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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines
Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)
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2007-05-17 16:11:26 +00:00
d83dcae6b1
Below patches with some re-structuring for trunk
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r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines
Issue #9681 - Handle www-auth on BYE
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2007-05-16 10:58:22 +00:00
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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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2007-05-16 10:09:42 +00:00
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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines
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r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines
Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)
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2007-05-16 10:02:06 +00:00
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r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines
Merged following patch with a lot of changes for 1.4
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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines
Issue #9726 - rlister - Better logging for ACL denials
While at it, also added better logging and handling of peers that are not supposed to register.
My patch, stole the issue report from Russell. My apologies, Russell :-)
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2007-05-16 08:51:39 +00:00
7532b0bc4b
Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej
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2007-05-16 07:58:43 +00:00
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
9b5454232c
XXX-XXX-XXX appears to be the standard ANSI pointcode format
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2007-05-15 20:45:20 +00:00
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
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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
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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
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
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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
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
b71e691b3e
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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
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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
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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
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
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r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, 10 May 2007) | 12 lines
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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
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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines
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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
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r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, 09 May 2007) | 10 lines
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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
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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines
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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
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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
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
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
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Adding external reference
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2007-05-07 18:18:43 +00:00
238bcd6c45
Doxyfication... There's a shortage of comments in this file...
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2007-05-07 18:17:18 +00:00
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
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
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
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
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
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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines
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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
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r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines
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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
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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
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
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
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
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
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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
a91f9b138d
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r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines
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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
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
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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
7d2102c081
Reformat some of iax2.h and convert comments to doxygen format
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2007-04-28 21:26:00 +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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27 21:11:46 +00:00
240bd841b0
Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27 14:40:28 +00:00
f9c592e50c
Merged revisions 62137 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27 14:37:10 +00:00
14d8979b2f
Merged revisions 62038 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-26 16:35:14 +00:00
Joshua Colp
Olle Johansson
Matthew Fredrickson
Russell Bryant
Steve Murphy
Jason Parker
Tilghman Lesher