6623 Commits (ab0b255455bcd8ccd78de24d61be8b1b1cf2f44d)

Author	SHA1	Message	Date
Richard Mudgett	ab0b255455	DNID not cleared when channel hang up (Affects PRI and SS7) ... The "dahdi show channels" CLI command still reports the DNID of the previous call even if the call is already hang up. The "dahdi show channels" command of older releases clear the DNID once the channel is hang up. Regression from the sig_analog/sig_pri extraction from chan_dahdi. (closes issue #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded by rmudgett (license 664) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	3819ba7ac7	update sip subscription debug message to a warning message ... If the Expire header of a SUBSCRIBE is less that our expiremin, a log warning will be displayed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d6e1c724e5	Remove built-in AES code and use optional_api instead ... Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	318798e932	send "423 Interval too small" Response to Subscribe with Expires less that min allowed ... [RFC3265]3.1.6.1.... The notifier MAY also check that the duration in the "Expires" header is not too small. If and only if the expiration interval is greater than zero AND smaller than one hour AND less than a notifier- configured minimum, the notifier MAY return a "423 Interval too small" error which contains a "Min-Expires" header field. The "Min- Expires" header field is described in SIP [1]. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	16b4813599	Fix invalid test for rxisoffhook in FXO channels ... This fixes some cases of no outgoing calls on FXO before an incoming call. Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO (KS/GS) channels.In some cases the bit would not be initialized properly before the first inbound call and thus prevent an outgoing call. If those tests are actually required by anybody, they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c . (closes issue #14577) Reported by: jkroon Patches: asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by frawd (license 610) Tested by: frawd Review: https://reviewboard.asterisk.org/r/699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	43b486453b	Properly set the port number for UDPTL media sessions. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a8c843199c	Change order so that it more closely matches the related SIP command. ... (closes issue #17648) Reported by: GMLudo Review: https://reviewboard.asterisk.org/r/789/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	d1b0bf0f2d	include stat.h for everybody, needed for device2chan ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7066a7f233	Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26791d5f8	fixes sip CANCEL race condition ... If Asterisk sends a 4xx error and the other side sends a CANCEl before receiving the 4xx and responding with the ACK, Asterisk will process the CANCEL and send a 487 Request Terminated as a new final response to the INVITE. Since we are issuing a new final response to the INVITE, the old one must be pretend_acked else it will keep retransmitting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	d27bb2d811	Only call ast_channel_cc_params_init() if allocating a channel succeeds. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cb5892bb67	Fix port setting of external address in SIP. ... There are two changes here: 1. Since the externip setting can now have a port attached to it, calling it "externip" is misleading. The option is now documented and parsed as "externaddr." This also extends to the "matchexterniplocally" setting. It is now documented and parsed as "matchexternaddrlocally." The old names for the options may still be used, but they are no longer used in the sip.conf.sample file. 2. If no port is set for the externaddr, and UDP is the transport to be used, then we will set the port of the externaddr to that of the udpbindaddr. This was how things worked prior to the IPv6 merge, so this is a regression fix. (closes issue #17665) Reported by: mmichelson Patches: 17665.diff#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	58061391a1	Fix regression with distinctive ring detection. ... The issue here is that passing an array to a function prohibits the ARRAY_LEN macro from returning the real size. To avoid this the size is now defined and use of ARRAY_LEN is avoided. (closes issue #15718) Reported by: alecdavis Patches: bug15718.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	6fa79e8f77	Make ACLs IPv6-capable. ... ACLs can now be configured to match IPv6 networks. This is only relevant for ACLs in chan_sip for now since other channel drivers do not support IPv6 addressing. However, once those channel drivers are outfitted to support IPv6 addressing, the ACLs will already be ready for IPv6 support. https://reviewboard.asterisk.org/r/791 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	5150954d4a	Merged revisions 277497 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested. FAX-128 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	34bc4b1dcb	Merged revisions 277419 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when reloading dahdi module During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	2289649901	Fix up some weird indentation problems in reqresp_parser.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	93373d7bdf	Formatting fixes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	cbe0a6dc02	Formatting changes (guideline corrections) ... Found a unused bag of curly brackets under my table. I always wondered where they had gone. They where indeed needed in chan_sip.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	e129b31fc6	Add ability to configure the Max-Forwards header in the dialplan, as well as in ... sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	dfba265a0b	Fix reversed logic of if statement. ... Found based on message from Philip Prindeville on the Asterisk Developers mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276909 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	44ae0033be	Correct not setting the bindport before attempting to open the socket. ... Related to changes from 276571, I was accidentally testing with a port set in my configuration causing me to miss this. Also moved the TCP handling as well to occur before build_peer is called. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	2b2a6123de	Fix MWI notification transmission problems over SIP. ... MWI updates were not being sent if no messages were found in the event cache. This was corrected since a phone may need to clear its MWI status configured previously from another mailbox. Upon module or sip reload, MWI updates could not be sent due to the sipsock socket not being set early enough in reload_config. The code handling the descriptor assignment and such has simply been moved before the call to build_peer. Issuing a sip reload cleared the IP address of the peer, but skipped checking the database for registration information. The database is now checked both for sip reload and actually reloading the module. If a transmission occurs before the do_monitor thread has started, do not attempt to send a signal to it. (closes issue #17398) Reported by: ip-rob git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	1e8c66e749	Fix errors where incorrect address information was printed. ... ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions) uses thread-local storage for storing the string that it creates. In cases where ast_sockaddr_stringify_fmt was being called twice within the same statement, the result of one call would be overwritten by the result of the other call. This usually was happening in printf-like statements and was resulting in the same stringified addressed being printed twice instead of two separate addresses. I have fixed this by using ast_strdupa on the result of stringify functions if they are used twice within the same statement. As far as I could tell, there were no instances where a pointer to the result of such a call were saved anywhere, so this is the only situation I could see where this error could occur. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	5d9aa45721	Make compile again. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d1c9a4b366	handle special case were "200 Ok" to pending INVITE never receives ACK ... Unlike most responses, the 200 Ok to a pending INVITE Request is acknowledged by an ACK Request. If the ACK Request for this Response is not received the previous behavior was to immediately destroy the dialog and hangup the channel. Now in an effort to be more RFC compliant, instead of immediately destroying the dialog during this special case, termination is done with a BYE Request as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is never received. The behavior of immediately hanging up the channel remains. This only affects how dialog termination proceeds for this one special case. RFC 3261 section 13.3.1.4 "If the server retransmits the 2xx response for 64*T1 seconds without receiving an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is accomplished with a BYE, as described in Section 15." git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id ... Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	e2599bc42c	collapse debug code in retrans_pkt into separate lines ... I've been working in this function a bunch lately, and these huge debug strings are getting annoying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	f4c665ee13	Do not skip sending MWI for a peer if an address is defined. Really just a merge mistake from IPv6 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring ... The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	23b6e621d2	chan_sip: RFC compliant retransmission timeout ... Retransmission of packets should not be based on how many packets were sent, but instead on a timeout period. Depending on whether or not the packet is for a INVITE or NON-INVITE transaction, the number of packets sent during the retransmission timeout period will be different, so timing out based on the number of packets sent is not accurate. This patch fixes this by removing the retransmit limit and only stopping retransmission after a timeout period is reached. By default this timeout period is 64*(Timer T1) for both INVITE and non-INVITE transactions. For more information on sip timer values refer to RFC3261 Appendix A. Review: https://reviewboard.asterisk.org/r/749/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	b42c6cab17	Revert early destruction of RTP sessions ... Some code improperly assumes that the sessions are still there, so revert the change until I can find all of them and fix them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	cb160a12b0	Destroy RTP fds when we schedule final dialog destruction ... Since we are only keeping the dialog around for retransmissions at this point and there is no possibility that we are still handling RTP, go ahead and destroy the RTP sessions. Keeping them alive for 32 past when they are used is unnecessary and can lead to problems with having too many open file descriptors, etc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6f8832735b	Don't try to ref authpeer when it isn't set ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	405d6cdf31	Add support for devices with less than 3 lines on the LCD. ... (closes issue #17600) Reported by: minaguib Patches: ast_unistim_height_v2.patch uploaded by minaguib (license 1078) Tested by: minaguib git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	7b1e28c6a1	Fix error in parsing SIP registry strings from ASTdb. ... It was essentially an off-by-one error. The easiest way to fix this was to use the handy-dandy AST_NONSTANDARD_RAW_ARGS macro to parse the pieces of the registration string out. Tested and it works wonderfully. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2fdf43f9fc	Get more information about the Bamboo test failures ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e46325f18c	Fix port parsing in check_via. ... If a Via header contained an IPv6 address, we would not properly parse the port. We would instead get the information after the first colon in the address. (closes issue #17614) Reported by: oej Patches: diff uploaded by sperreault (license 252) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	7e6f9b4e2d	Fix an issue where the port for p->ourip was being set to 0. ... This should fix all the CDR tests that were not passing. When they would originate a call, all fields in the INVITE that contained the source port would have the port set to 0. Most troubling of these was the Contact header. Tests are passing locally now and should also pass on the bamboo build agents. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	d2872c60e4	Merged revisions 275241 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul 2010) | 8 lines Fix logging message for stale nonce. (closes issue #17582) Reported by: kenner Patches: chan_sip.c.diff uploaded by kenner (license 1040) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d6011adab4	Weird, no output and Bamboo still fails... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	384681e182	Add some diagnostic feedback to our data tests ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	da8450323f	Kill some startup warnings and errors and make some messages more helpful in tracking down the source. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	5f92aed2ba	Return logic of sip_debug_test_addr() to its original functionality. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	0cc20f86ba	Fix sip_uri_parse test comparison. ... Part of the change with the IPv6 changes is to treat a host:port as a single 'domain' entity. This test was not updated to have the correct expectation after calling parse_uri(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
	53071af180	Copy the address into the peer structure after we set the default port ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c0fd67750b	Fix calls of ast_sockaddr_from_sin() from IPv6 integration. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. ... This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	a1b89a6a50	Implement AstData API data providers as part of the GSOC 2010 project, ... midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago