7039 Commits (935d40160bc66b8f2e23f3d3eb83f01bd9e3ed41)

Author	SHA1	Message	Date
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
Terry Wilson	ece8a5702a	Merged revisions 324237 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines Ignore media offers with a port of 0 Section 5.1 of RFC3264 states: A port number of zero in the offer indicates that the stream is offered but MUST NOT be used. (closes issue ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch uploaded by jacco (license 1277) Tested by: jacco, twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6f74606fda	Merged revisions 324174 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines Add header string to libpri debug output. Add header string to libpri debug output so the libpri output can be found/extracted easier from huge debug trace files. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	34e2305ae7	Merged revisions 324048 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	abd7ef817e	Merged revisions 323370 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	00181729b4	Merged revisions 323371 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this was causing NAT=Yes to always use rport when present which was against 1.6.2 behavior and the check itself was redundant since the only way this segment of code could be reached was if RPORT_PRESENT was already evaluated as true earlier. (closes issue ASTERISK-17789) Reported by: byronclark Patches: use_sip_nat_force_rport.patch uploaded by byronclark (license 1200) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	379370a396	Store sip peer name as var data on a outofcall msg. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	0bd877621e	Addition of "outofcall_message_context" sip.conf option. ... Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	4c459c2c85	Merged revisions 323040 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop. (closes issue ASTERISK-17798) tested by mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	53ef4bfc16	Merged revisions 322807 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	7df5d0d416	Add autoanswer to skinny. ... Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER. Initial value must be the time to autoanswer in ms, then optionally :BEEP to play a tone when answered and :MUTE to mute the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and play a beep. just 3000 would answer afer 3 secs of ringing with no beep and full two way audio. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	9598e5bc2f	Remove skinny do_monitor and use ast_sched_start instead ... The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything. Review: https://reviewboard.asterisk.org/r/1256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4cd9bc43c2	Merged revisions 322322 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines Make handle_request_publish do dialog expiration and destruction. This patch fixes handle_request_publish so that it does dialog expiration and destruction. Without this patch the incoming PUBLISH requests will get stuck in the dialog list. Restarting asterisk is the only way to remove them. Personal observation on one system the server hung up while looping through the channels rendering asterisk unusable and all sip phones unregisterd when they try reregister more requests are added. (closes issue #18898) Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: https://reviewboard.asterisk.org/r/1253 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ba625fa7d5	Correct some whitespace and a reference debug message. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	397c379a7d	Merged revisions 321812-321813 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription description parameter string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	9cd3cf2e71	Fix message destination extension. ... Don't send all messages to 's'. Get the destination from the request URI. (Found using automated test cases). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	3f4d0e8743	Support routing text messages outside of a call. ... Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	3588746c75	Merged revisions 321515 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines Chan_local locking cleanup. This patch removes all of the unnecessary deadlock avoidance loops that occur in chan_local. It also resolves an issue with a deadlock triggered by local channel optimizations. (issue #18028) Review: https://reviewboard.asterisk.org/r/1231/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	42907d40cd	Merged revisions 321511 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines Enhance NOTICE message to know who couldn't access the dialplan. (closes issue #19390) Reported by: lmadsen Patches: __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	0caae96609	Merged revisions 321273 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago