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Author	SHA1	Message	Date
Richard Mudgett	252cffd38d	Merged revisions 293806 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6dfc9dddd8	Avoid valgrind warnings for ast_rtp_instance_get_xxx_address ... The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	9a482b0724	Merged revisions 293723 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	643d500d66	Merged revisions 293647 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) | 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	2c98e867f4	If manager and tls are disabled, do not display TCP/TLS Bindaddress. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ea00bff969	Analog 3-way call would not connect all parties if one was using sig_pri. ... Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	8b53a54dc3	Use ast_sockaddr_from_sin function not memcpy ... This resolves some IAX2 registration issue report on the asterisk-users mailing list. (closes issue #18202) Reported by: pabelanger Patches: update_registry.patch.v2 uploaded by pabelanger (license 224) Tested by: pabelanger, Nic Colledge (mailing list) Review: https://reviewboard.asterisk.org/r/993 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	de440fa10d	Only offer codecs both sides support for directmedia ... When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	651d43c698	Merged revisions 293417 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) | 1 line Remove some more code that serves no purpose. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	55a1b13da8	Merged revisions 293340 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) | 1 line Remove some code that serves no purpose. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	da39c33809	Modify sip_setoption to not complain about unknown options. ... This now behaves just like the other setoption callbacks. For the curious the offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting passed due to a fix for chan_local in 286189. (closes issue #17985) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e90dbe2963	Merged revisions 293195-293196 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ r293196 | tilghman | 2010-10-28 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	971db9fc4e	Merged revisions 293158 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6cbda6ed92	Merged revisions 293118 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	70415ccdfd	No need to define the struct if there are no users. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d411ade931	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. ... Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	d1aceac167	Several more defines that need to be altered for compiling against an older version of libpri ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cdc420235e	Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. ... Fixes our Bamboo builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3020818583	Merged revisions 292867 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	739dcea39b	Don't create directories without at least o+x ... Also, making files that you are going to modify read-only is dumb. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	af6a9e6199	Make files readable only by the owner ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	9baf979137	Merged revisions 292786 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines Update the LDIF file for LDAP. The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	fff80bbe00	Prevent multiple runs of event_sub_test from producing false failure results. ... The array of test subscriptions was declared "static," meaning that the data.count field would retain its value between runs of the test. After the first test run, this would result in false reports of test failures. I chose to just remove the "static" keyword from the structure since it's not a huge deal to construct this structure during each run of the test. Another alternative would have been to zero out the data.count fields of each test subscription instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	177d65f59c	Add TLS cert helper script ... This script is useful for quickly generating self-signed CA, server, and client certificates for use with Asterisk. It is still recommended to obtain certificates from a recognized Certificate Authority and to develop an understanding how SSL certificates work. Real security is hard work. OPTIONS: -h Show this message -m Type of cert "client" or "server". Defaults to server. -f Config filename (openssl config file format) -c CA cert filename (creates new CA cert/key as ca.crt/ca.key if not passed) -k CA key filename -C Common name (cert field) For a server cert, this should be the same address that clients attempt to connect to. Usually this will be the Fully Qualified Domain Name, but might be the IP of the server. For a CA or client cert, it is merely informational. Make sure your certs have unique common names. -O Org name (cert field) An informational string (company name) -o Output filename base (defaults to asterisk) -d Output directory (defaults to the current directory) Example: To create a CA and a server (pbx.mycompany.com) cert with output in /tmp: ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp This will create a CA cert and key as well as asterisk.pem and the the two files that it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem and ca.crt somewhere (like /etc/asterisk) and set tlscertfile=/etc/asterisk.pem and tlscafile=/etc/ca.crt. Since this is a self-signed key, many devices will require you to import the ca.crt file as a trusted cert. To create a client cert using the CA cert created by the example above: ast_tls_cert -m client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \ "My Company" -d /tmp -o joe_user This will create client.crt/key/pem in /tmp. Use this if your device supports a client certificate. Make sure that you have the ca.crt file set up as a tlscafile in the necessary Asterisk configs. Make backups of all .key files in case you need them later. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	eeec27dafc	Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call. ... When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cb0d414f4b	Compile correctly on Linux (asterisk/localtime.h depends upon asterisk/autoconfig.h loading first). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	ce9d013cbd	Fix typo in SUSE init script. ... Reported by: Dave Cotton on asterisk-users list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	042d5edaa3	Fixes recursive lock problem in manager.c ... It is possible for a AMI session to freeze because of invalid use of recursive locks during the EVENT processing. This patch removes the unnecessary locks. (closes issue #18167) Reported by: sustav Patches: manager_locking_v1.diff uploaded by dvossel (license 671) Tested by: sustav git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	8c0627fada	Merged revisions 292556 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010) | 6 lines Change res_ldap.sample.conf to match the schema. (closes issue #17376) Reported by: jcovert Patches: res_ldap.conf.sample.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	d39850c2e9	Add var=value to log message on update failure, and add newline. ... ... just for you, Leif. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	7e28c7c90d	Send CONNECT_ACKNOWLEDGE for CIS calls too. ... The originator of the Q.SIG call completion signaling link was not changed to the active state when the CONNECT message came in. The T309 processing would immediately kill the signaling link because it was not in the active state. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	83ed33746e	Application not properly unregister in voicemail ... (closes issue #18128) Reported by: junky Patches: vm_unregister.diff uploaded by junky (license 177) Tested by: pabelanger, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	d4a74bd2ae	Merged revisions 292412 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6ff3d6b761	Oops. This module uses the generic timer and no longer uses DAHDI. ... This causes a problem with the Solaris and other system builds that have gcc 4.1 (where optional_api is non-optional). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	4b5ec2055b	Add resample and imap_tk dependencies. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	668d532d6b	Add sip show peer info about crypto and remove dated comment ... This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	b334bdef1e	Merged revisions 292229 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010) | 3 lines Fix typo in the sounds/Makefile. (Issue #17426) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	aecdf5d980	Merged revisions 292226 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	be6ddcedcd	Merged revisions 292224 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500 (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) | 9 lines Add support for the new English (Australian Accent) sound files. (closes issue #17426) Reported by: camsown Patches: core-sounds-en_AU.txt uploaded by camsown (license 1050) add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested by: camsown, lmadsen, jtodd, qwell ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	f10b21bd24	Resolve some compiler errors in ast_sockaddr_is_any(). ... These errors came up once this function was used from within netsock2.c. The errors were like the following: netsock2.c:393: error: dereferencing pointer ‘({anonymous})’ does break strict-aliasing rules The usage of a union here avoids this problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	64a0eebfa3	Fixes build error for systems not supporting IPV6_TCLASS. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	aa0351aac9	Fix the cmgr parser. ... (closes issue 0018152) Reported by: menschentier git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	df369754da	Fixes qos settings for sockets bound to any IPv6 or IPv4 address. ... (closes issue #18099) Reported by: jamesnet Patches: issues_18099_v3.diff uploaded by dvossel (license 671 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	d04d58c7c6	Disable use of inotify for call file handling as it is not working properly. ... (related to #18089) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	12211219dc	Merged revisions 292049 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) | 15 lines Base directory for MOH should be ASTDATADIR If the directive 'directory' is relative, make it relative to the datadir, rather than to the varlibdir. In the sample configuration it is relative ('moh'). This has no effect unless you have actively set the datadir explicitly (at build time or at run time). (closes issue #16906) Patches: moh_datadir uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/974/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	5b5faf3fc5	Ref/unref res_srtp when we create/destroy a session ... This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp tries to unload before chan_sip does. Thanks, Russell! (closes issue #18085) Reported by: st git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292016 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3e3ea54864	Fixes peer's host port information being lost on sip reload. ... (closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	67f8391cee	Merged revisions 291939 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400 (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct 2010) | 2 lines Clean up formatting. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	36a672b7e8	Merged revisions 291904 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) | 7 lines Don't crash or deadlock on module unload We can't hold the lock while pthread_join is called since aji_log_hook will attempt to lock from the other therad. We reorder the pthread_join and ast_aji_disconnect so that we don't do an SSL_read() while SSL_shutdown is running, causing a crash. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e735b3f28c	Set TCLASS field of IPv6 header when sip qos options are set. ... (closes issue #18099) Reported by: jamesnet Patches: issues_18099_v2.diff uploaded by dvossel (license 671) Tested by: dvossel, jamesnet git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago