Commit graph

4141 commits

Author SHA1 Message Date
Russell Bryant
d1ba4f90de - Remove debug variable that was only used in one place
- convert string handling to the ast_str API
 - Convert strdup() to ast_strdup() and check the result
 - Minor formatting changes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 03:28:39 +00:00
Mark Spencer
04e45cfda3 Add SendURL support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 02:23:08 +00:00
Kevin P. Fleming
48895d885d Merged revisions 65685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-23 21:01:18 +00:00
Kevin P. Fleming
5dc23536ec Merged revisions 65683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines

Merged revisions 65682 via svnmerge from 
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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
Kevin P. Fleming
04da47df24 Merged revisions 65679-65680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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2007-05-23 20:39:42 +00:00
Kevin P. Fleming
61a53d859e Merged revisions 65677 via svnmerge from
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r65677 | kpfleming | 2007-05-23 16:07:59 -0400 (Wed, 23 May 2007) | 10 lines

Merged revisions 65676 via svnmerge from 
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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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2007-05-23 20:08:59 +00:00
Matthew Fredrickson
cc016de696 Make sure we get the cause code in the REL
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2007-05-23 15:37:51 +00:00
Russell Bryant
5558323d0c Merged revisions 65589 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65589 | russell | 2007-05-23 08:07:13 -0500 (Wed, 23 May 2007) | 11 lines

Merged revisions 65588 via svnmerge from 
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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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2007-05-23 13:10:28 +00:00
Russell Bryant
d3ee017682 Merged revisions 65501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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2007-05-22 18:41:56 +00:00
Joshua Colp
a3e603cd0f Use ast_strlen_zero where possible. (issue #9774 reported by eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-22 15:08:08 +00:00
Christian Richter
47362f0906 Merged revisions 65342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65342 | crichter | 2007-05-22 10:12:20 +0200 (Di, 22 Mai 2007) | 9 lines

Merged revisions 65328 via svnmerge from 
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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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2007-05-22 08:21:59 +00:00
Olle Johansson
a3f9350ec2 Related to issue #9235 btw.
Merged revisions 65123 via svnmerge from 
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines

Merged revisions 65122 via svnmerge from 
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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
Olle Johansson
1bc7fdeb6b Merged revisions 65076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines

Merged revisions 65075 via svnmerge from 
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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
Christian Richter
b3231b9db6 Merged revisions 65039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65039 | crichter | 2007-05-18 14:40:46 +0200 (Fr, 18 Mai 2007) | 9 lines

Merged revisions 65007 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 12:43:59 +00:00
Christian Richter
37f024bb90 Merged revisions 64904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64904 | crichter | 2007-05-18 10:58:51 +0200 (Fr, 18 Mai 2007) | 9 lines

Merged revisions 64902 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 11:17:16 +00:00
Olle Johansson
491827a28a Merged revisions 64974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Olle Johansson
ac343d43c8 Makeup, darling.
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2007-05-18 10:28:56 +00:00
Christian Richter
c282567843 Merged revisions 64515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64515 | crichter | 2007-05-16 10:44:51 +0200 (Mi, 16 Mai 2007) | 9 lines

Merged revisions 64513 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:03:38 +00:00
Christian Richter
2ce08e259c Merged revisions 63534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r63534 | crichter | 2007-05-09 15:17:10 +0200 (Mi, 09 Mai 2007) | 17 lines

Merged revisions 62945,63402,63519 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 10:00:21 +00:00
Christian Richter
0f39ea1a98 Merged revisions 62912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62912 | crichter | 2007-05-03 16:36:32 +0200 (Do, 03 Mai 2007) | 17 lines

Merged revisions 61357,61770,62885 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 09:50:33 +00:00
Christian Richter
2cdaf67ae0 Merged revisions 59774 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59774 | crichter | 2007-04-03 09:20:27 +0200 (Di, 03 Apr 2007) | 17 lines

Merged revisions 59623-59624,59639 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 09:47:19 +00:00
Christian Richter
b05f83410a Merged revisions 59254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59254 | crichter | 2007-03-27 17:00:10 +0200 (Di, 27 Mär 2007) | 9 lines

Merged revisions 59252 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 09:42:42 +00:00
Christian Richter
e6cad524f8 Merged revisions 59064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59064 | crichter | 2007-03-20 14:16:06 +0100 (Di, 20 Mär 2007) | 21 lines

Merged revisions 58849-58850,59062-59063 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 09:36:16 +00:00
Christian Richter
ecfbebafe9 Merged revisions 58825-58826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

Merged revisions 57034,57523,57753,58558 via svnmerge from 
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 09:31:27 +00:00
Olle Johansson
f3ec447a23 Another fix for the support for recordings controlled by INFO-packets
We still lack a setting to enable/disable this per peer


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2007-05-18 08:49:34 +00:00
Joshua Colp
a769766c53 Merged revisions 64754 via svnmerge from
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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
Olle Johansson
d83dcae6b1 Below patches with some re-structuring for trunk
---
Merged revisions 64602 via svnmerge from 
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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
Olle Johansson
c472b899ef Merged revisions 64578 via svnmerge from
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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
Olle Johansson
c352f7b0d5 Merged revisions 64543 via svnmerge from
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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines

Merged revisions 64535 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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
Olle Johansson
09aec2f622 Merged revisions 64516 via svnmerge from
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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
------

Merged revisions 64514 via svnmerge from 
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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
Olle Johansson
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
Olle Johansson
90bad9d2f5 Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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2007-05-16 07:35:56 +00:00
Matthew Fredrickson
9b5454232c XXX-XXX-XXX appears to be the standard ANSI pointcode format
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2007-05-15 20:45:20 +00:00
Matthew Fredrickson
79f553a4ab Only print the SS7 UP once. Not every time we get the test messages on the
line.


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2007-05-14 22:25:56 +00:00
Olle Johansson
fa2622cb1d Merged revisions 64324 via svnmerge from
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r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines

Change -2 to XMIT_ERROR to clarify a bit more

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2007-05-14 19:35:58 +00:00
Russell Bryant
ee7217c011 Merged revisions 64306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 19:21:31 +00:00
Joshua Colp
4fbb449444 If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 18:21:30 +00:00
Steve Murphy
a76ff39c0e Merged revisions 64193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 14:13:45 +00:00
Olle Johansson
fc057b1890 Improve handling network errors on transmission to hosts that don't reply or are unreachable
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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-13 19:20:36 +00:00
Joshua Colp
b71e691b3e Merged revisions 64114 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 22:28:09 +00:00
Joshua Colp
38e951cfda Merged revisions 64086 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 21:12:18 +00:00
Joshua Colp
82a30356da Merged revisions 64044 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12 16:33:34 +00:00
Jason Parker
60f6a72123 Add/fix support for Redial, Speeddial, and Messages buttons.
Combined effort by DEA and mvanbaak.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-11 22:52:36 +00:00
Jason Parker
2768946cf5 Merged revisions 63830 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 23:16:33 +00:00
Olle Johansson
aa320037d2 Merged revisions 63749 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 20:51:59 +00:00
Matthew Fredrickson
e2ca869abd Merged revisions 63654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 17:35:54 +00:00
Joshua Colp
7e10164e20 Merged revisions 63611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 16:56:45 +00:00
Olle Johansson
c358b18a5a Merged revisions 63532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 13:07:44 +00:00
Russell Bryant
314c874d7d I noted this on the dev list but got no response, so I just did it myself.
Lock the call features when being used in chan_sip.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-08 16:41:35 +00:00
Olle Johansson
a39f95b94f Adding external referenses for doxygen
See http://www.asterisk.org/doxygen/trunk/extref.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07 18:25:56 +00:00