asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	41549fcd66	Merged revisions 331955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 \| rmudgett \| 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) \| 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	132a7e75a0	Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 \| dvossel \| 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) \| 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4e9515bab0	Merged revisions 331771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 \| rmudgett \| 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) \| 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a9a9bb4bda	Merged revisions 331714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 \| rmudgett \| 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) \| 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	678ece77d5	Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 \| kmoore \| 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) \| 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9bd1af5e42	Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 \| rmudgett \| 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) \| 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	82a2f7541e	Merged revisions 330705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330705 \| kmoore \| 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) \| 10 lines Call pickup broken for DAHDI channels when beginning with # The call pickup feature did not work on DAHDI devices for anything other than feature codes beginning with * since all feature codes in chan_dahdi were originally hard-coded to begin with *. This patch is also applied to chan_dahdi.c to fix this bug with radio modes. (closes issue AST-621) Review: https://reviewboard.asterisk.org/r/1336/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	e60c8ae0d0	Merged revisions 330581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330581 \| dvossel \| 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) \| 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2 resulting from an edge case in the way control frames are queued during calltoken negotiation is complete. (closes issue ASTERISK-17610) Reported by: mgrobecker ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	252a251ac2	Merged revisions 330578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 \| dvossel \| 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) \| 2 lines Optimization to buffer initialization fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	03fd1aca68	Merged revisions 330575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 \| dvossel \| 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) \| 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e1b5f9acb1	Merged revisions 330050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r330050 \| rmudgett \| 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) \| 22 lines Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 \| rmudgett \| 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) \| 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	8cf16c1c15	Merged revisions 329994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 \| qwell \| 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) \| 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	2fb05fb506	Merged revisions 329895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 \| seanbright \| 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) \| 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	aacc128812	Remove lastmsgssent from sip it has not been working since 1.6	14 years ago
Russell Bryant	94bbb01fdd	s/1.10/10.0/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	46dd023878	Merged revisions 329203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329203 \| rmudgett \| 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) \| 6 lines Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	af58e75ce1	Merged revisions 328935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 \| kmoore \| 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) \| 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, never inband. This fixes the regression introduced in revision 328823. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/2.0@328936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	98c4fee4cb	Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 \| kmoore \| 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) \| 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	c4b8e21079	Merged revisions 328608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328608 \| markm \| 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) \| 9 lines If the sip private structure is null, sip_setoption() will defref the null pointer and crash. Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash. (closes issue ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark Murawski ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	94dc903e12	Add SLA to skinny. Adds sublines to skinny lines. Each subline can be attached to an SLA station/trunk combo. Includes the following functionality: Callid is persistent for both in/out calls on all skinny devices. Can join, hold, resume. All sublines appear under a single line button. See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ee2096fe55	Make hint watcher callback take const strings for context and exten parameters. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cc8daa1385	Merged revisions 328302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328302 \| rmudgett \| 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) \| 6 lines Missing SIP pvt and channel unlock in sip_set_rtp_peer(). Regression introduced by -r326144. Add missing SIP pvt and channel unlock in sip_set_rtp_peer(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 \| lmadsen \| 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) \| 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3b4d9075f6	Merged revisions 327682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327682 \| twilson \| 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) \| 9 lines Update chan_gtalk to work with changed GMail-based calls The messages sent by the GMail client have changed, but include the old-style messages as well. This patch checks for this case and uses the old-style offer. (closes issue ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0e613fd544	Merged revisions 327211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327211 \| rmudgett \| 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) \| 9 lines INVITE 403 Forbidden response always retransmits the maximum times. Asterisk sends a 403 Forbidden response if authentication fails for an INVITE as required. However, it ignores the ACK and keeps retransmitting the response. * Made not delete the to-tag in the dialog so the expected ACK can be matched with the dialog and stop the retransmissions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	1353e83531	Merged revisions 327044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327044 \| russell \| 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) \| 2 lines Resolve some set-but-unused-variable warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	513c680b8c	Adds pass-through support for codec CELT. This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	ba1cc98f1a	Merged revisions 326683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326683 \| mnicholson \| 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) \| 3 lines use sips: or sip: depending on the transport in use when building reply digest URIs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	14553512ee	Merged revisions 326681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326681 \| mnicholson \| 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) \| 3 lines make the uri parameter used in reply digests more standards compliant in certain cases by prepending "sip:" or "sips:" to it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	a7c6f0445e	Fixes newlines from being stripped from out of dialog sip MESSAGES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	7d179abfd4	Merged revisions 326411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 \| tilghman \| 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) \| 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	14d510c5b7	Merged revisions 326291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 \| rmudgett \| 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) \| 23 lines Used auth= parameter freed during "sip reload" causes crash. If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	76e4e2e777	Merged revisions 326144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326144 \| rmudgett \| 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) \| 16 lines Better way to get chan and pvt lock for issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the channels in the defined order and avoid the need for a deadlock avoidance loop. * Lock the channel before getting the pointer to the private structure to be sure that the pointer will not change due to a masquerade or channel hangup. * To preserve sanity, check that chan and p->owner are the same. (Pointer rearangements should not happen without the protection of locks because bad things tend to happen otherwise.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	356e18629b	Fixes warning message caused by confbridge playback chan not being answered. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	39a7152df3	Merged revisions 325935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 \| rmudgett \| 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) \| 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	1d93d217f0	Merged revisions 325740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 \| kmoore \| 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) \| 7 lines chan_sip: cleanup from the introduction of ast_str Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	37d6d89d97	Merged revisions 325416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325416 \| kpfleming \| 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) \| 3 lines Fix random misspelling noticed on asterisk-users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	bb4e0c7f7c	Merged revisions 325339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325339 \| dvossel \| 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) \| 4 lines Fixes locking inversion caused by holding sip pvt lock during async_goto. (closes ASTERISK-17352) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	70be58c1a7	Merged revisions 325212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325212 \| rmudgett \| 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) \| 7 lines Use the device name and not the channel name to initialize the device state. Correct ASTERISK-11323 implementation as I don't see how it ever worked as claimed when it used the channel name and not the device name. (issue ASTERISK-11323) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	4812697542	Fixes issue with video and text not being reinvited correctly with directmedia If a SDP does not modify the session, we ignore it. However, we were defaulting no text and video support to true before checking to see if the sdp modified anything or not. This would result in process_sdp ignoring an sdp but removing video and text from the call during direct media reinvites. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04fc1c6cea	Don't forget to build the Via when sending MESSAGE git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	04226479b3	Merged revisions 324914 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324914 \| rmudgett \| 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) \| 21 lines When subscribing MWI to an unsolicited mailbox the first notification is incorrect. A remote peer subscribed to MWI with the unsolicited option and a local phone subscribed to the remote mailbox. The notify message-summary events are sent correctly except for the first one when subscribing, which will always be 0. This means the phone MWI indicator will be wrong until the mailbox read/unread count changes and the event is fired. Looks like this is a regression from ASTERISK-16149. * Fix the logic to check the cache and if allowed then fallback to manually counting mailbox messages. (closes issue ASTERISK-17997) Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA SWP-3551 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	3c10d69544	Merged revisions 324678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324678 \| kmoore \| 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) \| 11 lines Merged revisions 324643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r324643 \| kmoore \| 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) \| 4 lines Addresses AST-2011-008, memory corruption and remote crash in SIP driver. AST-2011-008 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	6693c49a6a	Merged revisions 324685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324685 \| dvossel \| 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) \| 8 lines Fixes sip crash when calling remove_uri_parameters with NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by: jaredmauch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d5ea9e5ae2	Merged revisions 324652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 \| dvossel \| 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) \| 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 \| dvossel \| 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) \| 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 \| dvossel \| 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) \| 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	10480072aa	Merged revisions 324491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324491 \| rmudgett \| 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) \| 1 line Use correct variable for text SRTP media. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	385b8c6f8b	Merged revisions 324484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 \| twilson \| 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) \| 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9000732418	Merged revisions 324481 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 Also fixed a reference leak in an error path in sip_msg_send(). ........ r324481 \| rmudgett \| 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) \| 19 lines Timout or error on INFO or MESSAGE transaction causes call to be lost. When exchanging INFO messages within a call, 4xx error causes the call to be disconnected although RFC 2976 explicitly states that such transactions do not modify the state of the dialog. When exchanging MESSAGE messages within a call, 4xx error causes the call to be disconnected. To provide least surprise, we should not disconnect the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428 Section 2) (closes issue ASTERISK-17901) Reported by: neutrino88 Review: https://reviewboard.asterisk.org/r/1257/ Review: https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e8c0be8fc2	Merged revisions 324479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324479 \| rmudgett \| 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) \| 1 line Comments and whitespace in chan_sip.c ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	b005d8dd53	Fixes issue with finding correct extension when message context is used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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7058 Commits (41549fcd66ef1e9faa1d38cf7207bae85c9d957a)