asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	60f80e5360	Merged revisions 229167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229167 \| dvossel \| 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) \| 9 lines don't crash on log message in solaris AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: bklang ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2cc2bade4b	Reverted revision 201717. (closes issue 0016175) Reported by: paul-tg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	d6b5df8715	Don't crash when bridge->tech_pvt == NULL This is a similar solution to what is in place for chan_agent (closes issue #16003) Reported by: atis Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	182ac0c503	Don't try to convert a 64-bit integer, where only a 32-bit integer is stored. (closes issue #16194) Reported by: habile git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c0b3c923a4	Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * cdr_mysql stored a pointer that was freed by realloc() * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. * The event subsystem sometimes creates an event with no IEs. Due to a corner condition, the code would read beyond the memory boundary. * res_pktccops did not correctly check whether its monitor thread was started. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	20e56c9d36	Created standard location to add options to chan_dahdi for ISDN dialing. Dial(DAHDI/g1[/extension[/options]]) Current options: K(<keypad_digits>) R Reverse charging indication (Collect calls) The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format was variable and did not allow for the easy addition of more options. The earlier 'C' prefix character for reverse charge indiation would conflict with the a-d DTMF digits if ISDN uses them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c17525391b	Missed these two channel drivers on the codec_bits merge git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	c205958f4c	Merged revisions 228547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228547 \| file \| 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) \| 4 lines Don't overwrite caller ID name on a trunk with the configured fullname when using users.conf (issue ABE-1989) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	45ad82fda3	Merged revisions 228078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 \| dbrooks \| 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) \| 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	5909d87926	Merged revisions 228079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228079 \| qwell \| 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) \| 8 lines Fix crash on VPB exception when no hardware is present. (closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	b3bd43366f	Modify the SDP parsing code to parse session and media level items separately. With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd Tested by: frawd, mnicholson, file Review: https://reviewboard.asterisk.org/r/414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	45f0f0cfef	Merged revisions 227700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227700 \| file \| 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) \| 5 lines Fix a security issue where sending a REGISTER with a differing username in the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	562a18f533	fix trunk building git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2bbda7a7c8	Two other trunk build fixes (reported by seanbright on #asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	a0198fc3ee	Resolve some dev-mode warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	4b69c3af69	Fixed a spelling error in the q850 reason header option in the output of sip show settings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b9ee743610	Code guidelines fixes only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8cd25fc043	user.conf entries in SIP were not having their peer type set. (closes issue #16120) Reported by: jsmith git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	ede3699c6e	Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 \| oej \| 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) \| 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	66579d9d49	Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networks (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: https://origsvn.digium.com/svn/asterisk/trunk@227049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	2c4d3b3168	SIP channel name uniqueness SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6406f39594	DAHDI ISDN channel names will not allow device state to work. (Interim solution.) Since ISDN works like SIP and not analog ports in regard to devices, the device state based on the ISDN channel number could not work. This has not been an issue until the advent of PTMP NT mode. Previously, ISDN lines were used as trunks and did not have to keep track of specific devices. As an interim solution until device states are properly implemented, the channel name is being changed to the following format to use the generic device state support: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will work with the following restrictions: * The number of devices/phones cannot exceed the number of B channels. (i.e., BRI has 2) * Each device/phone can only have one number. No shared MSN's. * The phones/devices probably should not use subaddressing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	93e43578ec	This patch adds support for a draft proposal for adding Q.850 reason headers 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: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	7fbd314a88	Cleanup some flags on DAHDI PRI channel hangup. * Cleanup some flags on DAHDI PRI channel hangup. (sig_pri split) * Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. * Remove some unused flags since sig_pri was split. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	b9c370da86	Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 \| file \| 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) \| 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	64e8fb465b	Doxygen documentation update git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	5825f68e8b	Add support for receiving unsolicited MWI NOTIFY messages. 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: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ea8b54fb9d	Fix building in REF_DEBUG mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ec0a1882c9	ACL check not present for verifying SIP INVITEs The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	71452322a2	Make conditionals create previous code when libpri/ss7 are present. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	2736168a6b	span numbers in pri debug / error messages Prefix PRI trace messages with the span number. This makes the trace readable even when you have a multi-port device. (closes issue #15054) Reported by: tzafrir Patches: dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	e5a57959eb	Re-arange code a bit to build in dev-mode without ss7 No change of functionality here. Just localized a variable and indented code into blocks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	d36cecd578	Make chan_dahdi build even without PRI / SS7 (Note: still some strange build warnings without SS7 in dev-mode) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	fb0196fce6	Improve performance of pedantic mode dialog searching in chan_sip. This patch changes chan_sip to use the new astobj2 OBJ_MULTIPLE iterator support to make pedantic mode dialog searching in find_call() not require a linear search of all dialogs in the list of dialogs. This patch does not change the dialog matching logic (more on that later), just improves the searching performance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	cff6d02b53	Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support. * 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: https://origsvn.digium.com/svn/asterisk/trunk@225692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2208fb171b	Fixes an iterator memory leak and uninitialized memory git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	63473616da	Search for the subaddress only within the extension section of the dial string. Dial(DAHDI/(g\|G\|r\|R)<group#(0-63)>[c\|r<cadance#>\|d][/extension]) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	776a14386a	SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1174a61612	Add support for calling and called subaddress. Partial support for COLP subaddress. The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3acfd4933c	Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 \| dvossel \| 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) \| 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	87ff40d3f3	Add 'mohsuggest' configuration option to 'sip show peer' CLI command and SIPShowPeer AMI action. (closes issue #15990) Reported by: _brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388) Review: https://reviewboard.asterisk.org/r/381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	01ab66275a	Add support for specifying the IP address to use for media streams in sip.conf 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: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	a31eb5bb35	Revert media_address commit, I'm going to roll a fix to the SDP generation in the next version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	984d6500ce	Merged revisions 225032 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	28d0ec5421	Add support for specifying the IP address to use for media streams in sip.conf (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c5b8e9af7c	Make PRI_SUBCMD_xxx handling subaddress friendly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	26638d3a55	Add dynamic range compression support for analog channels. (closes issue AST-29) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	fd9f9ab01e	Add a callback to sig_pri which is called when sig_pri is going to queue a control frame on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	03db5ef0e5	fix typo, sorry git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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6310 Commits (8e583db28f3d84ec7af52de732e7107ff3ded4f5)