Mark Spencer
04e45cfda3
Add SendURL support
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18 years ago
Kevin P. Fleming
48895d885d
Merged revisions 65685 via svnmerge from
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r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007) | 2 lines
start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
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18 years ago
Kevin P. Fleming
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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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Kevin P. Fleming
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r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007) | 2 lines
don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else)
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r65680 | kpfleming | 2007-05-23 16:35:50 -0400 (Wed, 23 May 2007) | 2 lines
clear the 'delay PBX' flag when we are ready to start the PBX
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Kevin P. Fleming
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r65677 | kpfleming | 2007-05-23 16:07:59 -0400 (Wed, 23 May 2007) | 10 lines
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r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines
if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it
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18 years ago
Matthew Fredrickson
cc016de696
Make sure we get the cause code in the REL
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18 years ago
Russell Bryant
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r65589 | russell | 2007-05-23 08:07:13 -0500 (Wed, 23 May 2007) | 11 lines
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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines
Revert revision 62417 as someone reported problems with it to Mark. This was
related to issue #9588 .
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Russell Bryant
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r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) | 3 lines
List res_smdi as a dependency for app_voicemail and chan_zap
(Thanks to mnicholson for pointing it out)
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18 years ago
Joshua Colp
a3e603cd0f
Use ast_strlen_zero where possible. (issue #9774 reported by eliel)
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18 years ago
Christian Richter
47362f0906
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r65342 | crichter | 2007-05-22 10:12:20 +0200 (Di, 22 Mai 2007) | 9 lines
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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line
we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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18 years ago
Olle Johansson
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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Olle Johansson
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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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18 years ago
Christian Richter
b3231b9db6
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r65039 | crichter | 2007-05-18 14:40:46 +0200 (Fr, 18 Mai 2007) | 9 lines
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r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line
fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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18 years ago
Christian Richter
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r64904 | crichter | 2007-05-18 10:58:51 +0200 (Fr, 18 Mai 2007) | 9 lines
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r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line
we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
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Olle Johansson
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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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18 years ago
Olle Johansson
ac343d43c8
Makeup, darling.
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18 years ago
Christian Richter
c282567843
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r64515 | crichter | 2007-05-16 10:44:51 +0200 (Mi, 16 Mai 2007) | 9 lines
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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line
in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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18 years ago
Christian Richter
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r63534 | crichter | 2007-05-09 15:17:10 +0200 (Mi, 09 Mai 2007) | 17 lines
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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18 years ago
Christian Richter
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r62912 | crichter | 2007-05-03 16:36:32 +0200 (Do, 03 Mai 2007) | 17 lines
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r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line
some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
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18 years ago
Christian Richter
2cdaf67ae0
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r59774 | crichter | 2007-04-03 09:20:27 +0200 (Di, 03 Apr 2007) | 17 lines
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r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line
we can now make 30 channels on a PRI (before we forgot chan 31..)
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r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line
don't be verbose if no need
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r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line
added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour
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18 years ago
Christian Richter
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r59254 | crichter | 2007-03-27 17:00:10 +0200 (Di, 27 Mär 2007) | 9 lines
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r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 Mär 2007) | 1 line
fixed #9355
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Christian Richter
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r59064 | crichter | 2007-03-20 14:16:06 +0100 (Di, 20 Mär 2007) | 21 lines
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r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line
added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers
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r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line
fixed the crypt_keys stuff
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r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one.
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r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line
modified a loglevel
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Christian Richter
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r58825 | crichter | 2007-03-12 13:43:24 +0100 (Mo, 12 Mär 2007) | 1 line
added UU transceiving and corect handling for rdnis
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r58826 | crichter | 2007-03-12 14:08:06 +0100 (Mo, 12 Mär 2007) | 21 lines
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r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line
fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302 , #303 , #304
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r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line
fixed typo
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r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line
fixed another place where the out_cause was hardcoded to 16
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r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line
we can free channel 31 as well, since we can occupy it
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18 years ago
Olle Johansson
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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18 years ago
Joshua Colp
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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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18 years ago
Olle Johansson
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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Olle Johansson
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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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Olle Johansson
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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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Olle Johansson
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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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18 years ago
Olle Johansson
7532b0bc4b
Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej
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18 years ago
Olle Johansson
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Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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18 years ago
Matthew Fredrickson
9b5454232c
XXX-XXX-XXX appears to be the standard ANSI pointcode format
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18 years ago
Matthew Fredrickson
79f553a4ab
Only print the SS7 UP once. Not every time we get the test messages on the
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Olle Johansson
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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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Russell Bryant
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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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18 years ago
Joshua Colp
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If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
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18 years ago
Steve Murphy
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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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18 years ago
Olle Johansson
fc057b1890
Improve handling network errors on transmission to hosts that don't reply or are unreachable
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With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.
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18 years ago
Joshua Colp
b71e691b3e
Merged revisions 64114 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Joshua Colp
38e951cfda
Merged revisions 64086 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Joshua Colp
82a30356da
Merged revisions 64044 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Jason Parker
60f6a72123
Add/fix support for Redial, Speeddial, and Messages buttons.
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Combined effort by DEA and mvanbaak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
2768946cf5
Merged revisions 63830 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, 10 May 2007) | 12 lines
Merged revisions 63828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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18 years ago
Olle Johansson
aa320037d2
Merged revisions 63749 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines
Merged revisions 63748 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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18 years ago
Matthew Fredrickson
e2ca869abd
Merged revisions 63654 via svnmerge from
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r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, 09 May 2007) | 10 lines
Merged revisions 63653 via svnmerge from
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r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines
Make sure we only create a DSP if it's requested on SUB_REAL
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18 years ago
Joshua Colp
7e10164e20
Merged revisions 63611 via svnmerge from
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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines
Merged revisions 63610 via svnmerge from
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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18 years ago
Olle Johansson
c358b18a5a
Merged revisions 63532 via svnmerge from
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r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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18 years ago
Russell Bryant
314c874d7d
I noted this on the dev list but got no response, so I just did it myself.
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Lock the call features when being used in chan_sip.
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18 years ago
Olle Johansson
a39f95b94f
Adding external referenses for doxygen
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See http://www.asterisk.org/doxygen/trunk/extref.html
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18 years ago
Olle Johansson
33214f76a7
Adding external reference
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18 years ago
Olle Johansson
238bcd6c45
Doxyfication... There's a shortage of comments in this file...
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18 years ago
Joshua Colp
28f4727e75
Lock iax2 pvt structure when passing off to the AMI function, and make sure it exists. (issue #9674 reported by arabe)
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18 years ago
Olle Johansson
d326d84ae0
- Adding some missing spaces
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- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily
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18 years ago
Steve Murphy
02337303ef
a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Steve Murphy
3ee0077f04
Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
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18 years ago
Olle Johansson
1b15d8852d
Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
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is needed to follow the call through the PBX.
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18 years ago
Joshua Colp
81cade7a4c
Merged revisions 62989 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines
Merged revisions 62987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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18 years ago
Tilghman Lesher
121561076e
Merged revisions 62692 via svnmerge from
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r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines
Merged revisions 62691 via svnmerge from
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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.
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18 years ago
Steve Murphy
fe7068a51b
Merged revisions 62689 via svnmerge from
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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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18 years ago
Russell Bryant
3d409eb793
Update the device state functionality of chan_local such that it will return
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NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048 , patch from tim_ringenbach)
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18 years ago
Olle Johansson
e1ec3f917c
Add a small message that we're doing something. On my systems, there's a long
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dead period with a non-responsive CLI after I issue "load chan_sip.so"
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18 years ago
Olle Johansson
1d51b2e161
More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
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But first, some serious SIP testing :-)
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18 years ago
Olle Johansson
8fee67c83b
Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly
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18 years ago
Olle Johansson
daefa6a8b4
Merged revisions 62624 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Russell Bryant
b419fc1134
Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540 , patch submitted by IgorG)
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18 years ago
Russell Bryant
a91f9b138d
Merged revisions 62419 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines
Merged revisions 62417 via svnmerge from
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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588 , reported and patched by softins)
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18 years ago
Russell Bryant
5cb08adc7a
Don't crash when invalid arguments are provided to the CHANNEL() function
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for a SIP channel.
(issue #9619 , reported by jtodd, original patch by Corydon76, committed patch
slightly modified by me)
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18 years ago
Russell Bryant
0efe511879
Merged revisions 62331 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Russell Bryant
7d2102c081
Reformat some of iax2.h and convert comments to doxygen format
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
b6b1bf3213
Merge changes from team/russell/events
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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18 years ago
Russell Bryant
c2468b4c32
Merged revisions 62218 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Olle Johansson
240bd841b0
Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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18 years ago
Olle Johansson
f9c592e50c
Merged revisions 62137 via svnmerge from
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r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines
Merged revisions 62126 via svnmerge from
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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18 years ago
Joshua Colp
14d8979b2f
Merged revisions 62038 via svnmerge from
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r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines
Merged revisions 62037 via svnmerge from
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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
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18 years ago
Kevin P. Fleming
ee95074173
Merged revisions 61914 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61913 via svnmerge from
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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines
handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286 )
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18 years ago
Russell Bryant
dc7514a746
Merged revisions 61870 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61866 via svnmerge from
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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines
If the callerid= option is specified, but empty, clear any previous data.
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18 years ago
Russell Bryant
891a005706
Merged revisions 61863 via svnmerge from
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r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61862 via svnmerge from
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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines
Ensure that callerid settings are reset on a reload.
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18 years ago
Russell Bryant
f5d4a16cda
Merged revisions 61799 via svnmerge from
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r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines
Merged revisions 61798 via svnmerge from
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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines
Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587 , reported and patched by xrg)
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18 years ago
Dwayne M. Hubbard
f3ab33014a
removed #if 0 block from chan_zap restart_monitor()
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18 years ago
Joshua Colp
721f85d084
Merged revisions 61772 via svnmerge from
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r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines
Merged revisions 61771 via svnmerge from
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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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18 years ago
Russell Bryant
b94378b5e6
Merge changes from team/russell/iax2_osp
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This set of changes adds OSP support to chan_iax2. However, I have modified
the patch a bit from what was submitted. You now use the CHANNEL() function
to get and set the OSP token for IAX2.
(issue #8531 , reported by and original patch by homesick, patch updated by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
49af71c100
Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-)
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This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason.
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18 years ago
Dwayne M. Hubbard
34469a8707
added CLI 'sip unregister <peer>' for issue 9326. thanks eliel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
4f04ff8597
Merged revisions 61648 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Steve Murphy
901413c76c
Merged revisions 61644 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 line
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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18 years ago
Joshua Colp
80ec0b13ba
Merged revisions 61641 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Joshua Colp
c4c2def716
Don't treat a host lookup as failed if sipregs is not in use when doing a realtime lookup. (issue #9255 reported by sergee)
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18 years ago
Russell Bryant
3c0b24bda8
Merged revisions 61477 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines
Merged revisions 61476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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18 years ago
Nadi Sarrar
9978dc647b
Merged revisions 61342,61372-61373,61443 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2 lines
AOCD's are now exported to asterisk channel variables.
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r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr 2007) | 2 lines
Ignore facility messages in case we don't have a corresponding channel object.
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r61373 | nadi | 2007-04-11 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines
Export AOCD variables on misdn_hangup.
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r61443 | nadi | 2007-04-11 17:39:14 +0200 (Mi, 11 Apr 2007) | 2 lines
Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..
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18 years ago
Russell Bryant
6b033eea04
Merged revisions 61427 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines
Merged revisions 61426 via svnmerge from
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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines
Fix a bug with switching between host=dynamic and using specific hosts for
peers. The code would only reset the peer's address when it is dynamic if
it was a new peer structure. Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515 , reported by caio1982, fixed by me)
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18 years ago
Russell Bryant
e34c67d308
Merged revisions 61377 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines
Merged revisions 61376 via svnmerge from
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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
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18 years ago
Joshua Colp
9fff461080
Remove duplicate prototype declaration. (issue #9517 reported by junky)
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18 years ago
Doug Bailey
7c9e46507b
Added zapata.conf parameter "cid_rxgain" to allow the user to adjust the gain bump used during CID acquisition.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Steve Murphy
28b5fb02bd
updated ast_channel_alloc() call to include the 4 extra args everyone got. Not much info there, as the config file evidently does not allow amaflags, or accountcode settings; and the pvt's exten doesn't sound like what we need in the cdr, either.
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18 years ago
Nadi Sarrar
0220ec0511
Merged revisions 61183 via svnmerge from
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r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di, 10 Apr 2007) | 10 lines
Merged revisions 61170 via svnmerge from
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r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 lines
msns config parameter defaults to '*'
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18 years ago
Steve Murphy
ecaf781933
Merged revisions 60989 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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18 years ago
Olle Johansson
b52f774850
Merged revisions 61072 via svnmerge from
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r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines
Merged revisions 61038 via svnmerge from
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r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 lines
- Don't send ActionID before Response: header.
- Don't use a blank in an AMI header
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18 years ago
Olle Johansson
4aef0155d6
use "ChannelType" in events to indicate which channel driver that generates the event. This replaces
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"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
a4bef3bb3a
Make RTP session ID and session version generation random. (issue #9456 reported by tjardick)
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18 years ago
Tilghman Lesher
c6c64af72d
Merged revisions 60847 via svnmerge from
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r60847 | tilghman | 2007-04-08 21:42:48 -0500 (Sun, 08 Apr 2007) | 10 lines
Merged revisions 60846 via svnmerge from
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r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines
Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid.
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18 years ago