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2009-07-11Add an API for reporting security events, and a security event logging module.Russell Bryant
This commit introduces the security events API. This API is to be used by Asterisk components to report events that have security implications. A simple example is when a connection is made but fails authentication. These events can be used by external tools manipulate firewall rules or something similar after detecting unusual activity based on security events. Inside of Asterisk, the events go through the ast_event API. This means that they have a binary encoding, and it is easy to write code to subscribe to these events and do something with them. One module is provided that is a subscriber to these events - res_security_log. This module turns security events into a parseable text format and sends them to the "security" logger level. Using logger.conf, these log entries may be sent to a file, or to syslog. One service, AMI, has been fully updated for reporting security events. AMI was chosen as it was a fairly straight forward service to convert. The next target will be chan_sip. That will be more complicated and will be done as its own project as the next phase of security events work. For more information on the security events framework, see the documentation generated from doc/tex/. "make asterisk.pdf" Review: https://reviewboard.asterisk.org/r/273/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10SIP register not using peer's outbound proxyDavid Vossel
If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Update comments about the level of T.38 support in Asterisk.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Merged revisions 205877 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Merged revisions 205804 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Eliminate extraneous LOG_DEBUG messages generated by app_fax.Kevin P. Fleming
The transmit_audio() and transmit_t38() functions in app_fax have processing loops that are supposed to wait for frames to arrive on the channel and then handle them, but they also have short timeouts so that the loops can have watchdog timers and do other required processing. This commit changes the loops to not actually call ast_read() and attempt to process the returned frame unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages and slightly improving performance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Merged revisions 205775 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-10Fix some remaining T.38 negotiation problems in app_fax.Kevin P. Fleming
Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09Fix mbl_fixup() in chan_mobile to update newchan->tech_pvt instead of oldchan.Matthew Nicholson
(closes issue #15299) Reported by: nikkk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.Kevin P. Fleming
Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09Convert func_odbc to use ast_dummy_alloc_channel()Matthew Nicholson
Review: https://reviewboard.asterisk.org/r/290/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09Merged revisions 205599 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09make this compile again under devmodeMichiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-09pthread_self returns a pthread_t which is not an unsigned int on allMichiel van Baak
pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Merged revisions 205471 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Fix a CEL related regression with hints updating by subscribing to ↵Matthew Nicholson
AST_DEVICE_STATE instead of AST_DEVICE_STATE_CHANGED. (closes issue #15440) Reported by: lmsteffan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Merged revisions 205409 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08missing comma in devstatestring arrayDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Merged revisions 205349 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines Prevent phantom calls to queue members. If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Merged revisions 205288 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Fixes Park() argument handlingDavid Brooks
Park() was not respecting the arguments passed to it. Any extension/context/priority given to it was being ignored. This patch remedies this. (closes issue #15380) Reported by: DLNoah git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Oops, fixing buildTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Merged revisions 205215 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Fix a few compilation problems found when building Asterisk against uClibc.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Merged revisions 205188 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines Add redirection warnings for the invalid language codes previously removed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Use tabs instead of spaces for indentation.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Blocked revisions 205149 via svnmergeRussell Bryant
........ r205149 | russell | 2009-07-08 10:54:21 -0500 (Wed, 08 Jul 2009) | 8 lines Make OpenSSL usage thread-safe. OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08Move OpenSSL initialization to a single place, make library usage thread-safe.Russell Bryant
While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-08FreeBSD now has autoconf 2.62 in the ports, 2.61 has disappeared.Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-07Permit setting custom headers from the peer definition.Tilghman Lesher
(closes issue #14059) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-07Fix a deadlock in sig_analogMatthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-06Add CEL transfer events to analog (chan_dahdi) transfers.Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-06Merged revisions 981 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk-addons/branches/1.4 ........ r981 | tilghman | 2009-07-06 16:30:13 -0500 (Mon, 06 Jul 2009) | 7 lines Don't reset reconnect time, unless a reconnect really occurred. (closes issue #15375) Reported by: kowalma Patches: 20090628__issue15375.diff.txt uploaded by tilghman (license 14) Tested by: kowalma, jacco ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-06Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for ↵Kevin P. Fleming
non-T.38-capable channels. This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-03Add a configure check for Reverse Charging Indication support in LibPRI.Sean Bright
Also go back and wrap all of the places that use the specific reverse charge APIs with preprocessor conditionals. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-03Wrap rtp_engine.h header comments to 80 characters.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02Merged revisions 204834 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-02Moved trigger for BRIDGE_END CEL event so that it is more accurate.Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204807 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
2009-07-02Merged revisions 204681 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204710 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
2009-07-01A bunch of CODING_GUIDELINES related fixes. Not even close to done.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Merged revisions 204556 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Remove an unnecessary #ifdefSean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Move the masquerade in local_attended_transfer to a point where we hold the ↵Mark Michelson
channel lock. Masquerading without the channel's lock held is a *horrible* idea. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Remove some bogus deadlock avoidance code from local_attended_transfer.Mark Michelson
First of all, the code was unnecessary. The goal was to lock a channel which was already locked. Second, the assumption of the deadlock avoidance loop was that the sip_pvt was already locked and we were trying to get the channel lock. The problem is that the sip_pvt was unlocked a few lines above. Basically, I'm removing 5 lines of no-op. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Merged revisions 204474 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30Recorded merge of revisions 204469 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204470 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
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-04 02:39:00 +0000