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2010-08-10Resolve a problem with channel name tab completion.russell4-38/+56
Hitting tab without typing any part of a channel name resulted in no results. This now results in getting a full list of active channels, just as it did in previous versions of Asterisk. Review: https://reviewboard.asterisk.org/r/818/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@281529 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-10Fixed the issue caused by EXTEN including user parameters.transnexus3-37/+49
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@281497 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Make this a little more deterministic... we want the latest value, not just ↵tilghman1-1/+1
a 1 somewhere. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Apparently, the values in makeopts are sometimes 1:1 and sometimes 1. ↵tilghman1-1/+1
Compensate for this. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280624 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280449 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500 (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280450 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Don't blow up if get_codec() was not provided in the RTP glue.russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280391 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280306 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Revert r280019 for now - This was poorly executed.seanbright62-67/+25527
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280020 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Add ability to use system libedit and update bundled libedit.seanbright62-25527/+67
The version of libedit that is bundled with asterisk is old and has some bugs. This patch updates the bundled version of libedit within asterisk, and also updates asterisk to use the system libedit instead if one is available (and pkg-config is available). This review integrates several patches from other users specifically kkm and tzafrir. (closes issue #15929) Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888) (issue #16858) Reported by: jw-asterisk (closes issue #17039) Reported by: tzafrir Patches: 0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/807/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280019 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Merged revisions 279946 via svnmerge from dvossel2-1/+21
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279949 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27fix sip transaction match with authentication, fix confusing log message ↵dvossel1-2/+4
when using getaddrinfo git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279817 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Use ast_sockaddr_setnull() when http is not enabled.pabelanger1-2/+2
Otherwise, ast_tcptls_server_start() will still start http. (closes issue #17708) Reported by: pabelanger Patches: http.patch uploaded by pabelanger (license 224) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279726 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Ignore a control subclass of -1 in ast_waitfordigit_full().russell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279636 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-25Don't assume qlog is open.tilghman1-4/+6
(closes issue #17704) Reported by: vrban Patches: issue17704.patch uploaded by pabelanger (license 224) Tested by: vrban git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279390 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-24Default sin_family to AF_INET for TCP / TLS Bindaddress. pabelanger1-6/+4
Otherwise, 'manager show settings' will generate errors if manager is not enabled. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279273 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merged revisions 278981 via svnmerge from tilghman1-6/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010) | 8 lines Avoid race with consolethread on shutdown (on parallel processors). (closes issue #17080) Reported by: sybasesql Patches: 20100721__issue17080.diff.txt uploaded by tilghman (license 14) Tested by: sybasesql ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278982 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merge the realtime failover branchtilghman1-42/+101
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278957 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Allow IPv6 addresses for UDPTL streams.mmichelson1-102/+70
Review: https://reviewboard.asterisk.org/r/795 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278908 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22Add method for finding XML doc files for systems that don't support GLOB_BRACE.jpeeler1-5/+62
In particular, Solaris and perhaps others do not support the above mentioned GNU extension. In this case the paths are simply expanded without the braces and the calls to glob are made separately. Note: I could not explain memory allocation failures that were being reported from within libxml itself when making calls to glob without using GLOB_NOCHECK. This is the only reason why that flag is being used. (closes issue #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded by snuffy (license 35), modified by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278708 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22Merged revisions 278618 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul 2010) | 13 lines Allow PLC to function properly when channels use SLIN for audio. If a channel involved in a bridge was using SLIN audio, then translation paths were not guaranteed to be set up properly since in all likelihood the number of translation steps was only 1. This patch enforces the transcode_via_slin behavior if transcode_via_slin or generic_plc is enabled and one of the formats to make compatible is SLIN. AST-352 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278620 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Remove built-in AES code and use optional_api insteadtwilson4-2059/+0
Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Update documentation for 'comebacktoorigin' in featuers.conf.russell1-10/+12
The documentation for this option did not match the code. Fix that along with some minor cleanups to the code along the way. Document a slight change in behavior (to something that was previously undocumented) in UPGRADE.txt. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278425 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Separate queue_log arguments into separate fields, and allow the text file ↵tilghman1-66/+178
to be used, even when realtime is used. (closes issue #17082) Reported by: coolmig Patches: 20100720__issue17082.diff.txt uploaded by tilghman (license 14) Tested by: coolmig git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Merged revisions 278167 via svnmerge from tilghman2-1/+28
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines Do not queue up DTMF frames while a call is on hold. (Fixes ABE-2110) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278272 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman3-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Merged revisions 278023 via svnmerge from tilghman1-5/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines Off-by-one error (closes issue #16506) Reported by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278024 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-19Merged revisions 277906 via svnmerge from jeang1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) | 7 lines Avoid trying to pickup a parked extension before the park operation is completed. A crash could occur if the extension is picked up while the parking extension is being announced. Testing pu->notquiteyet while searching for a parked extension resolves this crash. (ABE-2418) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277945 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-19Make ACLs IPv6-capable.mmichelson3-51/+229
ACLs can now be configured to match IPv6 networks. This is only relevant for ACLs in chan_sip for now since other channel drivers do not support IPv6 addressing. However, once those channel drivers are outfitted to support IPv6 addressing, the ACLs will already be ready for IPv6 support. https://reviewboard.asterisk.org/r/791 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277814 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-17Merged revisions 277568 via svnmerge from tilghman1-1/+30
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines Since we split values at the semicolon, we should store values with a semicolon as an encoded value. (closes issue #17369) Reported by: gkservice Patches: 20100625__issue17369.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277773 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277625 via svnmerge from tringenbach1-0/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer. ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, ast_bridge_call() is called for a second bridge on the same channel, and it clears that flag, which still needs to get set for when the original ast_bridge_call() gets control back and checks it. Review: https://reviewboard.asterisk.org/r/741 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277657 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Finally, a method that really fixes the assertions in chan_iax2.c related to ↵tilghman1-6/+21
cancelling lagid. No, replacing usleep(1) with sched_yield() did not have an effect. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277484 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277327 via svnmerge from mnicholson1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035) Reported by: francesco_r Patches: pbx.c.patch uploaded by viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277331 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277261 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines If variable gotten is not set, will segfault on Solaris. (closes issue #17636) Reported by: bklang ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277263 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Print f->subclass.integer instead of f->subclass.mnicholson1-1/+1
(fix build breakage introduced in r277250) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277262 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277247 via svnmerge from mnicholson1-0/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation. AST-362 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277250 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Avoid crashing when installing a duplicate translation path with a lower cost.seanbright1-0/+1
(closes issue #17092) Reported by: moy Patches: translate.rev254273.patch uploaded by moy (license 222) Tested by: moy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277143 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Formatting changesoej2-8/+17
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277102 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Fix build on FreeBSDtilghman1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276871 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-15Fix linking asterisk on CentOS 5, which is using gcc 4.1.1. Gcc 4.1.2 has ↵tilghman1-0/+3
the real fix. Review: https://reviewboard.asterisk.org/r/790/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276731 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-15Merged revisions 276652 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276653 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Fix errors where incorrect address information was printed.mmichelson2-4/+4
ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions) uses thread-local storage for storing the string that it creates. In cases where ast_sockaddr_stringify_fmt was being called twice within the same statement, the result of one call would be overwritten by the result of the other call. This usually was happening in printf-like statements and was resulting in the same stringified addressed being printed twice instead of two separate addresses. I have fixed this by using ast_strdupa on the result of stringify functions if they are used twice within the same statement. As far as I could tell, there were no instances where a pointer to the result of such a call were saved anywhere, so this is the only situation I could see where this error could occur. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276570 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Oops, merge reverted this fix.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276493 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Remove the old stub files, preferring the optional_api method.tilghman5-151/+14
(closes issue #17475) Reported by: tilghman Review: https://reviewboard.asterisk.org/r/695/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276490 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Don't try to call an embedded module's backup_globals() function untilkpfleming1-1/+1
after confirming it exists. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276441 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Expand the caller ANI field to an ast_party_idrmudgett3-42/+19
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett12-853/+1492
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Merged revisions 276123 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276124 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Merged revisions 275994 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) | 14 lines Access peer->cdr directly instead of through a saved off reference. At this point in the code, it is possible that peer_cdr may be invalid. Specifically, in the blind transfer code, CDRs are swapped between channels. So, peer_cdr is no longer == peer->cdr. The scenario that exposed a crash in this code was a blind transfer that hit the system call limit, causing the transferee channel to get destroyed after the transfer attempt failed. Even if it succeeds and this code doesn't crash, this code was still trying to reset a CDR on a channel that was now owned by a different thread, which is a BadThing(tm). (ABE-2417) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275995 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-12Add which ITU spec specifies the numbering plan.rmudgett1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275725 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-12Merged revisions 275665 via svnmerge from jpeeler1-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010) | 11 lines Change ast_write to not stop generator when called from ast_prod. For SIP channels configured with the progressinband option on, the ringback was being immediately stopped. This problem was due to ast_prod being moved for a deadlock fix in 259858. Prodding the channel after setting up the generator triggered the check in ast_write to stop the generator. The fix here should write the frame the same as was done before the call to ast_prod was moved. (closes issue #17372) Reported by: tech_admin ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275682 f38db490-d61c-443f-a65b-d21fe96a405b
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