asterisk

Commit Graph

Author	SHA1	Message	Date
Jeff Peeler	bbfe6967ab	Remove some unnecessary code and update sample config file with respect to GR-303. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	1fa4796b19	Update sample cdr_tds configuration to try and eliminate some confusion. Also change the preferred configuration option from 'hostname' (which was misleading because it didn't actually treat the value as a hostname) to 'connection' and added some verbage explaining that the user would need to refer to their freetds.conf file for those settings. 'hostname' was kept as a backwards compatible configuration parameter. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	68ba81dfe6	Add rtsavesysname to chan_iax chan_sip has an option to save the sysname on rtupdate. This patch copies that same logic to chan_iax. (closes issue #14837) Reported by: barthpbx Patches: iax2-rtsavesysname.patch uploaded by barthpbx (license 744) rt_iax.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	2c8cd1db92	keep backwards compatible chan_dahdi with older openr2 versions by not using the new skip category feature unless supported git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	b52abf3d21	added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	5fcf193d7b	Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	453a2f7331	Remove not used code in the Agent channel. This code was there because of the AgentCallbackLogin() application. ->loginchan[] member was only used by AgentCallbackLogin(). Agent where dumped to astdb if they where logged in using AgentCallbacklogin() so they are not being dumper anymore. Review: https://reviewboard.asterisk.org/r/267/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	58766cd2cf	Suggesting that only a single timing module be loaded is no longer necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f51bb019bb	Update references to bugs.digium.com and reviewboard.digium.com to the new URLs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	0941c2c32e	Make note of Exchange calendar support limitations git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	71a3a2ebf6	Add Calendaring support for Asterisk This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS Exchange calendars. Exchange support has only been tested on Exchange Server 2k3 and does not support forms-based authentication at this time (patches very welcome). Exchange support is also currently missing the ability to return a list of a meting's attendees (again, patches are very, very welcome). Features include: Querying a calendar for events over a specific time range Checking a calendar's busy status via the dialplan Writing calendar events via the dialplan (CalDAV and Exchange only) Handling calendar event notifications through the dialplan (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash Review: https://reviewboard.asterisk.org/r/58 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f22962a0c1	Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a7d813cae7	Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Gavin Henry	a5fc03b683	closes issue #15156 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	7d50dee3f8	Remove a file sample configuration file that is no longer used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6f80849582	Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in the sample configuration files. (closes issue #15207) Reported by: seandarcy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f50bb3bfa4	SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	df4dce6837	Rework the cdr_custom.conf.sample header a bit to reflect the changes in functionality (allowing multiple mappings). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7b4eeed257	Add basic support for handling connected line-related UPDATE requests. SIP purists may want to look the other way... When COLP/CONP support for SIP was committed, there was a condition under which Asterisk may transmit a SIP UPDATE in order to communicate the change in connected line information. The issue here is that while we could send a SIP UPDATE message, we were not prepared to receive such an UPDATE and would always responde with a 501 when we received an UPDATE. The situation was a bit rough. We really want to be able to receive UPDATEs having to do with connected line changes, but the amount of effort involved in properly supporting RFC 3311 was staggering. This commit represents a compromise. First, it was decided that it is important to only send a SIP UPDATE to an endpoint that is able to handle one. So, now we have added parsing of the Allow header into SIP. We store the allowed methods on SIP peers so that when we communicate with them, we already will know what we can and cannot send to them. We will parse the peer's allowed methods when he registers with us. If the peer is not the type to register with us, but the qualify option is enabled, then we will use the response to the OPTIONS request we send the peer to determine the peer's allowed methods. When the peer's registration expires, or when qualify deems the peer to be unreachable, we clear the allowed methods from the peer. For an actual call, we will copy the peer's allowed methods to the sip_pvt representing the call leg. If we are communicating with an endpoint which is not a peer, then we will just parse the Allow header from the first message we receive during the call and store the information in the sip_pvt. If, during communication with a peer, we receive a 501 response, then we will make sure to save the fact that we cannot use that method when communicating with that peer. Now, with all that infrastructure in place, the only actual place we use this information currently is when attempting to send a connected line change using an UPDATE request. If we cannot send the change immediately using an UPDATE, we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon as it is allowed. The second part of the changes here is for Asterisk to accept UPDATE requests that have connected line changes. Since we are not fully supporting RFC 3311, Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead, if you are communicating with what you know to be another Asterisk box, you may set the rpid_update parameter in sip.conf so that we will send UPDATEs to that Asterisk box. When we send a connected line update, we set a custom header called "X-Asterisk-rpid-update." On the receiving end, if Asterisk receives an UPDATE that does not have the "X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501 since media-changing UPDATEs are not supported. We should never get such UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow header. If the custom header is present in the received UPDATE, though, then we will check the incoming request for connected line updates and queue the update on the channel where the change occurred. ABE-1840 ABE-1822 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f223598207	Allow cdr_custom to write to multiple files instead of just one. Up to now, cdr_custom would only accept a single filename/format from cdr_custom.conf. This change allows you to specify multiple filename & format directives. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8b40aa0287	Merged revisions 194764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194764 \| russell \| 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) \| 2 lines Fix some spelling fail. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7872538b83	Add outgoing_colp misdn.conf port parameter. Select what to do with outgoing COLP information on this port. 0 - Send out COLP information unaltered. (default) 1 - Force COLP to restricted on all outgoing COLP information. 2 - Do not send COLP information. outgoing_colp=0 Also fixed sending the EctInform message so it always has the required redirectionNumber parameter when the status is active. JIRA ABE-1853 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	7893ab8fe7	Merged revisions 193193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193193 \| kpfleming \| 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) \| 7 lines Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f7e4f776ea	Ensure that by default only one console channel driver is loaded This configuration file was changed to ensure that only one console channel driver (chan_oss) is loaded by default, but the change would only work if chan_console was not built. Now it will work as expected; if chan_alsa or chan_console are built and installed, they will not be loaded unless explicity requested. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a3af213506	Remove rarely-used event_log/LOG_EVENT support In discussions today at the Europe Asterisk Developer Meet-Up, we determined that the event_log was used in only 9 places in the entire tree, and really was not needed at all. The users have been converted to use LOG_NOTICE, or the messages have been removed since other messages were already in place that provided the same information. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
TransNexus OSP Development	8612c7ac8a	Made security features optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a6adc84e69	SIP option to specify outbound TLS/SSL client protocol. chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip. (closes issue #14770) Reported by: TheOldSaint (closes issue #14768) Reported by: TheOldSaint Review: http://reviewboard.digium.com/r/240/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ca138fc807	Consistent SSL/TLS options across conf files ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though. Review: http://reviewboard.digium.com/r/237/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3b68be6aaa	Remove nonexistent option from sip.conf.sample. The option to choose which connected line header to use is not 'rpid_header' but 'sendrpid' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8f0b88c8c8	TLS/SSL private key option Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	6bb2b6c096	Added CCBS/CCNR Party A support and enhanced COLP support. This change adds the following features to chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. * Enhances COLP support for call diversion and explicit call transfer. These enhanced features require a modified version of mISDN. The latest modified mISDN v1.1.x based version is available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged versions of the modified mISDN code are available under: http://svn.digium.com/svn/thirdparty/mISDN/tags http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review: http://reviewboard.digium.com/r/218/ Merged from team/rmudgett/misdn_facility branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	1172c38647	Add service maintenance message support This is the companion commit to libpri r732. Service messages are now supported for switch types 4ess/5ess. A new option service_message_support has been added to chan_dahdi.conf and is noted in the sample config file. The service message support is turned off by default. The current implementation relies on AstDB to keep track of channel state, which allows the statuses to be preserved across Asterisk restarts. Below is a description of the storage format. The state and reason for the service state are in the form <state>:<reason>, where: <state> ::= { 'O' } // 'O' – Out Of Service <reason> ::= { '0' \| '1' \| '2' \| '3' }, where: '0' – No reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3' – both NEAR and FAR END The new CLI commands to handle channel service state are: pri service disable channel <chan> pri service enable channel <chan> Many people contributed to the development of this functionality. Because I entered at the very end I do not know the exact history. Special thanks to all who moved the bug forward one way or another: cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman, lmadsen, and especially dhubbard (he answered lots of my questions and did a large portion of the work) (closes issue #3450) Reported by: cmaj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2f048030bd	revert addition of LOG_SECURITY log channel; after further discussion, a much better solution will be used git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4d74179f20	Add a new option, mwi_from, to sip.conf. This allows for you to change the From header for outgoing MWI NOTIFY requests. Prior to this, the best you could do was to set a callerid in the general section of sip.conf. The problem was that this was used for all outbound requests, not just MWI NOTIFY requests. AST-201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b5f8c632df	add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts (inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	06061491ba	Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 \| tilghman \| 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) \| 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dababe2148	Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 \| mmichelson \| 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) \| 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08971ce205	Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 \| tilghman \| 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) \| 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 \| tilghman \| 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) \| 2 lines Fix for AST-2009-003 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	9fd753a30e	Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 \| rmudgett \| 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) \| 1 line Update the channel allocation method documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	da2230adf0	SIP preferred codec only feature Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use. (closes issue #12485) Reported by: bamby Review: http://reviewboard.digium.com/r/206/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3fd19b3ab6	Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 \| tilghman \| 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) \| 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	77a6840fd3	Add MFC/R2 support for chan_dahdi. This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1ae8e9f3b	Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e69803a2be	Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 \| mmichelson \| 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) \| 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3a14487abf	Allow for "magic" pickups to work when we wish to ignore the context When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8970f8caaa	Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 \| mmichelson \| 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) \| 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d2c5b0f1de	Mark res_ais as experimental, as the binary event format is subject to change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	ec6101595e	Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 \| murf \| 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) \| 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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