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2011-04-04Remove the channel parameter from sig_pri_handle_subcmds().Richard Mudgett
It was only used in a debug message and may not be correct anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-04Merged revisions 312575 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines Merged revisions 312574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-01Merged revisions 312509 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) | 22 lines When a call going out an NT-PTMP port gets rejected, Asterisk crashes. If a call is sent to an ISDN phone that rejects the call with RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes. I could not get my setup to crash. However, I could see the possibility from a race condition between queuing an AST_CONTROL_BUSY to the core and then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is processed before the AST_CONTROL_HANGUP is queued, the ast_channel could be destroyed out from under chan_misdn. Avoid this particular crash scenario by not queueing the AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy JIRA SWP-2679 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-01Fixing bad line break from 312384Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-01New Feature for chan_dahdi. 4 length pattern matching.Jonathan Rose
In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-31Merged revisions 312022 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines chan_misdn segfaults when DEBUG_THREADS is enabled. The segfault happens because jb->mutexjb is uninitialized from the ast_malloc(). The internals of ast_mutex_init() were assuming a nonzero value meant mutex tracking initialization had already happened. Recent changes to mutex tracking code to reduce excessive memory consumption exposed this uninitialized value. Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc(). Also eliminated redundant zero initialization code in the routine. (closes issue #18975) Reported by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-30Merged revisions 311874 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 Mar 2011) | 1 line Update some setup_dahdi_int() comments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-23Merged revisions 311612 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) | 9 lines Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null value. (closes issue #18821) Reported by: cmaj Patches: patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx uploaded by cmaj (license 830) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-23Merged revisions 311558 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) | 5 lines Don't use static declared buf in parse_name_andor_addr This function isn't used anywhere yet, but we definitely don't want to keep the same value for buf between calls to the function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-18Merged revisions 311352 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL. This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings. (closes issue #18759) Reported by: bklang Patches: null-strings.patch uploaded by bklang (license 919) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-18Merged revisions 311297 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines Race condition when ISDN CallRerouting/CallDeflection invoked. The queued AST_CONTROL_BUSY could sometimes be processed before the call_forward dial string is recognized. * Moved setting the call_forwarding dial string after sending a response to the initiator and just queue an empty frame to wake up the media thread instead of an AST_CONTROL_BUSY. * Added check for empty rerouting/deflection number and respond with an error. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-10Merged revisions 310231 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar 2011) | 9 lines Be more tolerant of what URI we accept for call completion PUBLISH requests. (closes issue #18946) Reported by: GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-08Merged revisions 310088 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | 9 lines Returns with an error notice if CHANNEL function of SIP channel is read without arguments. (Closes issue #18653) Reported by: wuwu Patches: diff.patch uploaded by jrose (license 1225) Tested by: jrose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-08Merged revisions 309994 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line Make pri parameter description consistent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-07Merged revisions 309808 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-05Merged revisions 309720 via svnmerge from Moises Silva
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar 2011) | 6 lines Fix caller id passed to openr2_chan_make_call (closes issue #18894) Reported by: malufrj Tested by: moy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-04Fix a buglet that prevented chan_nbs from loading (and subsequently stopped ↵Russell Bryant
Asterisk). In passing, convert the return codes to be the proper AST_MODULE_LOAD_* constants. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-04Merged revisions 309445 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-02Merged revisions 309256 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines Merged revisions 309255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP. Since it's a duplicate, nothing is going to be done, so delme doesn't need to be set at all. Strangely, when this was added, this was being set to 1 in 1.6, and 0 in trunk. (issue AST-439) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-01Merged revisions 309126 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal. Looks like an unintended change when sig_analog.c was extracted from chan_dahdi.c. Removed useless conditional around needed code and fixed resulting compiler warning. (closes issue #18667) Reported by: enegaard Patches: issue18667.patch uploaded by enegaard (license 1197) Tested by: enegaard JIRA SWP-2965 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-01Merged revisions 309084 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines Merged revisions 309083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines Fixes thread blocking issue in the sip TCP/TLS implementation. (closes issue #18497) Reported by: vois Patches: issues_18497.diff uploaded by dvossel (license 671) Tested by: vois, rossbeer, kowalma, Freddi_Fonet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-25Merged revisions 308945 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines Fix Deadlock with attended transfer of SIP call Call path sip_set_rtp_peer (locks chan then pvt) transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (locks p->owner) But by the time p->owner lock was attempted, seems as though chan and p->owner were different. So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods. (closes issue #18837) Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-24Merged revisions 308679 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-23Merged revisions 308622 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails. (closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵David Vossel
audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-17Add more verbage to CLI command 'pri show channels' usage.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-15Add CLI "pri show channels" command.Richard Mudgett
List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-15Fixes compile error in chan_phone for big endian David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-15Merged revisions 307879 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-10Fixes bug in chan_sip where nativeformats are not set correctly.David Vossel
The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Merged revisions 306979 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306979 | twilson | 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Use correct conditional for MCID send.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Pass a MCID request to the bridged channel.Richard Mudgett
Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Merged revisions 306619 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306619 | twilson | 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) | 24 lines Merged revisions 306618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-05Ignore voice frames in chan_dahdi native bridging. Hardware is handling them.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Add ISDN display ie text handling options to chan_dahdi.conf.Richard Mudgett
The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()Paul Belanger
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 306215 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) | 20 lines Fix SIP deadlock involving state changes. Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) has caused locking problems. Both of these functions lock the channel when the channel argument is passed in! In this case, the suspected problem (the backtrace makes it impossible to tell) was the private being locked in sip_set_rtp_peer and then: transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to verify that the fix was only required in 1.8 and later.) (closes issue #18491) Reported by: cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 306127 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines Merged revisions 306126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 305923 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02Replace link to old doc with new wiki page.Andrew Latham
Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-01Merged revisions 305692 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb 2011) | 7 lines Reverse sense of an error test when reading from astdb. (closes issue #18545) Reported by: jcovert Patches: chan_iax2.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-01Merged revisions 305343 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines Merged revisions 305342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines Obtain the pri lock for PRI queue counters. Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 305254 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-27Merged from revision 304341Richard Mudgett
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf pricpndialplan option. * Added from_channel value to prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-26Merged revisions 304245 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-26Merged revisions 304150 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600 (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25Merged revisions 303962 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines Merged revisions 303960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines Guard against retransmitting BYEs indefinitely In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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