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2009-11-23Change fax detection in chan_sip so it behaves as one would expect.file1-1/+1
Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230881 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Display a list of channel variables in each channel-oriented event.tilghman1-0/+6
(Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Add the capability to require a module to be loaded, or else Asterisk exits.oej1-0/+8
Review: https://reviewboard.asterisk.org/r/426/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229819 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-12Update sample config for ALSA mute and noaudiocaptureqwell1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229754 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-12Update sip.conf.sample.lmadsen1-5/+5
Just updating a spelling error and some capitalization in a documentation update that Olle added. May the Swenglish be with you. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229639 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-12Clarificationoej1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229607 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-12Clarify some security issues early in the sample configurationoej1-0/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229606 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-09Add the 'relative-periodic-announce' option to app_queue to allow for ↵mnicholson1-0/+6
calculating the time of announcments from the end of the previous announcment rather than from the beginning. (closes issue #15260) Reported by: tonils git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228947 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03This patch adds a sequence field to CDRs that can be combined with the ↵mnicholson1-1/+1
linkedid or uniqueid field to uniquely identify a CDR. (closes issue #15180) Reported by: Nick_Lewis Patches: cdr-sequence10.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227435 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Add support for using a hint when configuring a state interface using the ↵file1-0/+4
format hint:<extension>@<context>. (closes issue #15168) Reported by: p_lindheimer Patches: queue_extenstate5_1.4.svn.patch uploaded by GameGamer43 (license 894) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227424 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Additional fixes to the extensions.conf.sample file.lmadsen1-12/+12
Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227361 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Update extensions.conf.sample file to fix incorrect extensions.lmadsen1-3/+3
(closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227162 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networkstilghman2-16/+70
(closes issue #12950) Reported by: alea-soluciones Patches: ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones (license 514) Tested by: alea-soluciones, adomjan, urtho, nahuelgreco git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227049 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02This patch adds support for a draft proposal for adding Q.850 reason headers ↵mnicholson1-0/+5
to sip messages. (closes issue #13385) Reported by: adomjan Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487) sip-q850-hangupcause1.diff uploaded by mnicholson (license 96) Tested by: adomjan git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226687 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Merged revisions 226382 via svnmerge from lmadsen1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226384 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-27Add support for receiving unsolicited MWI NOTIFY messages.file1-0/+5
This change adds a configuration option to SIP peers, unsolicited_mailbox, which configures a virtual mailbox to use for received new/old MWI information. This virtual mailbox can then be used by any device supporting MWI. (closes issue #13028) Reported by: AsteriskRocks Patches: bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226060 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-23Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support.rmudgett1-0/+6
* Added handling of received HOLD/RETRIEVE messages and the optional ability to transfer a held call on disconnect similar to an analog phone. * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP. Will reroute/deflect an outgoing call when receive the message. Can use the DAHDISendCallreroutingFacility to send the message for the supported switches. * Added ability to send/receive keypad digits in the SETUP message. Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} * Added support for BRI PTMP NT mode. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225692 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Permit storage of voicemail secrets in a separate file, located within the ↵tilghman1-3/+22
spool directory. (closes issue #14276) Reported by: klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65) Tested by: jamesgolovich git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225406 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Add support for specifying the IP address to use for media streams in sip.conffile1-0/+6
This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225089 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Revert media_address commit, I'm going to roll a fix to the SDP generation ↵file1-6/+0
in the next version. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225034 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 225032 via svnmerge from dvossel2-0/+19
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225033 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Add support for specifying the IP address to use for media streams in sip.conffile1-0/+6
(closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225003 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Add dynamic range compression support for analog channels.mnicholson1-0/+12
(closes issue AST-29) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224637 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-18Clarify that "forcecommit" is NOT an alias for "autocommit", but instead ↵tilghman1-0/+5
controls the default disposition of uncommitted transactions. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224446 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-15chan_dahdi.conf.sample changes for DTMF CID detect dbailey1-0/+11
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: http://svn.digium.com/svn/asterisk/trunk@224144 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-14Allow for adding message body to the SIP NOTIFY messagejpeeler1-13/+10
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: http://svn.digium.com/svn/asterisk/trunk@224035 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 optionsdvossel1-23/+40
SWP-151 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223756 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-10Adding note about TLS usageoej1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223415 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-10Add an additional note on TLS supportoej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223414 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-10Adding some information on TLS supportoej1-6/+18
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223413 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-07Remove 'keepstats' queue option from sample config, as it's no longer used.qwell1-5/+0
https://reviewboard.asterisk.org/r/115/ (closes issue #15820) Reported by: kshumard git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222548 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06contact header port ignored transport when using externipdvossel1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@222398 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-05Allow non-compliant T.38 endpoints to be supportable via configuration option.kpfleming2-12/+18
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: http://svn.digium.com/svn/asterisk/trunk@222110 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Remove ability to control T.38 FAX error correction from udptl.conf.kpfleming1-5/+0
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: http://svn.digium.com/svn/asterisk/trunk@221592 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221360 via svnmerge from mnicholson1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@221432 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221153,221157,221303 via svnmerge from mnick1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@221368 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221086 via svnmerge from twilson1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@221266 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-25Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channelsphsultan1-1/+4
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: http://svn.digium.com/svn/asterisk/trunk@220457 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-24Documentation in the commit messages is soon forgotten, please add it to the ↵oej1-0/+3
docs in the product. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220295 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-23Add support for 'setvar=' for MGCP device lines, like other channel drivers ↵tilghman1-0/+3
provide. (closes issue #14818) Reported by: alea-soluciones Patches: chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219952 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-16Merged revisions 219023 via svnmerge from tilghman1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@219061 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-14Recorded merge of revisions 218331 via svnmerge from tilghman1-1/+4
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: http://svn.digium.com/svn/asterisk/trunk@218361 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Allow multiple rows to be fetched within the normal mode of operation.tilghman1-4/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216846 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Update sip.conf.sample documentation, reorganize a bitoej1-53/+62
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216694 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-04Merged revisions 216430 via svnmerge from oej1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@216438 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merge code associated with AST-2009-006dvossel1-0/+57
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215955 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ↵rmudgett1-0/+8
ISDN PTMP CPE spans. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215757 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-26Minor punctuation change.rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@214272 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-21Merged revisions 213493 via svnmerge from qwell1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@213494 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19Better parsing for the "register" linetilghman1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@213098 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-29 04:22:41 +0000