asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	f3a0d35db2	Restore SIP DTMF overlap dialing method. The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@345275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	ef82f32ca3	Cleanup references to sipusers and sipfriends dynamic realtime families Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 ........ Merged revisions 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6fd0d3805d	Merged revisions 336658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336658 \| rmudgett \| 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) \| 31 lines Made Dial d and H options no longer immediately auto-answer the calling leg. The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	04224c2036	Merged revisions 336234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336234 \| seanbright \| 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) \| 2 lines Make a note that inotify won't work with an NFS mounted spooler directory. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	7c03cb5331	Merged revisions 329027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329027 \| pabelanger \| 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) \| 2 lines Asterisk now requires libpri 1.4.11+ for PRI support. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/2.0@329055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0096238b52	Merged revisions 320823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 \| rmudgett \| 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) \| 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	928ec2b990	Merged revisions 309445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 \| rmudgett \| 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) \| 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	2a0983d0c5	Merged revisions 294535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294535 \| russell \| 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) \| 5 lines Tweak a couple of CLI commands back to their original form. The "module" in this case is two parts, so there are two words before the verb of the CLI command. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	dd1e62c095	Merged revisions 287193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287193 \| russell \| 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) \| 4 lines Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf. Review: https://reviewboard.asterisk.org/r/922/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	bcf5988caf	Merged revisions 283493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 \| dvossel \| 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) \| 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	eca5209181	Merged revisions 282302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282302 \| dvossel \| 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) \| 10 lines remove current STUN support from chan_sip.c This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	1990c4347e	Merged revisions 281650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281650 \| russell \| 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) \| 5 lines Change the default value for alwaysauthreject in sip.conf to "yes". (closes issue #17756) Reported by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	a84347029b	Merged revisions 279689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279689 \| pabelanger \| 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) \| 2 lines Updated documentation for FAX logger level. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	4bd366a926	Merged revisions 279566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279566 \| pabelanger \| 2010-07-26 15:51:39 -0400 (Mon, 26 Jul 2010) \| 8 lines Add documentation for FAX logger level. (closes issue #17715) Reported by: vrban Patches: 17715.patch uploaded by pabelanger (license 224) Tested by: vrban ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c61b87c5f6	Shuffle UPGRADE.txt files for 1.10. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

15 Commits (e4c84b18762ddb452b214be9cbe448eb7c5a35bb)