11-15 21:21:21 +00:00
Olle Johansson
015efdd558
- Don't reply to ACK
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- Improve SIP history for debugging
(Imported from 1.4)
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2006-11-15 20:40:22 +00:00
Luigi Rizzo
7dab91258a
fix indentation
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2006-11-15 17:40:09 +00:00
Olle Johansson
88a518c2dd
Send proper SIP error message improperly when we can't allocate dialog (out of file handles is one cause)
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2006-11-15 16:04:24 +00:00
Olle Johansson
5ef820e53c
Update doxygen docs to reflect the code...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-15 15:27:46 +00:00
Kevin P. Fleming
4d3ae066d6
Merged revisions 47635 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47635 | kpfleming | 2006-11-14 18:05:44 -0600 (Tue, 14 Nov 2006) | 2 lines
silence compiler warning on 64-bit platforms (this variable is an 'int' anyway, comparing it to 'signed long' is not useful)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-15 00:06:37 +00:00
Joshua Colp
676a0f2719
Merged revisions 47628 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47628 | file | 2006-11-14 17:05:03 -0500 (Tue, 14 Nov 2006) | 2 lines
Only keep the video RTP structure around if 1. Video support is enabled and 2. A video codec is enabled on the dialog
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-14 22:07:20 +00:00
Olle Johansson
063ebdf279
Adding some debug output to trace bug in realtime
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-14 13:17:38 +00:00
Olle Johansson
f9f8237a2f
Adding a new debug line for issue #7351 - trying to find where an ACK can overwrite the initreq...
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2006-11-14 12:32:46 +00:00
Olle Johansson
c2b4c7a9c7
Issue #8272 imported from 1.2/1.4 - Let the peerpoke system destroy it's own packets, please.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-14 11:26:06 +00:00
Olle Johansson
fb211aa610
Remove flags not used any more (thanks Luigi)
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2006-11-14 11:00:33 +00:00
Tilghman Lesher
98e17e1cb9
Merged revisions 47581 via svnmerge from
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r47581 | tilghman | 2006-11-13 14:20:01 -0600 (Mon, 13 Nov 2006) | 10 lines
Merged revisions 47580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) | 2 lines
Having more than 255 old messages caused corruption in the new/old count
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2006-11-13 20:21:33 +00:00
Olle Johansson
ea64cd393f
Small fix for uncommon scenario.
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2006-11-13 19:20:54 +00:00
Joshua Colp
24afed6a81
Make chan_h323 build again and make the CLI commands work. (reported on asterisk-dev mailing list by Di-Shi Sun)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-13 19:14:39 +00:00
Olle Johansson
ffc1dc331a
Only produce error message once, don't fill the screen with them...
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(Testing SIPP thanks to JerJer and Greg)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-13 16:08:12 +00:00
Luigi Rizzo
52148c96f5
merge from astobj2-r47450: use UNLINK to remove a packet from its queue,
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and related code rearrangement.
Approved by: oej
This could be made better if we declared
struct sip_pvt *dialpg = pkt->owner;
at the beginning of the function, and use it throughout the function.
I'll let the boss decide :)
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2006-11-13 14:29:31 +00:00
Luigi Rizzo
c26ccb97b9
merge from codename-pineapple and astobj2 47499:
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simplify __sip_ack() removing a strcmp for looking up packets.
no functional change, only performance, so don't need to merging
to earlier branches now.
Approved By: oej
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2006-11-13 14:14:54 +00:00
Tilghman Lesher
8dbe2063ed
Merged revisions 47523 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47523 | tilghman | 2006-11-12 19:12:01 -0600 (Sun, 12 Nov 2006) | 10 lines
Merged revisions 47522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) | 2 lines
Don't play dialtone if the seizing the channel fails (Bug 7754)
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2006-11-13 01:16:30 +00:00
Olle Johansson
139bf2b626
Part of patch in #7403 to improve tag checking in pedantic mode (stephen_dredge)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-12 20:47:25 +00:00
Russell Bryant
462407a3a6
The use of an ifdef to check for FreeBSD is useless here because the two
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versions of the format string are identical. Also, since each format is only
used once, get rid of the use of defines all together. (issue #8344 , julieng)
In passing, also clean up the formatting a but to get rid of the nesting
without the use of braces, as defined in the coding guidelines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-12 19:22:19 +00:00
Olle Johansson
2286ff9659
Restore auto-framing (DEA). Imported from 1.4
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2006-11-12 16:15:49 +00:00
Olle Johansson
aa275d4970
- Support UDPTL as well as udptl in T38 sdp.
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2006-11-12 15:40:03 +00:00
Olle Johansson
b744a76565
- Don't hangup because of failed re-invite. Go back to previous state.
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- Keep RTP running during T.38 session
We might improve the code to issue ast_rtp_stop if T.38 re-invite not fails
later on in the code, but I don't see many reasons to.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-12 15:30:48 +00:00
Olle Johansson
1f9dcf758c
- Add some comments to t.38 code
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- Remove improper blocking of ptime: in SDP
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2006-11-12 15:14:35 +00:00
Russell Bryant
e19fc06e2c
Merged revisions 47497 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47497 | russell | 2006-11-12 01:23:23 -0500 (Sun, 12 Nov 2006) | 12 lines
Merged revisions 47496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | 4 lines
Only do the check to determine whether the channel calling this function is an
IAX2 channel when getting the IP address using the special argument,
CURRENTCHANNEL. (issue #8341 , jcovert)
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2006-11-12 06:31:39 +00:00
Matthew Fredrickson
69d3416c12
Make sure we don't use 32bits for a value that only requires 1 bit. Also, fix a compiler warning for one of the SS7 functions.
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2006-11-10 22:06:11 +00:00
Olle Johansson
0c18d91817
Add some history and fix some debug output for autodestruct.
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2006-11-10 21:55:31 +00:00
Olle Johansson
2b6d3952e1
Proper fix for adding debug...
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2006-11-10 21:39:58 +00:00
Olle Johansson
d5243cc7aa
Make sure we destroy dialog in case of loop
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 20:31:57 +00:00
Olle Johansson
d69eb90f82
Cleanup imported from 1.4
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2006-11-10 20:25:32 +00:00
Joshua Colp
c6e8dbf189
Merged revisions 47457 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47457 | file | 2006-11-10 14:36:25 -0500 (Fri, 10 Nov 2006) | 2 lines
Let's give this a go...
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2006-11-10 19:40:01 +00:00
Matthew Fredrickson
554902f070
Add fix for 7321. Ability to hide calleridname from zapata.conf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 19:35:41 +00:00
Olle Johansson
d53979523c
Issue 8336- fix support for multipart SDP (imported from 1.2/1.4). (Alphaque)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 19:01:12 +00:00
Tilghman Lesher
691363656f
Merged revisions 47436 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47436 | tilghman | 2006-11-10 10:51:55 -0600 (Fri, 10 Nov 2006) | 2 lines
Discussion of these CLI changes resulted in more consistency (Bug 8236)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 17:01:06 +00:00
Olle Johansson
2c5bbdc938
Ripping out bad support for 491 replies to INVITE's. Let's try again properly later.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 13:14:44 +00:00
Olle Johansson
088d1f0def
Fix badly defined flag. Thanks fenlander!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 12:22:44 +00:00
Olle Johansson
7a3528c4a9
Small simplification and documentation correction.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 12:11:06 +00:00
Joshua Colp
7142180792
Merged revisions 47405 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47405 | file | 2006-11-09 22:44:36 -0500 (Thu, 09 Nov 2006) | 2 lines
Fix building of chan_h323 by completeing some structure definitions. (issue #8327 reported by Mithraen)
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2006-11-10 03:45:59 +00:00
Russell Bryant
36793b730c
Merged revisions 47391 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | 7 lines
Work around an issue that caused menuselect to display a bogus description for
app_voicemail and chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description. However, the way
we extract this information from the source files for menuselect is not smart
enough to figure this out.
(issue #8326 , #8328 )
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2006-11-09 21:32:27 +00:00
Joshua Colp
ead6c4655e
Merged revisions 47380 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, 09 Nov 2006) | 10 lines
Merged revisions 47379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines
Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods.
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2006-11-09 17:08:29 +00:00
Steve Murphy
2907433599
committed in behalf of bug 8190
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 22:52:01 +00:00
Olle Johansson
bbcd0d0450
- Add Max-Forwards higher in the packet, following recommendations
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- Fix documentation for sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf !!!
- Change doc for a sip_pvt setting
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2006-11-08 22:35:22 +00:00
Kevin P. Fleming
02c6f507bb
Merged revisions 47333 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47333 | kpfleming | 2006-11-08 12:07:16 -0600 (Wed, 08 Nov 2006) | 2 lines
add simple fix for SDP to report proper sample rate for G.722 media sessions
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 18:28:53 +00:00
Kevin P. Fleming
e108453e02
coding guidelines, coding guidelines, coding guidelines
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2006-11-08 15:28:39 +00:00
Luigi Rizzo
5e61077632
merge from team/rizzo/astobj2 rev.47271
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avoid doing p > 0 when p is a pointer;
move a lock closer to the place where it is needed
Approved By: oej
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2006-11-08 13:59:32 +00:00
Luigi Rizzo
d5b101e974
merge from team/rizzo/astobj2 rev.47246
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Same as for peers and users, replace ASTOBJ_UNREF(r, sip_registry_destroy)
with unref_registry(r);
Approved By: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 13:53:01 +00:00
Luigi Rizzo
09012d5f20
merge from team/rizzo/astobj2, rev 47243, 47244, 47245:
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Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with unref_peer(peer);
This places the name of the destructor in one place only (where it
should be), eliminates the chance of errors in case you specify the wrong
destructor, and also lets the compiler do type checking on the argument,
again helping with keeping the code clean.
Same for users.
remove two duplicate definitions.
Approved By: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 13:49:25 +00:00
Luigi Rizzo
24dfbe955e
merge rev.47224 from team/rizzo/astobj2:
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hide dialoglist lock/unlocking in wrapper functions.
Approved By: oej
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2006-11-08 13:44:27 +00:00
Luigi Rizzo
f58302fc9c
silence compiler about uninitialized variables.
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The compiler is wrong, but it has the last word.
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2006-11-08 13:43:06 +00:00
Olle Johansson
e644ba34c5
Destroy dialog properly at unload (rizzo)
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2006-11-08 08:01:44 +00:00
Luigi Rizzo
c8597704ce
fix compilation.
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Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.
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2006-11-08 07:21:45 +00:00
Steve Murphy
908f176cf3
A fair number of changes for the sake of bug 7506
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2006-11-07 21:47:49 +00:00
Joshua Colp
7659f6d524
Merged revisions 47287 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov 2006) | 2 lines
This is not the commit you are looking for...
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2006-11-07 20:16:02 +00:00
Joshua Colp
66d3792f11
Merged revisions 47284 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov 2006) | 2 lines
Make MOH work as it did before in chan_local, without this then it can go funky when transfers and MOH are involved. (issue #7671 reported by jmls)
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2006-11-07 20:12:30 +00:00
Olle Johansson
b557aeda38
Break -> continue to make stuff work... Thanks, Luigi!
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2006-11-07 17:33:24 +00:00
Olle Johansson
20af2870d9
issue #8265 - don't reply to ACK. Imported from 1.2, 1.4
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2006-11-07 13:26:31 +00:00
Olle Johansson
7d368caa98
Don't lock dialoglist if monitor runs __sip_destroy.
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Hmmm. I did not change pbx_dundi and yet it doesn't compile ;-)
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2006-11-04 22:16:57 +00:00
Olle Johansson
80293c2ed8
Move IP address selection for media out of add_sdp
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2006-11-04 21:48:02 +00:00
Olle Johansson
18b694d318
Make srvlookup global_srvlookup to follow the rest of the code
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2006-11-04 21:17:40 +00:00
Olle Johansson
fe5440b7c8
Simplify previous patch
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2006-11-04 19:20:30 +00:00
Olle Johansson
d900b47ccf
Adding new config option "limitpeersonly" to only apply call limits
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to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
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2006-11-04 19:13:30 +00:00
Olle Johansson
338c18486c
Importing patch for Invite/replaces from 1.4
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-04 18:40:51 +00:00
Olle Johansson
37e39111c5
Reverting rev 47093 until we have an agreement on how to
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implement this, if at all.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-04 17:45:47 +00:00
Steve Murphy
65beb9dc10
This fix introduced via bug 8233
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 23:27:58 +00:00
Luigi Rizzo
4a545a7662
remove old/useless usecnt stuff
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 18:17:08 +00:00
Luigi Rizzo
ec1525ac88
remove old/useless usecnt stuff.
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I think this module doesn't compile, anyways, because
it has not been updated to the new module interface.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 18:16:20 +00:00
Luigi Rizzo
b98a7bdaea
Save the 'From' header received in a REGISTER message so we can show it
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e.g. in the Manager interface. This information is available as
a callerid (or something like that) during a call, but not when a
device is registered but silent.
It may be useful to have it available e.g. when developing a user
interface/operator panel, to map numbers to names.
experimental, so not committed to 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 16:59:52 +00:00
Luigi Rizzo
39d94767d7
remove useless usecnt stuff
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 12:24:08 +00:00
Luigi Rizzo
74752c341c
remove useless usecnt stuff
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 12:22:24 +00:00
Luigi Rizzo
af014a5076
remove useless usecnt stuff
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 12:21:40 +00:00
Luigi Rizzo
4b70ae927c
remove useless usecnt stuff
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 12:19:22 +00:00
Luigi Rizzo
5397441866
remove useless usecnt handling
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 12:15:13 +00:00
Luigi Rizzo
d5ea99c868
remove useless usecnt handling code
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-03 12:14:00 +00:00
Tilghman Lesher
79f75ec09a
Merged revisions 47053 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006) | 2 lines
More changes making the CLI more consistent with "category verb arguments" (continuation of issue 8236)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 23:55:59 +00:00
Tilghman Lesher
10875731ec
Merged revisions 47051 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines
Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 23:16:09 +00:00
Matthew Fredrickson
eee1b9e1fb
Updates to do unblock correctly
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 21:34:03 +00:00
Olle Johansson
9dffcc2e75
Move check for codec translators to an earlier place in the call, so we can fail gracefully (imported from 1.4)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 20:24:10 +00:00
Olle Johansson
0df2c7a774
Disable code for not implemented functionality (T38 over RTP/TCP)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46999 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 19:01:02 +00:00
Matthew Fredrickson
d810d52606
Changes to show blocked/unblocked states, as well as in service, out of service state
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 16:50:55 +00:00
Kevin P. Fleming
30b47fb8a0
Merged revisions 46937 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46937 | kpfleming | 2006-11-02 10:45:32 -0600 (Thu, 02 Nov 2006) | 2 lines
don't send INVITE when we have determined that we can't offer any audio formats due to lack of trancoding support (or incorrect configuration)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 16:45:50 +00:00
Joshua Colp
946eebf16f
Merged revisions 46930 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46930 | file | 2006-11-02 11:06:39 -0500 (Thu, 02 Nov 2006) | 10 lines
Merged revisions 46920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 lines
Repeat after me oej: I will at least make sure my code compiles before I commit it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 16:14:03 +00:00
Matthew Fredrickson
040e112ebd
Add simple down event support
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 16:03:12 +00:00
Nadi Sarrar
a2cb7d974e
find_free_chan_in_stack: cleanup buggy usage
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 15:47:23 +00:00
Olle Johansson
5a87a3bbb2
Don't overwrite pkt->flags (imported from 1.2/1.4)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-02 15:31:53 +00:00
Matt O'Gorman
09ef9b465b
Merged revisions 46822 via svnmerge from
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https://svn.digium.com/svn/asterisk/branches/1.4
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r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006) | 2 lines
bind address support from bug 8164
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-01 20:38:05 +00:00
Joshua Colp
9f78577f5e
Merged revisions 46775 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2 lines
It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-01 18:26:12 +00:00
Russell Bryant
190cff5969
Merged revisions 46744 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46744 | russell | 2006-11-01 11:39:09 -0500 (Wed, 01 Nov 2006) | 2 lines
Prevent an infinite loop when config processing gets to a jitterbuffer option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-01 16:51:26 +00:00
Luigi Rizzo
bf3afe5641
remove old/useless usecount handling
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31 18:10:21 +00:00
Luigi Rizzo
e72c0db882
remove old/useless usecount stuff.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31 18:04:19 +00:00
Luigi Rizzo
fe28cd213b
remove old/useless usecount management code.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31 17:58:49 +00:00
Olle Johansson
602cab138b
Set #define for TIMER T1 value
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46650 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31 13:56:33 +00:00
Olle Johansson
9572ee68c2
Cleaning up code
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31 13:51:30 +00:00
Olle Johansson
b136baaff4
Fix rport handling.
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...where did the 1.2 properties come from, really? they're back.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31 10:29:24 +00:00
Olle Johansson
97be2ac76f
- If peer that register fails ACL, fail him
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- Remove the 1.2 props I've set by mistake earlier
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-31 09:24:55 +00:00