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2010-12-09Merged revisions 297959 via svnmerge from twilson1-0/+7
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@297960 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-06Merged revisions 297603 via svnmerge from jpeeler1-7/+30
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@297605 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-03The CLI command should not contain <placeholder>s, these are for descriptions.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297534 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297185 via svnmerge from oej1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5 lines If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event. If we answer 481 the subscription that we don't want will be cancelled. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297186 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 297072 via svnmerge from jpeeler1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@297073 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Missed initializations caused startup errors on Mac OS X (and possibly ↵tilghman1-0/+10
others, too). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296950 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Merged revisions 296670 via svnmerge from pabelanger1-3/+2
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@296671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296165 via svnmerge from rmudgett1-113/+197
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@296166 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Merged revisions 295628 via svnmerge from twilson1-3/+19
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@295672 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294821 via svnmerge from rmudgett1-2/+9
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@294822 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294688 via svnmerge from jpeeler1-1/+15
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@294733 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Go off hold when we get an empty reinvite telling us to.mnicholson1-31/+40
(closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294242 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Merged revisions 293805 via svnmerge from rmudgett1-11/+14
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@293806 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293722 via svnmerge from jpeeler1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@293723 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293639 via svnmerge from rmudgett1-2/+5
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@293647 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293416 via svnmerge from rmudgett1-11/+0
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@293417 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293339 via svnmerge from rmudgett1-11/+0
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@293340 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Merged revisions 292866 via svnmerge from dvossel1-150/+180
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@292867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-22Update the LDIF file for LDAP.lmadsen1-0/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@292786 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291643 via svnmerge from rmudgett1-76/+222
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@291655 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291392 via svnmerge from russell1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Make exit from handle_request_do() consistent.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291111 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291109 via svnmerge from rmudgett1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291110 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289797 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289798 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289699 via svnmerge from jpeeler1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289700 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01don't iterate through all dialogs to find and delete old subscribesschmitds1-35/+5
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. (closes issue #17950) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289622 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Properly handle channel allocation failures duing invites with replaces.mnicholson1-2/+3
ABE-2588 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289553 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Merged revisions 288746 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) | 5 lines Don't fail a masquerade if it is already being hung up This avoids noise on some Local channel situations where we don't use /n. Thanks to Alec Davis for the suggestion. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288747 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288499 via svnmerge from twilson1-0/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) | 8 lines Don't let a Local channel get bridged to itself If a local channel gets bridged to itself, it becomes orphaned with no devices left to actually tell it to hang up. This patch modifies local_fixup() to detect this case and deny it. Review: https://reviewboard.asterisk.org/r/934 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288500 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288416 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288417 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288343 via svnmerge from dvossel1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288192 via svnmerge from rmudgett1-144/+165
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) | 26 lines In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel. Near the beginning of schedule_delivery(), ast_bridged_channel() is called on iaxs[fr->callno]->owner. However, the channel is not locked, which can result in ast_bridged_channel() crashing should owner->tech change to a technology that doesn't implement bridged_channel. I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since the owner lock was not held there either. Converted the existing channel deadlock avoidance to use iax2_lock_owner(). Using the new function simplified some awkward code. In the process of fixing the locking on ast_bridged_channel(), I also found a memory leak in socket_process() for v1.6.2 and v1.8. The local struct variable ies.vars is not freed on early/abnormal function exits. (closes issue #17919) Reported by: rain Patches: issue17919_v1.4.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/926/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288193 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Setup timer before set_config().pabelanger1-4/+4
(closes issue #18019) Reported by: Netview Patches: issue_0018019.patch uploaded by pabelanger (license 224) Tested by: Netview git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288147 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 288112 via svnmerge from tilghman1-9/+22
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288113 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Don't crash when parking a non-bridged call.qwell1-10/+18
(closes issue #17680) Reported by: jmhunter Patches: chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: jmhunter, DEA git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287642 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286756 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Remove "Internal IP" from sip show settings, as it's not at all useful to ↵qwell1-1/+0
display. (closes issue #17840) Reported by: oej git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286456 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286222 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10 Sep 2010) | 1 line Return -1 if chan_local doesn't support an option ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286223 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286114 via svnmerge from pabelanger1-19/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines Load iax.conf before registering any functions/applications/actions. Review: https://reviewboard.asterisk.org/r/914/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286117 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286113 via svnmerge from rmudgett1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286116 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286059 via svnmerge from twilson1-0/+67
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286115 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Fixes an issue with dialplan pattern matching where the specificity for ↵bbryant8-402/+528
pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285710 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Merged revisions 280811 via svnmerge from tilghman1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines Prevent DAHDI channels from overriding the callerid, once it's been set by the user. (closes issue #16661) Reported by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jstapleton ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280812 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Change SIP NOTIFY requests to expect a response so authentication will work.jpeeler1-1/+1
This changes the request to be sent with the transmit type XMIT_RELIABLE so that sip_ack doesn't return false and cause the 401 to be ignored in cases where authentication is required. (closes issue #14255) Reported by: zktech git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29fixes wrong SRV query for TLS connectiondvossel1-5/+33
(closes issue #17612) Reported by: marcelloceschia Patches: chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079) Tested by: marcelloceschia, st, pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Implement support for ast_channel_queryoption on local channels. Currently ↵mnicholson1-0/+54
only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280306 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Add missing enum value "unknown" to the SS7 called_nai and calling_nai ↵rmudgett1-0/+4
config options. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280229 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Fix bad behavior of dynamic_exclude_static option in sip.conf.mmichelson1-7/+7
We were attempting to create a contactdeny rule based on the peer's IP address before the peer's IP address had been set. By moving the processing further down in the function, we can ensure stuff works as we expect for it to. (closes issue #17717) Reported by: mmichelson Patches: 17717.patch uploaded by mmichelson (license 60) Tested by: DennisD git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279784 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Backport sip_uri_params_cmp() fix from trunk to 1.6.2.mmichelson1-39/+25
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279112 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Two more typos to cancell.tzafrir1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@278934 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-19 07:24:42 +0000