2064 Commits

Author	SHA1	Message	Date
Olle Johansson	c358b18a5a	Merged revisions 63532 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ 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	d326d84ae0	- Adding some missing spaces ... - Correcting error messages - Disabling code that doesn't do anything - Making sure we always respond to this request, happily git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-05 08:05:38 +00:00
Steve Murphy	02337303ef	a small upgrade to the coding standard, and an update to the code that triggered the upgrade. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-04 17:49:20 +00:00
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. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-04 16:37:23 +00:00
Olle Johansson	1b15d8852d	Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that ... is needed to follow the call through the PBX. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-04 13:56:25 +00:00
Joshua Colp	81cade7a4c	Merged revisions 62989 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-03 16:45:39 +00:00
Olle Johansson	e1ec3f917c	Add a small message that we're doing something. On my systems, there's a long ... dead period with a non-responsive CLI after I issue "load chan_sip.so" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-02 12:12:02 +00:00
Olle Johansson	1d51b2e161	More username body parts to fix... If working, this needs to be backported to 1.2, 1.4. ... But first, some serious SIP testing :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-02 12:00:03 +00:00
Olle Johansson	8fee67c83b	Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-02 09:41:03 +00:00
Olle Johansson	daefa6a8b4	Merged revisions 62624 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-05-02 09:35:14 +00:00
Russell Bryant	b419fc1134	Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The ... file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-30 16:16:26 +00:00
Russell Bryant	5cb08adc7a	Don't crash when invalid arguments are provided to the CHANNEL() function ... for a SIP channel. (issue #9619, reported by jtodd, original patch by Corydon76, committed patch slightly modified by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-30 15:37:23 +00:00
Russell Bryant	b6b1bf3213	Merge changes from team/russell/events ... 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-28 21:01:44 +00:00
Olle Johansson	240bd841b0	Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks! ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-27 14:40:28 +00:00
Olle Johansson	f9c592e50c	Merged revisions 62137 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ 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. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-27 14:37:10 +00:00
Joshua Colp	721f85d084	Merged revisions 61772 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines Merged revisions 61771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-24 16:10:10 +00:00
Olle Johansson	49af71c100	Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-) ... This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-20 08:41:24 +00:00
Dwayne M. Hubbard	34469a8707	added CLI 'sip unregister <peer>' for issue 9326. thanks eliel ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-13 21:23:10 +00:00
Joshua Colp	4f04ff8597	Merged revisions 61648 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-13 17:21:53 +00:00
Joshua Colp	80ec0b13ba	Merged revisions 61641 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-13 16:35:33 +00:00
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) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-12 19:32:00 +00:00
Russell Bryant	3c0b24bda8	Merged revisions 61477 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-11 16:06:37 +00:00
Russell Bryant	6b033eea04	Merged revisions 61427 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines Merged revisions 61426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-11 15:13:12 +00:00
Russell Bryant	e34c67d308	Merged revisions 61377 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines Merged revisions 61376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-11 14:13:08 +00:00
Joshua Colp	9fff461080	Remove duplicate prototype declaration. (issue #9517 reported by junky) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-11 14:01:53 +00:00
Steve Murphy	ecaf781933	Merged revisions 60989 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-10 05:41:34 +00:00
Olle Johansson	b52f774850	Merged revisions 61072 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines Merged revisions 61038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-09 20:01:28 +00:00
Olle Johansson	4aef0155d6	use "ChannelType" in events to indicate which channel driver that generates the event. This replaces ... "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	2007-04-09 18:22:43 +00:00
Joshua Colp	a4bef3bb3a	Make RTP session ID and session version generation random. (issue #9456 reported by tjardick) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-09 12:33:49 +00:00
Joshua Colp	ed75ded048	Add counter for sip show registry CLI command. (issue #9352 reported by junky) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-09 00:47:06 +00:00
Olle Johansson	5bc2aa8ab1	Use the same parameter to the two "Registry" AMI events - ChannelDriver ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-06 19:26:01 +00:00
Joshua Colp	95a7dc0509	Merged revisions 60214 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, 05 Apr 2007) | 10 lines Merged revisions 60213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-05 12:57:35 +00:00
Russell Bryant	d1588ce2d5	Merged revisions 60112 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) | 3 lines Add a Content-Length of 0 to the response built by transmit_response_with_unsupported(). (issue #9454, reported by makoto, fixed by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-04 16:50:31 +00:00
Russell Bryant	bb53ef9d32	Merged revisions 60088 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60088 | russell | 2007-04-04 11:39:04 -0500 (Wed, 04 Apr 2007) | 12 lines Merged revisions 60083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines Fix the return value of handle_common_options() so that it always properly indicates whether it handled the option or not. (issue #9455, reported by Netview, fixed by me) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-04 16:40:01 +00:00
Russell Bryant	11ab6db24b	Merged revisions 59939 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59939 | russell | 2007-04-03 14:16:53 -0500 (Tue, 03 Apr 2007) | 12 lines Merged revisions 59938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines Don't attempt to report configuration errors in build_user(). oej pointed out that for a "friend" entry, this won't work, because all user options are valid for peers, but not the other way around. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-03 19:17:55 +00:00
Russell Bryant	3f14c8b6fc	Merged revisions 59936 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59936 | russell | 2007-04-03 13:55:57 -0500 (Tue, 03 Apr 2007) | 11 lines Merged revisions 59916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines Make chan_sip report when it encounters an unknown option. (issue #9440, reported by nightcrawler) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-03 18:57:52 +00:00
Russell Bryant	b908f9717a	Remove a duplicate function prototype. (issue #9444, junky) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-04-03 18:34:14 +00:00
Russell Bryant	93e2d66f13	Merged revisions 59262 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) | 3 lines Fix the check that ensures that the CHANNEL function's first argument is "rtpqos". Thanks, Corydon. :) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-27 18:18:36 +00:00
Russell Bryant	7c884d76ea	Merged revisions 59256 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines Convert the RTPQOS function to just be additional parameter of the CHANNEL function. This way, it will be possible for other RTP based channel drivers to expose this information in the future. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-27 16:25:02 +00:00
Tilghman Lesher	0e0600a446	Merged revisions 59227 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines Change this to a single dp function to make oej happy. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-26 21:44:59 +00:00
Russell Bryant	46b15992c7	Merged revisions 59209 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | 1 line Rename the new dialplan functions to match the variable name ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-26 17:57:50 +00:00
Russell Bryant	08e3a9bdc8	Merged revisions 59207 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-26 17:51:27 +00:00
Joshua Colp	cc22f60f30	Merged revisions 59195 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59195 | file | 2007-03-23 21:39:44 -0400 (Fri, 23 Mar 2007) | 10 lines Merged revisions 59194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines Only try to handle a response if it has a response code. (ASA-2007-011) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-24 01:42:11 +00:00
Kevin P. Fleming	e8e9e5e23c	Merged revisions 59182 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007) | 2 lines don't allow string input to overrun the buffer to hold it (ASA-2007-010) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-22 23:41:37 +00:00
Joshua Colp	af9c17025f	Minor tweak. Only queue up an unhold control frame if we are actually on hold. This would have shown itself when a call was initially being setup and the SDP data was being parsed in. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-21 03:33:57 +00:00
Joshua Colp	1d5be2d1c7	Merged revisions 59081 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2 lines Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-21 03:27:58 +00:00
Olle Johansson	dddb57b242	Merged revisions 59037 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3 lines Issue #9313, Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-18 20:39:37 +00:00
Russell Bryant	1bc728ed4b	Merged revisions 58906 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | 4 lines Some people like to put "limitonpeer" instead of "limitonpeers" in their configuration. While we're at it, support "limitonpeerz" and "limitonpeerssssss". (inspired by issue #9172) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-14 19:19:00 +00:00
Olle Johansson	f9c3f60ab9	Merged revisions 58848 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, 13 Mar 2007) | 10 lines Merged revisions 58847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2007-03-14 16:59:35 +00:00