asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	8ba7ff54b4	Merged revisions 298539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298539 \| tilghman \| 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) \| 8 lines Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. (closes issue #18464) Reported by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG (license 20) (plus a few additional lines by tilghman) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	fe98e1bcd6	Post AMI hold events on PRI spans when the remote party HOLD/RETRIEVEs the call. Part of JIRA SWP-2687/ABE-2691. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7f29edd140	Merged revisions 298195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298195 \| rmudgett \| 2010-12-13 11:11:43 -0600 (Mon, 13 Dec 2010) \| 33 lines Merged revisions 298194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298194 \| rmudgett \| 2010-12-13 11:04:41 -0600 (Mon, 13 Dec 2010) \| 26 lines Merged revisions 298193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298193 \| rmudgett \| 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) \| 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered transfers. Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING message is not received. The debug output shows that the DTMF begin event is seen, but the DTMF end event is missing. When the DTMF begin happens, the call is muted so we now have one way audio (until a DTMF end event is somehow seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin and DTMF end events if we are overlap dialing and have not seen a PROCEEDING message. * Added a debug message when absorbing a DTMF event. JIRA SWP-2690 JIRA ABE-2697 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	30f81f902d	Merged revisions 297965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297965 \| twilson \| 2010-12-09 16:18:19 -0600 (Thu, 09 Dec 2010) \| 28 lines Merged revisions 297960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297960 \| twilson \| 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) \| 21 lines Merged revisions 297959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 \| twilson \| 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) \| 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	316add7f12	Merged revisions 297957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297957 \| dvossel \| 2010-12-09 15:32:20 -0600 (Thu, 09 Dec 2010) \| 11 lines Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (closes issue #18412) Reported by: nevermind_quack Patches: fix uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	537d235460	Merged revisions 297607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297607 \| jpeeler \| 2010-12-06 16:06:37 -0600 (Mon, 06 Dec 2010) \| 25 lines Merged revisions 297605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297605 \| jpeeler \| 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) \| 18 lines Merged revisions 297603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 \| jpeeler \| 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) \| 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	df87ec438c	Merged revisions 297535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297535 \| seanbright \| 2010-12-03 12:41:30 -0500 (Fri, 03 Dec 2010) \| 9 lines Merged revisions 297534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r297534 \| seanbright \| 2010-12-03 12:40:52 -0500 (Fri, 03 Dec 2010) \| 3 lines The CLI command should not contain <placeholder>s, these are for descriptions. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a46bd43ae8	Merged revisions 297075 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297075 \| jpeeler \| 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) \| 37 lines Merged revisions 297073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297073 \| jpeeler \| 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) \| 30 lines Merged revisions 297072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 \| jpeeler \| 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) \| 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	597e913cd2	Merged revisions 296951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296951 \| tilghman \| 2010-11-30 19:46:32 -0600 (Tue, 30 Nov 2010) \| 9 lines Merged revisions 296950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296950 \| tilghman \| 2010-11-30 19:38:19 -0600 (Tue, 30 Nov 2010) \| 2 lines Missed initializations caused startup errors on Mac OS X (and possibly others, too). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	bd6f29dcb9	Merged revisions 296673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296673 \| pabelanger \| 2010-11-29 18:05:45 -0500 (Mon, 29 Nov 2010) \| 19 lines Merged revisions 296671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296671 \| pabelanger \| 2010-11-29 17:54:14 -0500 (Mon, 29 Nov 2010) \| 12 lines Merged revisions 296670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296670 \| pabelanger \| 2010-11-29 17:49:39 -0500 (Mon, 29 Nov 2010) \| 5 lines Make sure nothing else is needed before destroying the scheduler. (closes issue #18398) Reported by: pabelanger ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	40cc550f1f	Merged revisions 296628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296628 \| russell \| 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) \| 6 lines Complete some error handling in transmit_publish() in chan_sip.c. This error handling block caught my eye. It was missing a couple of things, but it should be safe now. Thanks to mmichelson for the quick peer review on IRC. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	267cf27744	Merged revisions 296582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296582 \| rmudgett \| 2010-11-29 14:46:03 -0600 (Mon, 29 Nov 2010) \| 24 lines Merged revision 296575 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r296575 \| rmudgett \| 2010-11-29 14:27:37 -0600 (Mon, 29 Nov 2010) \| 13 lines Invalid mISDN PTMP redirecting signaling as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY redirecting number and notification code, SETUP redirecting number) is also sent in PTMP/TE mode. It should only apply in PTMP/NT mode. The call setup proceeds but the network (Deutsche Telekom) reacts with ugly ISDN STATUS messages. Also don't send the redirecting number ie when PTP is also sending the DivertingLegInformation2 facility. The redirecting number ie is redundant and the network (Deutsche Telekom) complains about it. Patches: abe_2651_v4.patch uploaded by rmudgett (license 664) JIRA ABE-2651 JIRA SWP-2537 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brad Watkins	ad56a4d16e	Merged revisions 296352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296352 \| marquis \| 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010) \| 12 lines Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. This had the effect of sending one NOTIFY for every time a sip peer made a call, in one case eventually overwhelming the phone and causing it to reboot. (closes issue #18342) Reported by: nivek Patches: issue0018342p1.patch uploaded by nivek (license 636) Tested by: nivek Review: https://reviewboard.asterisk.org/r/1029/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ccdc417ab5	Merged revisions 296167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296167 \| rmudgett \| 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) \| 57 lines Merged revisions 296166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296166 \| rmudgett \| 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) \| 50 lines Merged revisions 296165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296165 \| rmudgett \| 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) \| 43 lines Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip. The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b1e7f85bce	Merged revisions 295747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295747 \| rmudgett \| 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010) \| 13 lines One way audio before answering call waiting call on analog port. * Analog call waiting Caller ID spills could get stuck resulting in one way audio until the waiting call is answered. This only happens on the second (and later) call waiting call if the active call is not the first call. * The CLI/AMI "dahdi show channel" command could report the wrong channel information. Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer in sync. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	e5ede71934	Merged revisions 295673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295673 \| twilson \| 2010-11-19 14:06:10 -0800 (Fri, 19 Nov 2010) \| 22 lines Merged revisions 295672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295672 \| twilson \| 2010-11-19 13:55:48 -0800 (Fri, 19 Nov 2010) \| 15 lines Merged revisions 295628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 \| twilson \| 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) \| 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f6edd47dd6	Merged revisions 295516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295516 \| rmudgett \| 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010) \| 13 lines Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed initial value of struct analog_pvt.use_callerid. It may get forced on depending upon other config options. * Call analog_dnd() instead of manual inlined code. * Removed unused struct analog_pvt.usedistinctiveringdetection. * Removed the struct analog_pvt.unknown_alarm flag. It was really the struct analog_pvt.inalarm flag. * Use ast_debug() instead of ast_log(LOG_DEBUG). * Rename several function's index variable to idx. * Some formatting tweaks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	5d1cd7863a	Merged revisions 294823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294823 \| rmudgett \| 2010-11-11 20:45:22 -0600 (Thu, 11 Nov 2010) \| 25 lines Merged revisions 294822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294822 \| rmudgett \| 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) \| 18 lines Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 \| rmudgett \| 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) \| 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	99a698efb7	Merged revisions 294734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294734 \| jpeeler \| 2010-11-11 15:58:25 -0600 (Thu, 11 Nov 2010) \| 32 lines Merged revisions 294733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 \| jpeeler \| 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) \| 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 \| jpeeler \| 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) \| 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	3adb425b25	Merged revisions 294349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294349 \| rmudgett \| 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) \| 17 lines Analog lines do not transfer CONNECTED LINE or execute the interception macros. Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	2df9e23e35	Merged revisions 294243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294243 \| mnicholson \| 2010-11-08 14:56:30 -0600 (Mon, 08 Nov 2010) \| 15 lines Merged revisions 294242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 \| mnicholson \| 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) \| 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	18553bb804	Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 \| rmudgett \| 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) \| 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ******** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ******** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	bbffb7fb07	Merged revisions 294084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294084 \| bbryant \| 2010-11-05 18:03:11 -0400 (Fri, 05 Nov 2010) \| 9 lines Fixed deadlock avoidance issues while locking channel when adding the Max-Forwards header to a request. (closes issue #17949) (closes issue #18200) Reported by: bwg Review: https://reviewboard.asterisk.org/r/997/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	97a1489960	Perform proper handling of forked outbound INVITE requests. RFC3261 section 12 about dialog creation says an INVITE transaction results in an established dialog once it receives the 200 OK response. It is possible to receive multiple differing 200 OK responses for a single outbound INVITE Request, and this should result in establishing multiple dialogs. This patch allows for all differing 200 OK responses to an INVITE request to establish a separate dialog, but only the first dialog is kept. All other resulting dialogs from the initial request are immediately ACKed and then immediately terminated with a BYE request. Review: https://reviewboard.asterisk.org/r/946/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	f38f888416	Merged revisions 293924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293924 \| dvossel \| 2010-11-04 16:39:51 -0500 (Thu, 04 Nov 2010) \| 4 lines Fixes ringback tone on sip semi-attended transfer. ABE-2168 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	dcd6dae413	Merged revisions 293887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293887 \| pabelanger \| 2010-11-04 09:27:54 -0400 (Thu, 04 Nov 2010) \| 8 lines Do not output port in IPaddress for AMI sippeers. (closes issue #18248) Reported by: orn Patches: ami_sippeers.patch uploaded by pabelanger (license 224) Tested by: orn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	abc94089cd	Merged revisions 293803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293803 \| twilson \| 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) \| 25 lines Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cbd42ce6eb	Merged revisions 293807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293807 \| rmudgett \| 2010-11-03 13:35:19 -0500 (Wed, 03 Nov 2010) \| 34 lines Merged revisions 293806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293806 \| rmudgett \| 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) \| 27 lines Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 \| rmudgett \| 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) \| 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	9528e27b8c	Merged revisions 293724 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293724 \| jpeeler \| 2010-11-02 18:09:06 -0500 (Tue, 02 Nov 2010) \| 22 lines Merged revisions 293723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 \| jpeeler \| 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) \| 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 \| jpeeler \| 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) \| 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ed500a9e99	Merged revisions 293648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293648 \| rmudgett \| 2010-11-02 16:29:25 -0500 (Tue, 02 Nov 2010) \| 20 lines Merged revisions 293647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293647 \| rmudgett \| 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) \| 13 lines Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 \| rmudgett \| 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) \| 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5a28a27b0b	New CLI command 'gtalk show settings'. Review: https://reviewboard.asterisk.org/r/984/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	10cbc4a132	Merged revisions 293530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293530 \| rmudgett \| 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010) \| 10 lines Analog 3-way call would not connect all parties if one was using sig_pri. Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	53149a69df	Merged revisions 293496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293496 \| pabelanger \| 2010-11-01 12:09:05 -0400 (Mon, 01 Nov 2010) \| 13 lines Use ast_sockaddr_from_sin function not memcpy This resolves some IAX2 registration issue report on the asterisk-users mailing list. (closes issue #18202) Reported by: pabelanger Patches: update_registry.patch.v2 uploaded by pabelanger (license 224) Tested by: pabelanger, Nic Colledge (mailing list) Review: https://reviewboard.asterisk.org/r/993 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8e45c743d1	Merged revisions 293418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293418 \| rmudgett \| 2010-10-29 20:53:29 -0500 (Fri, 29 Oct 2010) \| 16 lines Merged revisions 293417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293417 \| rmudgett \| 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) \| 9 lines Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 \| rmudgett \| 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) \| 1 line Remove some more code that serves no purpose. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	611b8d72c9	Merged revisions 293341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293341 \| rmudgett \| 2010-10-29 19:46:41 -0500 (Fri, 29 Oct 2010) \| 16 lines Merged revisions 293340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293340 \| rmudgett \| 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) \| 9 lines Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 \| rmudgett \| 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) \| 1 line Remove some code that serves no purpose. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a491f69be6	Merged revisions 293305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293305 \| jpeeler \| 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010) \| 9 lines Modify sip_setoption to not complain about unknown options. This now behaves just like the other setoption callbacks. For the curious the offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting passed due to a fix for chan_local in 286189. (closes issue #17985) Reported by: globalnetinc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f6cdefbc07	Merged revisions 293081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293081 \| rmudgett \| 2010-10-26 11:32:59 -0500 (Tue, 26 Oct 2010) \| 1 line No need to define the struct if there are no users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b6c5dde767	Merged revisions 293046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293046 \| rmudgett \| 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010) \| 4 lines Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f96d27b917	Merged revisions 292969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292969 \| tilghman \| 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010) \| 2 lines Several more defines that need to be altered for compiling against an older version of libpri ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7bc278bd06	Merged revisions 292906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292906 \| tilghman \| 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) \| 4 lines Allow the DAHDI driver to compile, even with a sufficiently older version of libpri. Fixes our Bamboo builds. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	7189a944be	Merged revisions 292868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292868 \| dvossel \| 2010-10-25 14:07:50 -0500 (Mon, 25 Oct 2010) \| 39 lines Merged revisions 292867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292867 \| dvossel \| 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) \| 32 lines Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 \| dvossel \| 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) \| 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	8de8e4a11c	Merged revisions 292787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292787 \| lmadsen \| 2010-10-22 16:28:43 -0500 (Fri, 22 Oct 2010) \| 21 lines Merged revisions 292786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292786 \| lmadsen \| 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) \| 13 lines Update the LDIF file for LDAP. The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	64845d73c7	Merged revisions 292704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292704 \| rmudgett \| 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) \| 19 lines Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call. When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0bcdff65ec	Merged revisions 292667 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292667 \| tilghman \| 2010-10-21 17:09:25 -0500 (Thu, 21 Oct 2010) \| 2 lines Compile correctly on Linux (asterisk/localtime.h depends upon asterisk/autoconfig.h loading first). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	136b89e1bc	Merged revisions 292489 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292489 \| rmudgett \| 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010) \| 7 lines Send CONNECT_ACKNOWLEDGE for CIS calls too. The originator of the Q.SIG call completion signaling link was not changed to the active state when the CONNECT message came in. The T309 processing would immediately kill the signaling link because it was not in the active state. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	9653b5d500	Merged revisions 292309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292309 \| twilson \| 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) \| 10 lines Add sip show peer info about crypto and remove dated comment This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8be13e128f	Merged revisions 291942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291942 \| dvossel \| 2010-10-15 15:12:04 -0500 (Fri, 15 Oct 2010) \| 8 lines Fixes peer's host port information being lost on sip reload. (closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	58ea3034ce	Merged revisions 291827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291827 \| dvossel \| 2010-10-14 16:27:42 -0500 (Thu, 14 Oct 2010) \| 18 lines Safer xml parsing, treat all clients the same, and better local candidate selection. The gtalk channel driver was doing several unsafe operations in regards to how it parsed incoming XML messages. I have cleaned that code up so it should be much safer now. We now treat all clients types the same. We have no reason to distinguish between GMAIL and GOOGLE VOICE clients anymore because they all work the same way. I also modified how the local ip is found. If no bindaddress is provided in the config file, we attempt to determine the local ip we would use to connect to google.com. If that fails, then we fall back to the ast_find_ourip() function as a last resort. Using the new method makes it much less likely that we would ever advertise a local RTP candidate as a loopback address. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	b1cc567e3f	Merged revisions 291758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291758 \| pabelanger \| 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) \| 11 lines Add the ability for ast_find_ourip to return IPv4, IPv6 or both. While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	f91cda9566	Merged revisions 291656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291656 \| rmudgett \| 2010-10-13 18:45:11 -0500 (Wed, 13 Oct 2010) \| 34 lines Merged revisions 291655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291655 \| rmudgett \| 2010-10-13 18:36:50 -0500 (Wed, 13 Oct 2010) \| 27 lines Merged revisions 291643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291643 \| rmudgett \| 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) \| 20 lines Deadlock between dahdi_exception() and dahdi_indicate(). There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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6821 Commits (8ba7ff54b4f7c3359cd1baf4e2748b7ff4f4881e)