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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/1.6.2@250484 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/1.6.2@250396 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/1.6.2@250253 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/1.6.2@249895 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/1.6.2@249580 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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@249321 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/1.6.2@249238 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/1.6.2@249104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-23Merged revisions 248397 via svnmerge from dvossel1-19/+53
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/1.6.2@248398 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 228798 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.2@248011 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 248003 via svnmerge from moy1-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1 line mfcr2 issue 0016844 - Fix portability bit fields and make mfcr2_immediate_accept work again, reported and patched by korihor ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@248005 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 247914 via svnmerge from rmudgett1-31/+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/1.6.2@248004 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-19Merged revisions 247915 via svnmerge from dvossel1-21/+19
https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 | dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines handle_request_invite revise comment, fix coding guideline issues I'm working with this code right now trying to analyze a deadlock. This change is just to clean up a few things before I make a more complex patch. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@247916 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/1.6.2@247792 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/1.6.2@246073 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/1.6.2@245794 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-09Merged revisions 245727 via svnmerge from mnicholson1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 | mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2 lines This commit removes an extra newline in T.38 generated SDP packets. This bug was caused by the fix introduced in r243860. (closes issue #16766) Reported by: raivisr Patches: t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) Tested by: raivisr ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@245728 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/1.6.2@245581 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/1.6.2@244508 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-02Merged revisions 244443 via svnmerge from dvossel1-36/+63
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/1.6.2@244445 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/1.6.2@244073 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-28Merged revisions 243943 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r243943 | tilghman | 2010-01-28 14:00:09 -0600 (Thu, 28 Jan 2010) | 2 lines Informational message, not an error. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@243945 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-28Merged revisions 243860 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r243860 | russell | 2010-01-28 12:35:15 -0600 (Thu, 28 Jan 2010) | 2 lines Add a missing line terminator for T.38 SDP. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@243861 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/1.6.2@243854 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/1.6.2@243485 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/1.6.2@242234 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/1.6.2@241318 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-13Show proper stats in "sip show channelstats"oej1-9/+10
SIP show channelstats show current RTCP statistics for calls - if we have it. Calls bridged in RTP p2p bridge doesn't have any statistics. In calls where the remote end doesn't send RTCP or we can't receive it due to NAT, there's no reliable data as well. Thanks, Klaus, for the patch. Sorry for the delay. (closes issue #15819) Reported by: klaus3000 Patches: asterisk-sip-show-channelstats-1.6.2.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, oej git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@239703 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/1.6.2@239428 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/1.6.2@238499 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/1.6.2@238416 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 238405 via svnmerge from dvossel1-4/+4
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/1.6.2@238406 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-06Merged revisions 237968 via svnmerge from tilghman1-5/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 | tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines Whoa, duplicate setting (dead code). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@237969 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/1.6.2@237322 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/1.6.2@237139 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/1.6.2@236985 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/1.6.2@236804 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/1.6.2@236064 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-17Merged revisions 235521 via svnmerge from file1-22/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 | file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines Remove some old code for going to the 'fax' extension when a T.38 switchover occurs. This would have already happened when we detected the CNG tone so this was basically a noop. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@235522 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-15reverses minor sip registration regressiondvossel1-6/+26
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/1.6.2@235134 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/1.6.2@234559 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-10Merged revisions 234129 via svnmerge from tilghman1-6/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r234129 | tilghman | 2009-12-10 10:24:26 -0600 (Thu, 10 Dec 2009) | 16 lines Merged revisions 234095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009) | 9 lines When we receive no response at all to our INVITE, allow the channel to be destroyed. (closes issue #15627) Reported by: falves11 Patches: 20091209__issue15627__1.6.0.diff.txt uploaded by tilghman (license 14) 20091209__issue15627__1.4.diff.txt uploaded by tilghman (license 14) Tested by: falves11 Review: https://reviewboard.asterisk.org/r/446/ (closes issue #15716) Reported by: dant (closes issue #16270) Reported by: corruptor (closes issue #15356) Reported by: falves11 (issue #16382) Reported by: lftsy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@234133 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-07Merged revisions 233472 via svnmerge from dvossel1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) | 15 lines Merged revisions 233471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines fixes missing Contact header angle brackets (closes issue #16298) Reported by: mgernoth Patches: reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@233473 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-07Merged revisions 233394 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 lines Do not reject SDP packets describing only non audio streams. (closes issue #16387) Reported by: zalex1953 Patches: media-level-c-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, zalex1953 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@233396 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232345 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response. (closes issue #16186) Reported by: atis Patches: sip_t38_response_415.patch uploaded by atis (license 242) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@232348 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232230 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 | file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario causing code to execute during reload that should not. (issue AST-263) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@232232 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232091 via svnmerge from jpeeler1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines Do not modify the gain settings on data calls. (The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@232094 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231692 via svnmerge from kpfleming1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@231696 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231602 via svnmerge from file1-3/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines When receiving SDP that matches the version of the last one do not treat it as a fatal error. (closes issue #16238) Reported by: seandarcy ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@231605 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-23Merged revisions 230881 via svnmerge from file1-34/+59
https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@230884 f38db490-d61c-443f-a65b-d21fe96a405b
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