asterisk

Commit Graph

Author	SHA1	Message	Date
Jeff Peeler	e3f473f4f3	Allow for adding message body to the SIP NOTIFY message Ability has been added to both manager command SIPnotify as well as console command sip notify. Message body is stored in the "Content" variable. An example is present in sip_notify.conf. (closes issue #13926) Reported by: jthurman Patches: sip-notify-svn189463.diff uploaded by gareth (license 208) Tested by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ccdfe1c2d7	Updates CHANGES to reflect the new externtcpport and externtlsport sip options git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	dd0c76a9d3	Move DAHDI/ISDN channel naming note from CHANGES to UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9c05faf76d	Prevent deadlock if chan_dahdi attempts to change PRI channel names. The PRI channels can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ISDN-<span>-<sequence-number> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Philippe Sultan	b11b94a083	Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over XMPP to process calls. SendText can be used instead of JabberSend in the context of XMPP based voice channels (chan_gtalk and chan_jingle). (closes issue #12569) Reported by: eech55 Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo Review: https://reviewboard.asterisk.org/r/88/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c68a2d9d30	Add support for 'setvar=' for MGCP device lines, like other channel drivers provide. (closes issue #14818) Reported by: alea-soluciones Patches: chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cd88adfc6a	Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up. In addition, there's a bit of cleanup to the arguments and documentation, in which I discovered that the last feature added to this application duplicated an option (oops!) and changed that option so that it now works. (closes issue #14909) Reported by: junky Patches: __20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10) Tested by: amilcar, junky, flujan, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	75d8960740	Allow multiple rows to be fetched within the normal mode of operation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	595ab444af	Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ISDN PTMP CPE spans. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	eca8f9082c	Adding MUTEAUDIO() dialplan function and MuteAudio AMI action (pinepeach) Review: https://reviewboard.asterisk.org/r/345/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2cfddf8cb6	Add MASTER_CHANNEL() dialplan function, as well as a useful usage. (closes issue #13140) Reported by: cpina Patches: 20090807__issue13140.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen Change-type: feature git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	29e1e05e13	Add two new dialplan variables when using features Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature. Added DYNAMIC_PEERNAME which holds the unique channel name on the other side and is set when a dynamic feature is triggered. (closes issue #14663) Reported by: tamiel Patches: 20090313_features.diff uploaded by tamiel (license 712) Tested by: tamiel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	39c9838d77	Add COLP support to chan_dahdi/sig_pri. Add Connected Line Presentation (COLP) support to chan_dahdi/libpri as an addition to issue 8824. This is the chan_dahdi/sig_pri portion. COLP support is now available for any switch for which libpri supports COLP (currently ETSI PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/340/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	bcff592839	Merged 207316 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r207316 \| rmudgett \| 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) \| 20 lines Fixed incoming calls being matched to MSNs without type-of-number prefix added. For an incoming ISDN call the dialed.number is incorrectly matched against the configured MSNs in misdn.conf. The numbers passed to the dialplan include the configured prefix for the dialed.number_type, whereas the check against the configured MSNs (to decide if the call is accepted at all), is executed without the configured prefix. e.g., dialed.number = 241168020, TON = national, configured national prefix is "0". (This is the TON which is used by ISDN providers in the Netherlands.) In chan_misdn.c:cb_events() in case EVENT_SETUP the call to misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57 lines later the call to read_config() adds the prefix, and the dialed.number is now 0241168020, which is then used in the dialplan. misdn_cfg_is_msn_valid() must use the normalized number, too. JIRA ABE-1912 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c9aefb32a1	note the security events API in CHANGES git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 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	17 years ago
Sean Bright	088ea24f7c	Another CHANGES spelling fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	b7feca3685	Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203960 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
Russell Bryant	bb1fc3e12f	Add 's' option to ChanSpy, which makes the app exit when no channels are left to spy on. (closes issue #14594) Reported by: JimDickenson Patches: chanspy.diff uploaded by JimDickenson (license 710) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203842 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
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
Joshua Colp	ae87ba45b5	Add support for multicast RTP paging. (closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0bbd5c9424	Ignore voicemail messages that are just silence. (closes issue #2264) Reported by: pfn Patches: silent-vm-1.6.2.txt uploaded by pfn (license 810) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	25ccb5e850	Add note about the addition of calendar support git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201944 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
Kevin P. Fleming	5b79acdb42	Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200726 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
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	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
Kevin P. Fleming	d9d2779008	Add buffer and echo canceller control to CHANNEL() dialplan function for DAHDI channels Adds ability for CHANNEL() dialplan function, when used on DAHDI channels, to temporarily change the number of buffers and/or the buffer policy, and also to enable, disable, or switch the echo canceller between FAX/data and voice modes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 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
Richard Mudgett	d35fd35ae3	Outgoing PTP redirected calls did not wait for the COLR from the redirected-to party. For outgoing PTP redirected calls, you now need to use the inhibit(i) option on all of the REDIRECTING statements before dialing the redirected-to party. You still have to set the REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The PTP call will update the redirecting-to presentation when it becomes available and queue the redirecting update to the calling channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191175 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
Richard Mudgett	89d06c7759	Make PTP DivertingLegInformation3 message behavior closer to the specifications. * Wait for a DivertingLegInformation3 message after receiving a DivertingLegInformation1 message to complete the redirecting-to information before queuing a redirecting update to the other channel. * A DivertingLegInformation2 message should be responded to with a DivertingLegInformation3 when the COLR is determined. If the call could or does experience another redirection, you should manually determine the COLR to send to the switch by setting REDIRECTING(to-pres) to the COLR and setting REDIRECTING(to-num) = ${EXTEN}. * A DivertingLegInformation2 message must have an original called number if the redirection count is greater than one. Since Asterisk does not keep track of this information, we can only indicate that the number is not available due to interworking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190735 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
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 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	50ecc19ca0	change some capitalization git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188378 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
Jeff Peeler	de4af72f9f	Add ability for dialplan execution to continue when caller hangs up. The F option to app_dial has been modified to accept no parameters and perform the above functionality. I don't see anywhere else that is doing function overloading, but this really is the best place for this operation because: - It makes it close to the 'g' option in the argument list which provides similar functionality. - The existing code to support the current F option provides a very convienient location to add this new feature. (closes issue #12381) Reported by: michael-fig git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4eaa651a8a	Add support for changing the outbound codec on a SIP call using a dialplan variable. This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls the codec offered for an outgoing SIP call. This is much like the SIP_CODEC dialplan variable and has the same restrictions. The codec set must be one that is configured for the call. (closes issue #13243) Reported by: samdell3 Patches: 13243.diff uploaded by file (license 11) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186624 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
Mark Michelson	378c2e9d2a	Allow the AMI Hangup command to accept a Cause header. (closes issue #14695) Reported by: mneuhauser Patches: cause-for-hangup-manager-action.patch uploaded by mneuhauser (license 425) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185704 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
Russell Bryant	16fc1993ef	Add support for the "name" option in the CHANNEL() function. Review: http://reviewboard.digium.com/r/199/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	bf2895bae8	Fixing CHANGES in rev 182596. Progress DTMF was added into app_dial's D() option. In CHANGES it should have been updated under 1.6.3 rather than 1.6.2. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e559cae4ec	Option to send DTMF when receiving PROGRESS status The D() option in app_dial is only able to send DTMF after the call has been answered. A progress option has been added to D() to allow DTMF to be sent upon receiving PROGRESS. This allows DTMF to be sent before the call is answered. (closes issue #12123) Reported by: VoipForces Patches: app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419) dtmf_progress.patch uploaded by dvossel (license 671) Tested by: VoipForces, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5e256effa7	Update UPGRADE.txt and CHANGES for 1.6.3 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182362 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

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464 Commits (e3f473f4f3ab6871f123e12ab381e84b34c6b238)