asterisk

Commit Graph

Author	SHA1	Message	Date
Paul Belanger	1ed8cd087a	Outgoing calls with Google Voice Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) ........ Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341436 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5f8648892f	Don't use is_int() since it doesn't link well on all platforms Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	2816ccc516	Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS ........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341377 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b0076c5be1	Don't resolve numeric hosts or contact unresolved hosts If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341315 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	10de040b6e	More parking issues. * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	9f83c2b513	Initialize variables before calling parse_uri If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341190 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2cb5178d29	Don't try to remove peers without IPs from peers_by_ip (closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341089 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	899df042f5	Fix simple switch to not progress a call when call already progressed. If a simple switch was started on a device and then a specific call made (such as redial or speed dial), on timeout of the simple switch the call would be attempted again. This patch only allows the simple switch to make a call if the substate is still in the collecting digits mode. Also added small debug message to dialAndAactivate sub. Tested by snuff and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	4b9546abdf	Merged revisions 340971 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340971 \| kmoore \| 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) \| 15 lines Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 \| kmoore \| 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) \| 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	c48bee8e82	Merged revisions 340718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340718 \| schmidts \| 2011-10-13 06:59:50 +0000 (Thu, 13 Oct 2011) \| 9 lines Merged revisions 340717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 \| schmidts \| 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) \| 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5c77498afd	Initialize ast_sockaddr before calling ast_sockaddr_resolve Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340716 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	ee8844782c	Merged revisions 340577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340577 \| schmidts \| 2011-10-12 20:33:37 +0000 (Mit, 12 Okt 2011) \| 9 lines Merged revisions 340576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 \| schmidts \| 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) \| 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e7ebf7d5ab	Merged revisions 340578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340578 \| twilson \| 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) \| 16 lines Merged revisions 340534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 \| twilson \| 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) \| 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3bc3e9bbb7	Initialize the PRI channel alarms properly on startup. The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen ........ Merged revisions 340522 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340523 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	f2cc666a99	Fix verbose messages when IPv6 logic was added (closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340419 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9abab10b66	Add protection for SS7 channel allocation and better glare handling. * Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340366 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b63c1cc545	Fix some potential deadlocks pointed out by helgrind. * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	4ec8d57454	Merged revisions 340165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340165 \| mjordan \| 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) \| 20 lines Merged revisions 340164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340164 \| mjordan \| 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) \| 13 lines Updated chan_sip to place calls on hold if SDP address in INVITE is ANY This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	7cb2ac8664	Add skinny version 17 protocol support. Added some data to skinny packet structures to make compatible with v17. Added protocolversion to device, set on registration based on the version provided by device. v17 includes some increased ip space for ip6. This patch increases ip space in the packets but still only uses ip4. Some packet structures duplicated (ip4 and ip6 types). ip4 type used unless version is greater or equal to 17. Tested by snuff and myself on 7961 with recent 8.5 firmware. Also tested compatible with old 7960 and older 30VIPs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	c5546e2bd7	Increase SKINNY_MAX_PACKET and add some logging. Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages in v17 that are greater than the old 1000 bytes. Also add some useful logging regarding packet and session handling. A device (with protocol v17) was sending a packet with length greater than 1000 which resulted in the TCP session being destroyed and registration being retryed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	0ac40dc255	Merged revisions 340031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340031 \| wedhorn \| 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) \| 8 lines Return -1 to skinny_session if register rejected. If device registration is rejected, return -1 so that the session is destroyed immediately. Previously, a segfault would occur on a graceful shutdown if a register is rejected and the skinny_session has not yet timed out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	b90964eda5	Merged revisions 339992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339992 \| wedhorn \| 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) \| 9 lines Remove log message on traverse session list. On destroying a session, a list of sessions is traversed to find the matching session. For each session not matching, skinny erroneously logged that the session was not matched. While technically correct the message was misleading, and tended to indicate errors that were not there. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	7e5ce2ac49	Merged revisions 339942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339942 \| igorg \| 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) \| 12 lines Merged revisions 339938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339938 \| igorg \| 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) \| 6 lines Fix compilation issue, caused by missed session structure (closes issue ASTERISK-18694) Reported by: alex70 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	326c3a39d5	Merged revisions 339885 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339885 \| igorg \| 2011-10-08 22:46:27 +0700 (Сбт, 08 Окт 2011) \| 13 lines Merged revisions 339884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339884 \| igorg \| 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) \| 7 lines Fix segfault in Unistim channel (closes issue ASTERISK-18638) Reported by: jonnt ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	a01b34f488	Merged revisions 339831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339831 \| igorg \| 2011-10-08 22:01:35 +0700 (Сбт, 08 Окт 2011) \| 14 lines Merged revisions 339830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339830 \| igorg \| 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) \| 8 lines Fix char array cast as short array in send_client() function (for ARM platform) (closes issue ASTERISK-17314) Reported by: jjoshua ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	d59bb319fc	Fixed segfault on core stop gracefully. There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2f82296096	Merged revisions 339626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339626 \| rmudgett \| 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) \| 25 lines Merged revisions 339625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339625 \| rmudgett \| 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) \| 18 lines Fix debugging messages generated by 'udptl debug'. * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	34bf1527e8	Merged revisions 339148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339148 \| lmadsen \| 2011-10-03 15:13:16 -0500 (Mon, 03 Oct 2011) \| 14 lines Merged revisions 339147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339147 \| lmadsen \| 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) \| 6 lines Remove duplicated Maxforwards line in AMI output. (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2644af39b4	Merged revisions 339088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339088 \| twilson \| 2011-10-03 11:44:27 -0700 (Mon, 03 Oct 2011) \| 17 lines Merged revisions 339086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339086 \| twilson \| 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) \| 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cb0a0a9f29	Merged revisions 338801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338801 \| rmudgett \| 2011-09-30 17:06:48 -0500 (Fri, 30 Sep 2011) \| 19 lines Merged revisions 338800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338800 \| rmudgett \| 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) \| 12 lines Fix segfault in analog_ss_thread() not checking ast_read() for NULL. NOTE: The problem was reported against v1.6.2. It is unlikely to ever happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The version in sig_analog.c has largely replaced it. (closes issue ASTERISK-18648) Reported by: Stephan Bosch Patches: jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Stephan Bosch ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	260648043b	Formatting changes only --Denna och nedanstående rader kommer inte med i loggmeddelandet-- M channels/chan_sip.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	977742747d	Fix formatting of AMI header for SIP show peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny ........ Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 338664 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	c4a7d0e2c7	Merged revisions 338417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338417 \| irroot \| 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) \| 19 lines Merged revisions 338416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 \| irroot \| 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) \| 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	50350a47ea	Merged revisions 338323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338323 \| rmudgett \| 2011-09-28 17:36:57 -0500 (Wed, 28 Sep 2011) \| 12 lines Merged revisions 338322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338322 \| rmudgett \| 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) \| 5 lines Make duplicate call ptr warning message more helpful. * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	a6c29b931e	Merged revisions 338228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338228 \| qwell \| 2011-09-28 15:54:35 -0500 (Wed, 28 Sep 2011) \| 9 lines Merged revisions 338227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338227 \| qwell \| 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) \| 1 line Add support levels to non-module sections of menuselect (cflags, utils, etc). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	36a8264892	Merged revisions 338225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338225 \| rmudgett \| 2011-09-28 15:26:39 -0500 (Wed, 28 Sep 2011) \| 12 lines Merged revisions 338224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338224 \| rmudgett \| 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) \| 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. (closes issue ASTERISK-18357) Reported by: Matthew Nicholson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	6e0f7be7c9	Whitespace (red blobs) fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e39f6bba33	Merged revisions 337721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337721 \| rmudgett \| 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) \| 25 lines Merged revisions 337720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 \| rmudgett \| 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) \| 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5982bdcb7c	Merged revisions 337595,337597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 \| jrose \| 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) \| 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 \| jrose \| 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) \| 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	308ec93d64	Merged revisions 337487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337487 \| irroot \| 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) \| 16 lines Merged revisions 337486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337486 \| irroot \| 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) \| 10 lines If IP address is used in chan_h323 host parameter of peer configuration. module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7fe331fd59	Merged revisions 337008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337008 \| rmudgett \| 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) \| 22 lines Merged revisions 337007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337007 \| rmudgett \| 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) \| 15 lines Check if a channel was created before using the pointer in sig_ss7_new_ast_channel(). Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing libss7 access lock protection. * Prevent cancelling the ss7_linkset() thread at inoportune times just like the pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett (attached to related ASTERISK-17966) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b3768f04c3	Merged revisions 336978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336978 \| rmudgett \| 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) \| 28 lines Merged revisions 336977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336977 \| rmudgett \| 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) \| 21 lines Fix deadlock from not releasing SS7 linkset lock. sig_ss7_hangup() failed to release the SS7 linkset lock if the call had the alreadyhungup flag set. * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the alreadyhungup flag is set. * Made ss7_start_call() not hold any locks while creating the channel for an incoming call to prevent deadlock. * Made ss7_grab() a void function, since it could never fail, to simplify calling code. * Made obtain the channel lock to do softhangup in some places. Patches: jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett JIRA AST-668 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8493c46308	Merged revisions 336936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336936 \| irroot \| 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) \| 14 lines Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	098efb6641	Merged revisions 336792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336792 \| twilson \| 2011-09-19 17:13:34 -0500 (Mon, 19 Sep 2011) \| 9 lines Merged revisions 336791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 \| twilson \| 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) \| 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0f9330b58c	Merged revisions 336570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336570 \| rmudgett \| 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) \| 11 lines Merged revisions 336569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336569 \| rmudgett \| 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) \| 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	1ec4cb8ea0	Merged revisions 336502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336502 \| oej \| 2011-09-19 15:38:53 +0200 (Mån, 19 Sep 2011) \| 12 lines Merged revisions 336501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 \| oej \| 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) \| 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	d9306c4087	Merged revisions 336500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336500 \| irroot \| 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) \| 19 lines Merged revisions 336499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336499 \| irroot \| 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) \| 13 lines A long time ago in a galaxy far far away a IPv6 update was made, chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	5b4b76d3aa	Merged revisions 336381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336381 \| oej \| 2011-09-19 12:05:00 +0200 (Mån, 19 Sep 2011) \| 16 lines Merged revisions 336378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 \| oej \| 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) \| 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	beae2df26e	Merged revisions 336307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 \| jrose \| 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) \| 20 lines Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 \| jrose \| 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) \| 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	b5a641d1fe	Merged revisions 336167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336167 \| irroot \| 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) \| 22 lines Merged revisions 336166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336166 \| irroot \| 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) \| 16 lines The round robin routing routine in chan_misdn.c is broken. it rotates between ports but never checks the channels in the ports. i have extensivly tested it and verified it works on 1 upto 4 ports. before the patch only 1 out of each port was used now all are used as expected. (closes issue ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed by: irroot Review: https://reviewboard.asterisk.org/r/1410/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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