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2010-04-30Merged revisions 260437 via svnmerge from jpeeler1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500 (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@260440 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Merged revisions 260231 via svnmerge from rmudgett1-12/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500 (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@260233 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259957 via svnmerge from mmichelson1-3/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 | mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 lines Don't override peer context with domain context. (closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@259958 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259870 via svnmerge from dvossel1-5/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500 (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines resolves deadlocks in chan_local Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@259930 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Merged revisions 259538 via svnmerge from rmudgett1-4/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500 (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@259615 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Merged revisions 259307 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) | 21 lines Merged revisions 259270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@259309 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Merged revisions 258305 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 | dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines fixes issue with double "sip:" in header field This is a clear mistake in logic. Future discussions about how to avoid having to handle uri's like this should take place in the future, but this fix needs to go in for now. (closes issue #15847) Reported by: ebroad Patches: doublesip.patch uploaded by ebroad (license 878) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@258334 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Merged revisions 257493 via svnmerge from tilghman1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) | 20 lines Merged revisions 257467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@257508 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-13Merged revisions 257191 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 | tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 lines Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@257208 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-06Fix malformed if test. Regression of issue 15883.rmudgett1-4/+10
Converted if statement to a switch statement for clarity. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@256365 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-06Merged revisions 256265 via svnmerge from rmudgett1-7/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500 (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@256267 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/1.6.1@256017 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/1.6.1@255412 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/1.6.1@254720 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/1.6.1@254547 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Fix potential crashes from trying to reference non-existent RTP streams.mmichelson1-7/+14
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@254541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Merged revisions 253536 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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@253621 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/1.6.1@253158 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Merged revisions 252442 via svnmerge from tilghman1-6/+22
https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 | tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines THIS IS NOT PYTHON. Indentation doesn't matter, only braces do. (closes issue #17025) Reported by: smurfix Patches: sip.patch uploaded by smurfix (license 547) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@252443 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-13Merged 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/1.6.1@252135 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/1.6.1@251996 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/1.6.1@251990 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/1.6.1@250483 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/1.6.1@250397 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/1.6.1@250260 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/1.6.1@249896 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-01Merged revisions 249538 via svnmerge from jpeeler1-34/+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/1.6.1@249548 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/1.6.1@249363 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/1.6.1@249237 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/1.6.1@249103 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/1.6.1@248399 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/1.6.1@248009 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/1.6.1@247946 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/1.6.1@247790 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10Merged revisions 246070 via svnmerge from jpeeler1-0/+35
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/1.6.1@246072 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/1.6.1@245795 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/1.6.1@245580 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/1.6.1@244507 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/1.6.1@244446 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/1.6.1@244074 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/1.6.1@243944 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/1.6.1@243853 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/1.6.1@243484 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/1.6.1@242231 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/1.6.1@241317 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-13Show proper stats in "sip show channelstats"oej1-9/+10
(closes issue #15819) Reported by: klaus3000 Patches: asterisk-sip-show-channelstats-1.6.1.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, oej git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@239706 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/1.6.1@239440 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/1.6.1@238497 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/1.6.1@238430 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/1.6.1@238407 f38db490-d61c-443f-a65b-d21fe96a405b
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