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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
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
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
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
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Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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2007-04-27 14:40:28 +00:00
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r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines
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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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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
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
34469a8707
added CLI 'sip unregister <peer>' for issue 9326. thanks eliel
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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
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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
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
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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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r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines
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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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r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines
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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
9fff461080
Remove duplicate prototype declaration. (issue #9517 reported by junky)
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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
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r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines
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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
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
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
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
5bc2aa8ab1
Use the same parameter to the two "Registry" AMI events - ChannelDriver
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2007-04-06 19:26:01 +00:00
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r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, 05 Apr 2007) | 10 lines
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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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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
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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
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r59939 | russell | 2007-04-03 14:16:53 -0500 (Tue, 03 Apr 2007) | 12 lines
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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
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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
b908f9717a
Remove a duplicate function prototype. (issue #9444 , junky)
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2007-04-03 18:34:14 +00:00
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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
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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
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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
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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
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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
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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
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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
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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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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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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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
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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
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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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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
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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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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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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
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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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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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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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r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines
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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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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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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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r56011 | russell | 2007-02-21 18:57:36 -0600 (Wed, 21 Feb 2007) | 11 lines
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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
cdf9cab49d
Clarify in the doxygen docs abou RFC2833 compensation flag.
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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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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
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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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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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r55086 | file | 2007-02-16 20:16:59 -0500 (Fri, 16 Feb 2007) | 10 lines
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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
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Formatting, whitespace fixes
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2007-02-16 14:31:18 +00:00
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
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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
fef74f1574
Add callgroup and pickupgroup to SIPPEER function. (thanks ramon)
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2007-02-15 15:52:35 +00:00
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
276b570c3e
Issue #9060 - host= parameter in sip.conf stopped working
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caused by outbound proxy patch.
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2007-02-14 16:20:48 +00:00
0653be0c33
Add port number to SIPPEER dialplan function
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2007-02-14 15:27:49 +00:00
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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
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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
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
6e139adc56
Small fix in outbound proxy support.
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2007-02-11 20:04:49 +00:00
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.
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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
cee4bd43dc
Rename this instance of "busy limit" to "busy level" as well
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2007-02-08 17:19:27 +00:00
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
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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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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
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Add onHold value to sip show inuse as well.
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2007-02-02 18:21:46 +00:00
cfe66e6b26
Patch based on this patch with small changes for trunk...
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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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r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines
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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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r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines
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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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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
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.
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2007-02-01 20:43:49 +00:00
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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
38b87ec4b7
Signal HOLD status to phones that subscribe for status.
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2007-02-01 19:04:47 +00:00
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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
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r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines
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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
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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
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
300f980223
Use atomic operation functions for use/ringing/hold manipulation.
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2007-01-25 18:10:18 +00:00
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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
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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
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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
273f1d78c7
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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
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
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
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
1a5dfca2a1
Remove (to quote Rizzo) "useless" variable.
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2007-01-22 19:00:25 +00:00
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
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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
c56f9184e8
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r51198 | russell | 2007-01-17 15:18:35 -0600 (Wed, 17 Jan 2007) | 11 lines
Merged revisions 51197 via svnmerge from
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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
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
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
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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
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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
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
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
5d4f272a90
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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
835e2ed684
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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
975dba04bc
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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
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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
4866a44535
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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
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
87b9abc892
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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
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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
f7ee0abcd2
Removing propably accidentally added debug messages sent to verbose channel
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2007-01-02 20:02:37 +00:00
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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
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
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
7674260e75
Add version number to useragent string - Issue #8700 , blanchet - THANKS!
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2006-12-31 09:31:23 +00:00
fc86ea2eed
Removing extra output
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2006-12-29 11:02:28 +00:00
b54feaffec
Only include include files once (imported from 1.4)
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2006-12-27 21:27:57 +00:00
906a51a1af
Only set "rfc2833compensate" option once
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2006-12-27 20:30:03 +00:00
8794577966
Only handle T38 options once
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2006-12-27 20:26:30 +00:00
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
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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
29ed493b40
Be politically correct
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2006-12-27 18:02:10 +00:00
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
3713ce7ed1
Cleanup of handle_common_options
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2006-12-27 16:35:41 +00:00
a12dc35eb3
Reset invitestate when sending new invite
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2006-12-27 16:26:02 +00:00
d69e9086cc
Issue #8600 - bogus SDP Content Length in T.38 re-invite
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2006-12-27 16:01:07 +00:00
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
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
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
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
bf2172faac
Update to latest IANA specs
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2006-12-15 10:54:06 +00:00
666d526aad
Fix various spelling mistakes in comments.
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2006-12-12 22:32:20 +00:00
2b42422940
Handle multiple 487's correctly
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06 16:54:27 +00:00
c39f887ac8
Don't send Contact in SIP Messages (imported from 1.2/1.4).
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Reported by Gunnar at Omnitor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06 12:34:58 +00:00
a44e55a3f3
Constify a bunch of the usage strings for CLI commands.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06 07:23:32 +00:00
22142e1aa0
Cleaning up handle_response a bit. (Imported from 1.4)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-02 22:03:14 +00:00
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-02 20:30:58 +00:00
fc6de9e982
Invitestate updates
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-02 19:11:02 +00:00
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-02 12:05:40 +00:00
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-01 18:16:16 +00:00
813f8d26fe
Issue #8319 (imported from 1.2, 1.4) - Increment nonce-count properly (noriyuki)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-30 20:55:43 +00:00
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-30 09:23:25 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-30 03:29:42 +00:00
42fb134171
Change error message (imported from 1.4)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-27 06:44:40 +00:00
e46e42ceec
Changing ERROR to lesser level. Imported from 1.2/1.4
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-25 09:48:30 +00:00
a4cc4d03c9
Remove unused memory allocation
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-23 11:04:25 +00:00
4221020191
Treat 101 as 100, not 183 session progress
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-21 15:25:38 +00:00
230376c7c0
Properly reset schedule items (rizzo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-20 11:36:45 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-18 18:03:42 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-16 21:13:17 +00:00
a6f5adefa1
Make it possible to enable/disable onhold tracking, in order to make life easier
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for realtime users.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-16 19:29:28 +00:00
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! :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-16 16:02:41 +00:00
a427a2a89a
- CANCEL never uses authentication
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- Add docs on canreinvite
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-16 15:12:30 +00:00
5231f5aff8
Hunting the initreq change for an ACK
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-15 21:21:21 +00:00
015efdd558
- Don't reply to ACK
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- Improve SIP history for debugging
(Imported from 1.4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-15 20:40:22 +00:00
88a518c2dd
Send proper SIP error message improperly when we can't allocate dialog (out of file handles is one cause)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-15 16:04:24 +00:00
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
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
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
f9f8237a2f
Adding a new debug line for issue #7351 - trying to find where an ACK can overwrite the initreq...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-14 12:32:46 +00:00
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
fb211aa610
Remove flags not used any more (thanks Luigi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-14 11:00:33 +00:00
98e17e1cb9
Merged revisions 47581 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-13 20:21:33 +00:00
ea64cd393f
Small fix for uncommon scenario.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-13 19:20:54 +00:00
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
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 :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-13 14:29:31 +00:00
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-13 14:14:54 +00:00
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
2286ff9659
Restore auto-framing (DEA). Imported from 1.4
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-12 16:15:49 +00:00
aa275d4970
- Support UDPTL as well as udptl in T38 sdp.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-12 15:40:03 +00:00
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
1f9dcf758c
- Add some comments to t.38 code
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- Remove improper blocking of ptime: in SDP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-12 15:14:35 +00:00
0c18d91817
Add some history and fix some debug output for autodestruct.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 21:55:31 +00:00
2b6d3952e1
Proper fix for adding debug...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 21:39:58 +00:00
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
d69eb90f82
Cleanup imported from 1.4
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-10 20:25:32 +00:00
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
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
2c5bbdc938
Ripping out bad support for 491 replies to INVITE's. Let's try again properly later.
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2006-11-10 13:14:44 +00:00
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
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
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 22:35:22 +00:00
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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2006-11-08 18:28:53 +00:00
e108453e02
coding guidelines, coding guidelines, coding guidelines
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2006-11-08 15:28:39 +00:00
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 13:59:32 +00:00
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
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
24dfbe955e
merge rev.47224 from team/rizzo/astobj2:
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hide dialoglist lock/unlocking in wrapper functions.
Approved By: oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 13:44:27 +00:00
f58302fc9c
silence compiler about uninitialized variables.
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The compiler is wrong, but it has the last word.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 13:43:06 +00:00
e644ba34c5
Destroy dialog properly at unload (rizzo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-08 08:01:44 +00:00
908f176cf3
A fair number of changes for the sake of bug 7506
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-07 21:47:49 +00:00
b557aeda38
Break -> continue to make stuff work... Thanks, Luigi!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-07 17:33:24 +00:00
20af2870d9
issue #8265 - don't reply to ACK. Imported from 1.2, 1.4
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-07 13:26:31 +00:00
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 ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-04 22:16:57 +00:00
80293c2ed8
Move IP address selection for media out of add_sdp
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2006-11-04 21:48:02 +00:00
18b694d318
Make srvlookup global_srvlookup to follow the rest of the code
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-04 21:17:40 +00:00
fe5440b7c8
Simplify previous patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-04 19:20:30 +00:00
Olle Johansson
Russell Bryant
Joshua Colp
Dwayne M. Hubbard
Steve Murphy
Tilghman Lesher
Kevin P. Fleming
Jason Parker
Luigi Rizzo