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2010-04-29updates blocker fixes for RCdvossel1-5/+14
(closes issue 0017052) Reported by: dvossel Tested by: dvossel (closes issue 0016196) Reported by: atis (closes issue 0017052) Reported by: dvossel Tested by: dvossel git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.27-rc3@260066 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-13Merge changes from 1.6.0 branch and update ChangeLog.v1.6.0.27-rc2lmadsen1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.27-rc2@257215 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-02Merged revisions 256015 via svnmerge from russell1-24/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r256015 | russell | 2010-04-02 18:46:45 -0500 (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel() (closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@256016 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-30Merged revisions 255410 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r255410 | russell | 2010-03-30 15:56:26 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does not start. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@255411 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Merged revisions 254718 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) | 2 lines chan_usbradio depends on alsa. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@254719 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Undo unnecessary commit. Sean Bright beat me to the punch on this one.mmichelson1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@254550 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Fix potential use of uninitialized value.mmichelson1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@254549 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Initialize stream to avoid a compilation error.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@254546 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Fix crashes resulting from reading non-existent RTP streams.mmichelson1-6/+13
Specifically, when using the CHANNEL dialplan function, it was possible to crash Asterisk by trying to get the rtpdest of a stream type that is not in use by the channel. This commit fixes that issue. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@254540 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Merged revisions 253536-253538,253540 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix build issues I had with this module on FreeBSD. ........ r253537 | russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve a compiler warning on FreeBSD. ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD. ........ r253540 | russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve more compiler warnings on FreeBSD. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@253626 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-17Revert API change in release branchestwilson4-8/+8
This re-renames ast_rtp_update_source to ast_rtp_new_source git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@253158 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Merged revisions 252089 via svnmerge from twilson4-29/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@252134 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Forward declaring dahdi_pri was already done.rmudgett1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@251995 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-12Merged revisions 251987 via svnmerge from rmudgett1-5/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r251987 | rmudgett | 2010-03-12 13:40:16 -0600 (Fri, 12 Mar 2010) | 9 lines Merged revisions 251986 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010) | 1 line Make chan_dahdi wakeup_sub() prototype not conditional. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@251988 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250481 via svnmerge from jpeeler1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@250482 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250395 via svnmerge from dvossel1-3/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@250398 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Merged revisions 250246 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 | dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines fixes signed to unsigned int comparision issue for FaxMaxDatagram value. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@250265 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Merged revisions 249893 via svnmerge from dvossel14-11/+25
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@249907 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Merged revisions 249538 via svnmerge from jpeeler1-35/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines Modify queued frames from local channels to not set the other side to up In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@249539 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27overlap receiving: automatically send CALL PROCEEDING when dialplan startsalecdavis1-1/+14
Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis (closes issue #16789) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@249364 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-27Merged revisions 249235 via svnmerge from kpfleming1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line add a reference to the now-published IAX2 RFC ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@249236 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-26Merged revisions 249101 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines Merged revisions 249100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. (closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@249102 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-23Merged revisions 248397 via svnmerge from dvossel1-18/+52
https://origsvn.digium.com/svn/asterisk/trunk ................ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines Merged revisions 248396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@248400 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 228798 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk (closes issue #16470) Reported by: kjotte ........ r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@248008 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 247914 via svnmerge from rmudgett1-27/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@247922 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18fixes dialog ref count crash isolated to the 1.6.0 branchdvossel1-2/+1
(closes issue #16375) Reported by: kobaz (closes issue #16796) Reported by: kobaz git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@247839 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18Merged revisions 247787 via svnmerge from tilghman1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@247789 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10Merged revisions 246070 via svnmerge from jpeeler1-0/+36
https://origsvn.digium.com/svn/asterisk/trunk ........ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) | 22 lines Change channel state on local channels for busy,answer,ring. Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => *9700,n,wait(3) ;3 works, 1 did not exten => *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did not exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@246071 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-09Merged revisions 245793 via svnmerge from dvossel1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600 (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@245796 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-08Merged revisions 245578 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and channels/ Makefiles. They were previously passed correctly, but they simply weren't used. This caused issues with various platforms whose builds needed to pass special linker flags via the configure script. (closes issue #16596) Reported by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@245579 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-03Merged revisions 244505 via svnmerge from tilghman1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines The chanvar= setting should inherit the entire list of variables, not just the first one. (closes issue #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded by raarts (license 937) Tested by: raarts ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@244506 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-02Merged revisions 244443 via svnmerge from dvossel1-36/+64
https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported by: krn (closes issue #16724) Reported by: barthpbx (closes issue #16517) Reported by: bklang (closes issue #16485) Reported by: elsto ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@244447 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-01Merged revisions 244071 via svnmerge from tilghman1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r244071 | tilghman | 2010-02-01 11:53:39 -0600 (Mon, 01 Feb 2010) | 22 lines Merged revisions 244070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010) | 16 lines Revert previous chan_local fix (r236981) and fix instead by destroying expired frames in the queue. (closes issue #16525) Reported by: kobaz Patches: 20100126__issue16525.diff.txt uploaded by tilghman (license 14) 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: kobaz, atis (closes issue #16581) Reported by: ZX81 (closes issue #16681) Reported by: alexr1 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@244072 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-28Merged revisions 243780 via svnmerge from russell1-4/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r243780 | russell | 2010-01-28 09:07:23 -0600 (Thu, 28 Jan 2010) | 9 lines Merged revisions 243779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010) | 2 lines Fix a bogus third argument to ast_copy_string(). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@243852 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Merged revisions 243482 via svnmerge from russell1-32/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r243482 | russell | 2010-01-27 11:32:07 -0600 (Wed, 27 Jan 2010) | 13 lines Fix the ability to specify an OSP token for an outbound IAX2 call. When this patch was originally submitted, the code allowed for the token to be set via a channel variable. I decided that a cleaner approach would be to integrate it into the CHANNEL() function. Unfortunately, that is not a suitable approach. It's not possible to get the value set on the channel soon enough using that method. So, go back to the simple channel variable method. (closes issue #16711) Reported by: homesick Patches: iax-svn.diff uploaded by homesick (license 91) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@243483 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-23Backporting register line parsing from trunk to fix a bad parsing error in ↵tilghman1-66/+224
1.6.0. (closes issue #16491) Reported by: jamicque Patches: 20100114__issue16491.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines Better parsing for the "register" line Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman ........ r213635 | dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines fixes sip register parsing when user@domain is used (issue #15008) (issue #15672) ........ r215222 | tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines Fix register such that lines with a transport string, but without an authuser, parse correctly. (AST-228) ........ r215801 | tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines Default the callback extension to "s". This is a regression. (closes issue #15764) Reported by: elguero Change-type: bugfix ........ r235132 | dvossel | 2009-12-15 12:43:06 -0600 (Tue, 15 Dec 2009) | 14 lines reverse minor sip registration regression A registration regression caused by a code tweak in (issue #14331) and a bug fix in (issue #15539) caused some sip registration config entries to be constructed incorrectly. Origially issue #14331 contained the code tweak as well as a bug fix, but since the issue was reported as a tweak the bug fix portion was moved into issue #15539. Both the tweak and the bug fix contained minor incorrect logic that resulted in some SIP registrations to fail. (issue #14331) (issue #15539) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@242514 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-22Merged revisions 242227 via svnmerge from oej1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r242227 | oej | 2010-01-22 10:28:34 +0100 (Fre, 22 Jan 2010) | 11 lines Merged revisions 242226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3 lines Initialize notify_types to NULL ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@242230 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-19Merged revisions 241314 via svnmerge from jpeeler1-1/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r241314 | jpeeler | 2010-01-19 12:46:11 -0600 (Tue, 19 Jan 2010) | 20 lines Merged revisions 241227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 Jan 2010) | 13 lines Fix deadlock in agent_read by removing call to agent_logoff. One must always lock the agents list lock before the agent private. agent_read locks the private immediately, so locking the agents list lock is not an option (which is what agent_logoff requires). Because agent_read already has access to the agent private all that is necessary is to do the required hanging up that agent_logoff performed. (closes issue #16321) Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@241316 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-12Merged revisions 239427 via svnmerge from dvossel1-20/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r239427 | dvossel | 2010-01-12 10:14:41 -0600 (Tue, 12 Jan 2010) | 14 lines fixes text support in sdp answer The code that handled setting 'm=text' in the sdp was not executing in the correct order. The check to see if text was needed came after the check to add 'm=text' to the sdp, this resulted in 'm=text' always being set to 0 because it looked like text was never required. (closes issue #16457) Reported by: peterj Patches: textportinsdp.diff uploaded by peterj (license 951) issue16457.diff uploaded by dvossel (license 671) Tested by: peterj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@239447 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 209400 via svnmerge from tilghman5-28/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238494 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 238412 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in chan_iax A signed short was used to represent a callnumber. This is makes it possible to attempt to access the iaxs array with a negative index. (closes issue #16565) Reported by: jensvb ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238441 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 238405 via svnmerge from dvossel1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines Change in sip show channels display format allowing more digits for CID (closes issue #16459) Reported by: Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238408 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Something clearly went wrong with a merge somewhere, because these are all ↵tilghman1-65/+0
duplicates (and therefore dead code). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@237966 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-04Merged revisions 237319 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines It's also possible for the Local channel to directly execute an Application. Reviewboard: https://reviewboard.asterisk.org/r/452/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@237320 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-02Merged revisions 237136 via svnmerge from oej1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 lines Merged revisions 237135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines Release memory of the contact acl before unloading module ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@237137 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-30Merged revisions 236982 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines Don't queue frames to channels that have no means to process them. (closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@236983 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-29Merged revisions 236802 via svnmerge from tilghman1-4/+20
https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (closes issue #16452) Reported by: corruptor Patches: 20091221__issue16452.diff.txt uploaded by tilghman (license 14) Tested by: corruptor ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@236805 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-22Merged revisions 236063 via svnmerge from dvossel1-3/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) | 18 lines Merged revisions 236062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines fixes issue with p->method incorrectly set to ACK It is possible for a second ACK to come in for a retransmitted message. If an ack does not match an unacked message in our queue, restore the previous p->method as this ACK is completely ignored. (closes issue #16295) Reported by: omolenkamp Patches: issue16295_v2.diff uploaded by dvossel (license 671) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@236066 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-15reverses minor sip registration regressiondvossel1-7/+14
reverses the changes caused by issue #15539. The issue reported was expected behavior. (issue #15539) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@235136 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-14Merged revisions 234526 via svnmerge from oej1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r234526 | oej | 2009-12-14 11:46:20 +0100 (Mån, 14 Dec 2009) | 16 lines Merged revisions 234492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 lines Stop sending 183's after call hangup. There where still cases where the 183 keep-alive mechanism would not stop sending 183's even though the Asterisk server had sent a final reply to the invite. EDVX-28 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@234528 f38db490-d61c-443f-a65b-d21fe96a405b
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