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r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 lines
Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)
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2007-06-04 19:32:08 +00:00
93f2be2675
Fix a couple of places where "tos" was used instead of "cos".
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(issue #9540 , patch by IgorG)
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2007-06-04 15:14:52 +00:00
3d2b58751f
To satisfy some legal concerns, add an option for chan_sip to include a
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disclaimer along with SIP messages in the header, X-Disclaimer. This is off by
default. Also, the text of the disclaimer can be customized in sip.conf.
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2007-05-31 19:41:03 +00:00
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r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines
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r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines
It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)
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2007-05-31 16:18:14 +00:00
0b2db74e5a
Issue #9842 - Doxygen updates by snuffy. Thanks!
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(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)
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2007-05-31 10:26:55 +00:00
4c1068c136
oops. Thanks Terry.
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2007-05-29 19:53:40 +00:00
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r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 lines
Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.
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2007-05-29 19:35:43 +00:00
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r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines
Don't issue hangup on hangup on hangup on hangup (for jcmoore)
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2007-05-29 19:17:49 +00:00
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r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines
Don't reset hangupcause if we already have one
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2007-05-29 16:19:53 +00:00
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r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines
Tracking down hanging channels, killing them one by one. Issue #9235 and related
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2007-05-29 16:17:17 +00:00
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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines
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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines
Issue #9802 - Change inuse counter on CANCEL
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2007-05-29 10:02:31 +00:00
4d03b4f268
Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel)
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2007-05-28 23:28:52 +00:00
39e9b3112c
Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj)
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2007-05-28 23:24:04 +00:00
4b3a3fb14c
Add a new API call for creating detached threads. Then, go replace all of the
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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2007-05-24 18:30:19 +00:00
202fbe363a
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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines
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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines
Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)
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2007-05-24 14:43:49 +00:00
9b95428cce
Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there
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but is still ok... Sorry. Lack of Tea, really.
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2007-05-24 14:41:43 +00:00
0ec502099f
Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure...
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2007-05-24 11:38:20 +00:00
d1ba4f90de
- Remove debug variable that was only used in one place
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- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
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2007-05-24 03:28:39 +00:00
04e45cfda3
Add SendURL support
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2007-05-24 02:23:08 +00:00
5dc23536ec
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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines
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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines
ensure that variables are set on a newly created channel before we start a PBX on it
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2007-05-23 20:53:10 +00:00
a3f9350ec2
Related to issue #9235 btw.
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines
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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines
Not getting an ACK to a 200 OK in the initial invite is critical to the call.
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2007-05-18 18:18:59 +00:00
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines
Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.
A special Thank You to WeBRainstorm that gave me access to his system.
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2007-05-18 15:20:39 +00:00
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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines
Issue 9487 - stop media flows at hangup of call
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2007-05-18 10:41:31 +00:00
ac343d43c8
Makeup, darling.
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2007-05-18 10:28:56 +00:00
f3ec447a23
Another fix for the support for recordings controlled by INFO-packets
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We still lack a setting to enable/disable this per peer
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2007-05-18 08:49:34 +00:00
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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
c472b899ef
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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
c352f7b0d5
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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
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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
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
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
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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
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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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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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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
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
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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
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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
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
daefa6a8b4
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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
240bd841b0
Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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2007-04-27 14:40:28 +00:00
f9c592e50c
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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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2007-04-27 14:37:10 +00:00
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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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2007-04-13 21:23:10 +00:00
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
80ec0b13ba
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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
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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2007-04-11 16:06:37 +00:00
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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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2007-04-11 15:13:12 +00:00
e34c67d308
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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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2007-04-11 14:01:53 +00:00
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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
95a7dc0509
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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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2007-04-05 12:57:35 +00:00
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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
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
11ab6db24b
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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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2007-03-21 03:27:58 +00:00
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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
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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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2007-03-13 10:14:13 +00:00
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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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2007-03-06 20:45:29 +00:00
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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
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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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2007-02-22 18:53:22 +00:00
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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
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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
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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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2007-02-21 17:23:42 +00:00
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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
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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
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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
e88dda8ca9
Merged revisions 53064 via svnmerge from
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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
b233892198
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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
d11f8b7ccd
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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
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
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
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
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
1a06a58250
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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
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
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
Joshua Colp
Russell Bryant
Olle Johansson
Kevin P. Fleming
Mark Spencer
Steve Murphy
Dwayne M. Hubbard
Tilghman Lesher
Jason Parker