asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	97d93fbfca	Make the default extconfig.conf match entries with the sample res_mysql.conf. This eliminates a future source of possible confusion with the configuration of 1.6.1 and higher. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	5583a4e955	This patch adds support for choosing a realm based on the domain in the From or To header in the incoming request. Eligible domains are taken from the domains list in the config file. This functionality is enabled when domainsasrealm is enabled in the config file. (closes issue #11361) Reported by: arkadia Patches: sip_realm_mnich_to_added_2.patch uploaded by arkadia (license 233) Tested by: arkadia git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e9d22f802e	Rename 'canreinvite' option to 'directmedia', with backwards compatibility. It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	d0664ba6af	Add an 'sms' option to mobile.conf to manually enable or disable SMS support. (closes issue #15071) Reported by: ughnz Patches: optional-sms1.diff uploaded by mnicholson (license 96) Tested by: ughnz, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	c058252718	Add configuration sample code for previous commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ba8dcde549	Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 \| mmichelson \| 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) \| 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	126bf8eeb5	add default alias reload to run module reload. Requiring 'module reload' to reload everything, including core etc makes russell very unhappy. The default configuration already loads the 'friendly' aliases template. Added 'reload=module reload' to that template. Also removed the comment in main/cli.c that reload should come back. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	496b509c42	Update some missing allowed options for overlapdial git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8bf870e4af	Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 \| dvossel \| 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) \| 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	9d9a8a4fa3	fix a typo in sample config file for option change git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	719917fe59	Support setting and receiving Reverse Charging Indication over ISDN PRI. This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse Charging Indication in LibPRI. This patch adds the ability to specify RCI on the outbound leg of a PRI call from within Asterisk, by prefixing the dialed number with a capital 'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)}) Thanks again to rmudgett for the thorough review. (closes issue #13760) Reported by: mrgabu Review: https://reviewboard.asterisk.org/r/303/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Ryan Brindley	d92d4d21d6	- cfgbasic.html has been replaced by index.html in the GUI for some time now	16 years ago
Russell Bryant	37ddf46a40	Rename res_config_sqlite.conf to res_config_sqlite.conf.sample (missing .sample). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1ae0291374	Rename ooh323.conf to chan_ooh323.conf, make module support both names git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	564b7aa848	Rename mobile.conf to chan_mobile.conf, make module support old name, too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d806ae0da0	Rename res_mysql.conf to res_config_mysql.conf, make module support both git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	65317d3861	Rename mysql.conf to app_mysql.conf, make module support both names git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c511a26749	Move Asterisk-addons modules into the main Asterisk source tree. Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	e840307ad1	Reorganize this adaptive CEL config a bit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	caa71e6f0d	Add common headers to CEL related configs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f2a94ef51c	Remove invalid entries in the config. This might seem like a legitimate comment that merely needed semicolon prefixes, but in reality, the adaptive layer is designed to allow arbitrary CDR variables, without needing the use of a userfield to store multiple items. It's therefore not only invalid syntax but also goes against the intent of the adaptive method. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a4284a507b	Add a new module, cdr_syslog, which allows writing CDRs to syslog. The original patch for this was written by Brett Bryant, and I split it out into it's own module. (closes issue #12876) Reported by: bbryant Patches: 06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36) 05212009_cdr_syslog.patch uploaded by seanbright (license 71) Tested by: seanbright Review: https://reviewboard.asterisk.org/r/297/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	48f7381af0	Fix the 'nat' option to actually do RFC3581 as expected and extend the configurable values for finer control. (closes issue #8855) Reported by: mikma Tested by: klaus3000, file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	59c1998d67	Improve T.38 negotiation by exchanging session parameters between application and channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
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

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1230 Commits (cc21007de174e228efa1a349e0cada08be7537ca)