18363 Commits (496b509c42b8b1d6d969c141d746bdbfb379a123)

 

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Author	SHA1	Message	Date
Jeff Peeler	496b509c42	Update some missing allowed options for overlapdial ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	06cace8c42	Blocked revisions 207092 via svnmerge ... ........ r207092 | jpeeler | 2009-07-17 14:13:27 -0500 (Fri, 17 Jul 2009) | 11 lines Enhance configuration option for overlapdial allowing direction choice Previously overlap dialing could only be turned on or off for both incoming and outgoing calls. New parameters incoming, outgoing, and both have been added to allow further control. There is no change in default behavior with these new options and allows in band DTMF to be accepted in one direction if required. (closes issue #14471) Reported by: eboscani ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8c4d0ace54	Blocked revisions 207033 via svnmerge ... ........ r207033 | dvossel | 2009-07-17 13:00:38 -0500 (Fri, 17 Jul 2009) | 4 lines sip option flags handled incorrectly (issue #15376) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	65388d4e21	sip option flags handled incorrectly ... (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	8270339965	Fix segfault in sig_analog when using callwaiting, respect callwaiting options ... Sig_analog handles allocating the sub channel for callwaiting, so no longer try to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as allocated upon success of the alloc_sub callback, which was responsible for the segfault. Also, the callwaiting and callwaitingcallerid options were being unconditionally set to true. Now, the options are properly set from chan_dahdi.conf. (closes issue #15508) Reported by: elguero Tested by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	0ce3fa1c22	Merged revisions 206938 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	82ce0f4efc	TIMEOUT(absolute) returned negative value. ... (closes issue #15513) Reported by: ys git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8bf870e4af	Merged revisions 206872 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e0a8fc8c0e	Merged revisions 206867 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f8c37545ad	Merged revisions 206807 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f91bc197cd	Session timer were not activated if Supported header field in INVITE had both "timer" and other options. ... (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license 38) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	646cd02c09	The dialing flag was mistakingly removed from sig_pri. ... This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	e9e753d6f3	Merged revisions 206706 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	3402f34e9b	callerid(num) is wrong when username is missing ... A domain only sip uri <sip:123.123.123.123> would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6b5dbba90c	Merged revisions 206635 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	9d9a8a4fa3	fix a typo in sample config file for option change ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b13740d1b1	Document all meetme realtime fields, and in the process, make some field lengths more consistent. ... (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	b9e898017e	Restore some missing functionality to sig_analog. ... The main purpose of this commit is to restore missing functionality present in the ss_thread before all the sig related work was done. Two of the biggest missing things were distinctive ring detection and cid handling for V23. fxsoffhookstate and associated mwi variables have been moved inside sig_analog as they were not being set properly as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5e51a6bb1e	I AM A TERRIBLE PERSON ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	58b440bc29	Merged revisions 206487 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b25242a819	Reset the sentringing indication when redirects occur. ... If a redirecting control frame is processed or a call forward occurs, we need to reset the sentringing flag so that we can send another ringing indication to the phone that may contain a connected line update. AST-164 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e55d1b11b9	Merged revisions 206385 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c90a8c0921	Merged revisions 206284 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	6891ccad28	dns lookup of peername rather than peer's host in transmit_register() ... (closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	5a2ef47b2f	Make sure that since we are passing -c to asterisk that we have a console. ... Without this line, Asterisk will busy-loop trying to read and write to /dev/null (woops... my bad). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	76b48c5dae	Remove reference to non-existent help file ... (closes issue #15427) Reported by: brushtyler Patches: app_voicemail.c.diff uploaded by brushtyler (license 821) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	31d642cbe8	Blocked revisions 206126 via svnmerge ... ........ r206126 | russell | 2009-07-13 10:12:08 -0500 (Mon, 13 Jul 2009) | 7 lines Print CID match in "show dialplan". (closes issue #14702) Reported by: klaus3000 Patches: patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206127 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	56c72ed684	Bump up cleancount so that existing checkouts will update themselves properly for the 'Addons' -> 'ADDONS' change. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	c9c739b20f	Make the menuselect category for Add-Ons consistent with the other directories (uppercase). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c9aefb32a1	note the security events API in CHANGES ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4cf8a968fd	Add an API for reporting security events, and a security event logging module. ... This commit introduces the security events API. This API is to be used by Asterisk components to report events that have security implications. A simple example is when a connection is made but fails authentication. These events can be used by external tools manipulate firewall rules or something similar after detecting unusual activity based on security events. Inside of Asterisk, the events go through the ast_event API. This means that they have a binary encoding, and it is easy to write code to subscribe to these events and do something with them. One module is provided that is a subscriber to these events - res_security_log. This module turns security events into a parseable text format and sends them to the "security" logger level. Using logger.conf, these log entries may be sent to a file, or to syslog. One service, AMI, has been fully updated for reporting security events. AMI was chosen as it was a fairly straight forward service to convert. The next target will be chan_sip. That will be more complicated and will be done as its own project as the next phase of security events work. For more information on the security events framework, see the documentation generated from doc/tex/. "make asterisk.pdf" Review: https://reviewboard.asterisk.org/r/273/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	c01286976a	SIP register not using peer's outbound proxy ... If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	7574941cb4	Update comments about the level of T.38 support in Asterisk. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5aab96f0b7	Merged revisions 205877 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	fe493cf85e	Merged revisions 205804 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	44af00a28f	Eliminate extraneous LOG_DEBUG messages generated by app_fax. ... The transmit_audio() and transmit_t38() functions in app_fax have processing loops that are supposed to wait for frames to arrive on the channel and then handle them, but they also have short timeouts so that the loops can have watchdog timers and do other required processing. This commit changes the loops to not actually call ast_read() and attempt to process the returned frame unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages and slightly improving performance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	aafa57cf4b	Merged revisions 205775 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	c75a129323	Fix some remaining T.38 negotiation problems in app_fax. ... Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	f9d62e3c32	Fix mbl_fixup() in chan_mobile to update newchan->tech_pvt instead of oldchan. ... (closes issue #15299) Reported by: nikkk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	67d1957e60	Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. ... Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	728fbf077e	Convert func_odbc to use ast_dummy_alloc_channel() ... Review: https://reviewboard.asterisk.org/r/290/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f3560397be	Merged revisions 205599 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	b31302d55d	make this compile again under devmode ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	0d19ab4efe	pthread_self returns a pthread_t which is not an unsigned int on all ... pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ba2a8457b8	Merged revisions 205471 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	a638000451	Fix a CEL related regression with hints updating by subscribing to AST_DEVICE_STATE instead of AST_DEVICE_STATE_CHANGED. ... (closes issue #15440) Reported by: lmsteffan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e39a252b1e	Merged revisions 205409 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b99f857fbd	missing comma in devstatestring array ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	fd52c5834e	Merged revisions 205349 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines Prevent phantom calls to queue members. If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	b47c2a282d	Merged revisions 205288 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@205291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago