aboutsummaryrefslogtreecommitdiffstats
path: root/channels
AgeCommit message (Collapse)AuthorFilesLines
2011-07-22Merge branch 'master' of 192.168.0.100:/repos/git/asteriskHEADmasterPatrick McHardy22-207/+764
2011-07-21Merged revisions 329257 via svnmerge from russell2-25/+25
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines s/1.10/10.0/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@329258 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-21Merged revisions 329204 via svnmerge from rmudgett1-8/+1
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines 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: http://svn.digium.com/svn/asterisk/trunk@329205 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-20Merged revisions 328936 via svnmerge from kmoore1-10/+2
https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines 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: http://svn.digium.com/svn/asterisk/trunk@328937 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-19Merged revisions 328824 via svnmerge from kmoore1-6/+7
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines 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: http://svn.digium.com/svn/asterisk/trunk@328825 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-18Merged revisions 328611 via svnmerge from markm1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines 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: http://svn.digium.com/svn/asterisk/trunk@328612 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-15Add SLA to skinny.wedhorn1-118/+630
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. (closes issue ASTERISK-17947) Review: https://reviewboard.asterisk.org/r/1239/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328381 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-15Merged revisions 328329 via svnmerge from rmudgett2-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328344 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-14Merged revisions 328317 via svnmerge from rmudgett1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328317 | rmudgett | 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) | 13 lines 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: http://svn.digium.com/svn/asterisk/trunk@328318 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-14Merged revisions 328247 via svnmerge from lmadsen21-1/+39
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines 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: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-11Merged revisions 327682 via svnmerge from twilson1-1/+8
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: http://svn.digium.com/svn/asterisk/trunk@327683 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-08Merged revisions 327211 via svnmerge from rmudgett1-1/+0
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: http://svn.digium.com/svn/asterisk/trunk@327212 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-08Merged revisions 327044 via svnmerge from russell1-4/+4
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: http://svn.digium.com/svn/asterisk/trunk@327045 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Adds pass-through support for codec CELT.dvossel1-25/+29
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: http://svn.digium.com/svn/asterisk/trunk@326855 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Merged revisions 326683 via svnmerge from mnicholson1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@326684 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-07Merged revisions 326681 via svnmerge from mnicholson1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@326682 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-06Fixes newlines from being stripped from out of dialog sip MESSAGES.dvossel1-1/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326544 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-06Merge branch 'master' of 192.168.0.100:/repos/git/asteriskPatrick McHardy9-335/+629
2011-07-05Merged revisions 326411 via svnmerge from tilghman6-9/+9
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: http://svn.digium.com/svn/asterisk/trunk@326412 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05Merged revisions 326291 via svnmerge from rmudgett2-77/+217
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: http://svn.digium.com/svn/asterisk/trunk@326321 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-01Merged revisions 326144 via svnmerge from rmudgett1-29/+23
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: http://svn.digium.com/svn/asterisk/trunk@326145 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Fixes warning message caused by confbridge playback chan not being answered.dvossel1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325937 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-30Merged revisions 325935 via svnmerge from rmudgett1-12/+17
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: http://svn.digium.com/svn/asterisk/trunk@325936 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-29Merged revisions 325740 via svnmerge from kmoore2-36/+26
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: http://svn.digium.com/svn/asterisk/trunk@325741 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325416 via svnmerge from kpfleming1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@325417 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325339 via svnmerge from dvossel1-7/+15
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: http://svn.digium.com/svn/asterisk/trunk@325345 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Merged revisions 325212 via svnmerge from rmudgett1-1/+9
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: http://svn.digium.com/svn/asterisk/trunk@325213 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Fixes issue with video and text not being reinvited correctly with directmediadvossel1-6/+3
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: http://svn.digium.com/svn/asterisk/trunk@325151 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-28Don't forget to build the Via when sending MESSAGEtwilson1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@325046 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-27Merged revisions 324914 via svnmerge from rmudgett1-6/+7
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: http://svn.digium.com/svn/asterisk/trunk@324915 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324678 via svnmerge from kmoore1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@324708 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324685 via svnmerge from dvossel1-4/+12
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: http://svn.digium.com/svn/asterisk/trunk@324689 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-23Merged revisions 324652 via svnmerge from dvossel1-5/+15
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: http://svn.digium.com/svn/asterisk/trunk@324664 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-22Merged revisions 324491 via svnmerge from rmudgett1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@324495 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-22Merged revisions 324484 via svnmerge from twilson1-24/+37
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: http://svn.digium.com/svn/asterisk/trunk@324487 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-22Merged revisions 324481 via svnmerge fromrmudgett1-51/+136
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: http://svn.digium.com/svn/asterisk/trunk@324482 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-22Merged revisions 324479 via svnmerge from rmudgett1-27/+38
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: http://svn.digium.com/svn/asterisk/trunk@324480 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-21Fixes issue with finding correct extension when message context is used.dvossel1-8/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324302 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-20Merged revisions 324237 via svnmerge from twilson1-8/+8
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: http://svn.digium.com/svn/asterisk/trunk@324238 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-17Merged revisions 324174 via svnmerge from rmudgett1-24/+41
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: http://svn.digium.com/svn/asterisk/trunk@324175 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-17Merge 192.168.0.100:/repos/git/asteriskPatrick McHardy4-115/+170
2011-06-16Merged revisions 324048 via svnmerge from twilson2-21/+27
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: http://svn.digium.com/svn/asterisk/trunk@324050 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Merged revisions 323370 via svnmerge from twilson1-1/+15
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: http://svn.digium.com/svn/asterisk/trunk@323374 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Merged revisions 323371 via svnmerge from jrose1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@323372 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Store sip peer name as var data on a outofcall msg.dvossel1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323325 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13Addition of "outofcall_message_context" sip.conf option.dvossel2-2/+26
Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323212 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-10Merged revisions 323040 via svnmerge from mnicholson1-3/+10
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: http://svn.digium.com/svn/asterisk/trunk@323041 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Merged revisions 322807 via svnmerge from mnicholson1-6/+7
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: http://svn.digium.com/svn/asterisk/trunk@322808 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Add autoanswer to skinny.wedhorn1-5/+74
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: http://svn.digium.com/svn/asterisk/trunk@322544 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-08Merge 192.168.0.100:/repos/git/asteriskPatrick McHardy25-1656/+2731
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-19 19:26:54 +0000