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2009-10-18Clarify that "forcecommit" is NOT an alias for "autocommit", but instead ↵Tilghman Lesher
controls the default disposition of uncommitted transactions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-15chan_dahdi.conf.sample changes for DTMF CID detect Doug Bailey
Explains new options for detecting DTMF CID on fxo lines (issue #9096) Reported by: fleed Patches: chan_dahid_sample_config.patch uploaded by sum (license 766) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-14Allow for adding message body to the SIP NOTIFY messageJeff Peeler
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
2009-10-12Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 optionsDavid Vossel
SWP-151 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-10Adding note about TLS usageOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-10Add an additional note on TLS supportOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-10Adding some information on TLS supportOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-07Remove 'keepstats' queue option from sample config, as it's no longer used.Jason Parker
https://reviewboard.asterisk.org/r/115/ (closes issue #15820) Reported by: kshumard git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-06contact header port ignored transport when using externipDavid Vossel
This patch adds support for TCP/TLS in the Contact header when using NAT, specifically externip or externhost. The original issue was that Asterisk sent 5060 as the port in the contact header whether TLS was used or not. Additionally, this patch adds 2 config options to sip.conf, specifically externtcpport and externtlsport. This allows a user to specify different external ports for TCP and TLS other than those used internally, this is especially useful in in a PAT/port redirection setup. Thanks to ebroad for reporting the issue and providing the patch! (closes issue #15880) Reported by: ebroad Patches: portmap.patch uploaded by ebroad (license 878) externtXXport_v2.patch uploaded by ebroad (license 878) Tested by: ebroad Review: https://reviewboard.asterisk.org/r/392/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-05Allow non-compliant T.38 endpoints to be supportable via configuration option.Kevin P. Fleming
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-01Remove ability to control T.38 FAX error correction from udptl.conf.Kevin P. Fleming
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Merged revisions 221360 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Merged revisions 221153,221157,221303 via svnmerge from Matthias Nick
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-30Merged revisions 221086 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-25Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channelsPhilippe Sultan
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
2009-09-24Documentation in the commit messages is soon forgotten, please add it to the ↵Olle Johansson
docs in the product. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-23Add support for 'setvar=' for MGCP device lines, like other channel drivers ↵Tilghman Lesher
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
2009-09-16Merged revisions 219023 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-14Recorded merge of revisions 218331 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Allow multiple rows to be fetched within the normal mode of operation.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Update sip.conf.sample documentation, reorganize a bitOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04Merged revisions 216430 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Merge code associated with AST-2009-006David Vossel
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-02Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ↵Richard Mudgett
ISDN PTMP CPE spans. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-26Minor punctuation change.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-21Merged revisions 213493 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | 5 lines Clarify queues.conf comments to specify that variables should be set in the dialplan. (closes issue #15755) Reported by: trendboy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-19Better parsing for the "register" lineTilghman Lesher
Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-18Make the default extconfig.conf match entries with the sample res_mysql.conf.Tilghman Lesher
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
2009-08-12This patch adds support for choosing a realm based on the domain in the From ↵Matthew Nicholson
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
2009-08-03Rename 'canreinvite' option to 'directmedia', with backwards compatibility.Kevin P. Fleming
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
2009-08-03Add an 'sms' option to mobile.conf to manually enable or disable SMS support.Matthew Nicholson
(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
2009-07-31Add configuration sample code for previous commit.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27Merged revisions 209131 via svnmerge from Mark Michelson
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
2009-07-25add default alias reload to run module reload.Michiel van Baak
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
2009-07-17Update some missing allowed options for overlapdialJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-16Merged revisions 206872 via svnmerge from David Vossel
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
2009-07-14fix a typo in sample config file for option changeJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02Support setting and receiving Reverse Charging Indication over ISDN PRI.Sean Bright
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
2009-07-01- cfgbasic.html has been replaced by index.html in the GUI for some time nowRyan Brindley
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Rename res_config_sqlite.conf to res_config_sqlite.conf.sample (missing ↵Russell Bryant
.sample). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Rename ooh323.conf to chan_ooh323.conf, make module support both namesRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Rename mobile.conf to chan_mobile.conf, make module support old name, tooRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Rename res_mysql.conf to res_config_mysql.conf, make module support bothRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Rename mysql.conf to app_mysql.conf, make module support both namesRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Move Asterisk-addons modules into the main Asterisk source tree.Russell Bryant
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
2009-06-29Reorganize this adaptive CEL config a bit.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-29Add common headers to CEL related configs.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-29Remove invalid entries in the config.Tilghman Lesher
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
2009-06-26Add a new module, cdr_syslog, which allows writing CDRs to syslog.Sean Bright
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
2009-06-26Fix the 'nat' option to actually do RFC3581 as expected and extend the ↵Joshua Colp
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
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