asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	08aeb74d7a	Merged revisions 287929 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287929 \| dvossel \| 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) \| 4 lines Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header. ABE-2258 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	4a356afb7d	Merged revisions 287895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287895 \| russell \| 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) \| 10 lines Don't use ast_strdupa() from within the arguments to a function. (closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: russell Review: https://reviewboard.asterisk.org/r/927/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9b4cfb0d28	Merged revisions 287893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287893 \| tilghman \| 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) \| 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes issue #17981) Reported by: avalentin Patches: sip-anonymous-aastra.patch uploaded by avalentin (license 1107) (plus an additional fix by me) Tested by: avalentin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	f92fd39b5c	Merged revisions 287683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287683 \| rmudgett \| 2010-09-20 18:14:42 -0500 (Mon, 20 Sep 2010) \| 9 lines The inalarm flag was not set in sig_analog struct if the port is initially in alarm. Fixed initial inalarm value for sig_analog ports. Along with -r261007, this gets the inalarm flag in sync with chan_dahdi for sig_analog ports. (closes issue #16983) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e2d002a144	Merged revisions 287645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287645 \| dvossel \| 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) \| 9 lines Fixes issue with registrations not working properly with pedantic=yes. (closes issue #18017) Reported by: schmidts Patches: issues_18017_v1.diff uploaded by dvossel (license 671) Tested by: schmidts ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	27fbd5e156	Merged revisions 287643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287643 \| qwell \| 2010-09-20 16:29:46 -0500 (Mon, 20 Sep 2010) \| 15 lines Merged revisions 287642 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r287642 \| qwell \| 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) \| 8 lines Don't crash when parking a non-bridged call. (closes issue #17680) Reported by: jmhunter Patches: chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: jmhunter, DEA ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	7c77cebd4e	We do not handle AST_CAUSE_INTERWORKING which we set on a lot of incoming SIP messages. Adding error based on RFC 3398 recommendations. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c1af98603b	Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 \| rmudgett \| 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) \| 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 \| rmudgett \| 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) \| 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	41b95ee887	Merged revisions 286931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 \| jpeeler \| 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) \| 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b3fa5ec3be	Merged revisions 286904-286905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286904 \| rmudgett \| 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) \| 12 lines Unable to originate calls using E&M over T1. When originating a call from Unit Under Test to Reference Unit using E&M RBS signaling mode, I get the following warning message: "Ring/Off-hook in strange state 3 on channel 1". Fixed the sig_analog outgoing flag. It was never set when sig_analog was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408 ........ r286905 \| rmudgett \| 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) \| 1 line Simplify some code in sig_analog. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	f9c7f53a1f	Merged revisions 286868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286868 \| mnicholson \| 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) \| 16 lines Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling. This fixes a regression introduced in r274783. (closes issue #17960) Reported by: adriavidal Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mich, mnicholson, adriavidal (closes issue #17676) Reported by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c994bfae3d	Merged revisions 286834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286834 \| dvossel \| 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) \| 2 lines Sets subscribed type for outgoing MWI subscriptions so correct Event header is used. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2bb5307c8d	Merged revisions 286758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286758 \| mnicholson \| 2010-09-14 14:28:38 -0500 (Tue, 14 Sep 2010) \| 27 lines Merged revisions 286757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286757 \| mnicholson \| 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) \| 20 lines Merged revisions 286756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 \| mnicholson \| 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) \| 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	7b2c877fcb	Merged revisions 286457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286457 \| qwell \| 2010-09-13 14:40:05 -0500 (Mon, 13 Sep 2010) \| 12 lines Merged revisions 286456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286456 \| qwell \| 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) \| 5 lines Remove "Internal IP" from sip show settings, as it's not at all useful to display. (closes issue #17840) Reported by: oej ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	a6480ff889	Formatting changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	d04046fbe7	Merged revisions 286189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286189 \| twilson \| 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) \| 30 lines Merged revisions 286115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286115 \| twilson \| 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) \| 23 lines Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 \| twilson \| 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) \| 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	b51f922a34	Merged revisions 286120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286120 \| pabelanger \| 2010-09-10 17:11:08 -0400 (Fri, 10 Sep 2010) \| 18 lines Merged revisions 286117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286117 \| pabelanger \| 2010-09-10 16:55:06 -0400 (Fri, 10 Sep 2010) \| 11 lines Merged revisions 286114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286114 \| pabelanger \| 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) \| 4 lines Load iax.conf before registering any functions/applications/actions. Review: https://reviewboard.asterisk.org/r/914/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1efb27a045	Merged revisions 286118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286118 \| rmudgett \| 2010-09-10 15:55:37 -0500 (Fri, 10 Sep 2010) \| 25 lines Merged revisions 286116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286116 \| rmudgett \| 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) \| 18 lines Merged revisions 286113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 \| rmudgett \| 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) \| 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	83bc091ac3	Merged revisions 285568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285568 \| dvossel \| 2010-09-08 17:14:19 -0500 (Wed, 08 Sep 2010) \| 16 lines Merged revisions 285567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285567 \| dvossel \| 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) \| 9 lines Merged revisions 285566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285566 \| dvossel \| 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) \| 2 lines In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ede9032f92	Merged revisions 285564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285564 \| dvossel \| 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) \| 60 lines Merged revisions 285563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285563 \| dvossel \| 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) \| 54 lines Fixes interoperability problems with session timer behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require" header. This is not to our benefit and RFC 4028 section 7.1 even warns against it. It is possible for one endpoint to perform session-timer refreshes while the other endpoint does not support them. If in this case the end point performing the refreshing puts "timer" in the Require field during a refresh, the dialog will likely get terminated by the other end. 2. Change the behavior of 'session-timer=accept' in sip.conf (which is the default behavior of Asterisk with no session timer configuration specified) to only run session-timers as result of an incoming INVITE request if the INVITE contains an "Session-Expires" header... Asterisk is currently treating having the "timer" option in the "Supported" header as a request for session timers by the UAC. I do not agree with this. Session timers should only be negotiated in "accept" mode when the incoming INVITE supplies a "Session-Expires" header, otherwise RFC 4028 says we should treat a request containing no "Session-Expires" header as a session with no expiration. Below I have outlined some situations and what Asterisk's behavior is. The table reflects the behavior changes implemented by this patch. SITUATIONS: -Asterisk as UAS 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header 5. Outgoing INVITE: HAS "Session-Expires". Active - Asterisk will have an active refresh timer regardless if the other endpoint does. Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does. XXXXXXX - Not possible for mode. ______________________________________ \|SITUATIONS \| 'session-timer' MODES \| \|___________\|________________________\| \| \| originate \| accept \| \|-----------\|------------\|-----------\| \|1. \| Active \| Inactive \| \|2. \| Active \| Active \| \|3. \| XXXXXXXX \| Active \| \|4. \| XXXXXXXX \| Inactive \| \|5. \| Active \| XXXXXXXX \| -------------------------------------- (closes issue #17005) Reported by: alexrecarey ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	dc7e1c6183	Merged revisions 285455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285455 \| qwell \| 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) \| 8 lines Don't automatically add domains for wildcard bindaddrs. (closes issue #17832) Reported by: oej Patches: 17832-wildcard.diff uploaded by qwell (license 4) Tested by: qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	9b6fac435b	Merged revisions 285369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285369 \| qwell \| 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) \| 7 lines Add note to 'sip show settings' regarding dual-stack support, and a :: bindaddress. (closes issue #17831) Reported by: oej Patches: 17831-v6wildcardbind.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6c5e3d5966	Merged revisions 285195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285195 \| rmudgett \| 2010-09-07 12:47:34 -0500 (Tue, 07 Sep 2010) \| 20 lines Merged revisions 285193 via svnmerge from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 285192 via svnmerge from https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........ r285192 \| rmudgett \| 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010) \| 8 lines COLP/CONP and chan_misdn missing update chan_misdn does not update the caller id of the channel if a new connected number or ECT-INFORM (w/ new peer number on call transfer) is received. JIRA ABE-2502 JIRA SWP-2058 ........ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	3b5727bf38	Merged revisions 285017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285017 \| twilson \| 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010) \| 4 lines Call correct lock function as transferer is a sip_pvt not a channel Both functions are #defined to ao2_lock, but still... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1b2039e7db	Merged revisions 285006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285006 \| dvossel \| 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) \| 9 lines Disables auth_options_request option by default. The auth_options_request option was created to do authentication on OPTIONS request just like INVITES are done. Since it has been noted that some endpoints use OPTIONS requests as a way of qualifying a peer and that a 401 authentication response could result in interoperability issues, this option has been disabled by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	5e97e23de0	Merged revisions 284967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284967 \| bbryant \| 2010-09-03 14:19:53 -0400 (Fri, 03 Sep 2010) \| 15 lines Merged revisions 284958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284958 \| bbryant \| 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) \| 8 lines This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response. (closes issue #17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by alexkuklin (license 1115) Tested by: alexkuklin ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	16eac93882	Merged revisions 284952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284952 \| dvossel \| 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) \| 2 lines During OPTIONS authentication, the authpeer does not need to be returned for any reason. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d17eded2e9	Merged revisions 284950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284950 \| dvossel \| 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) \| 14 lines authenticate OPTIONS requests just like we would an INVITE OPTIONS requests should be treated the same as an INVITE This includes authentication. This patch adds the ability for incoming out of dialog OPTION requests to be authenticated before providing a response indicating whether an extension is available or not. The authentication routine works the exact same way as it does for incoming INVITEs. This means that if a peer has 'insecure=invite' in their peer definition, the same will be true for the processing of the OPTIONS request. Review: https://reviewboard.asterisk.org/r/881/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	3403dbf374	Merged revisions 284779-284780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284779 \| rmudgett \| 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) \| 8 lines Made output libpri event names if pri debugging is enabled when sig_pri processes them. * Simplified CLI "pri debug xx span xx" command code and removed redundant debugging enabled messages. * Made CLI "pri debug xx span xx" command only close the debugging log file if it was opened. ........ r284780 \| rmudgett \| 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) \| 2 lines Simplified pri_dchannel() poll timeout duration code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	804c8c38fd	Merged revisions 284705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284705 \| dvossel \| 2010-09-02 11:56:43 -0500 (Thu, 02 Sep 2010) \| 20 lines Merged revisions 284704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284704 \| dvossel \| 2010-09-02 11:48:51 -0500 (Thu, 02 Sep 2010) \| 13 lines Merged revisions 284703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284703 \| dvossel \| 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) \| 7 lines Removed relatedpeer code from sip_autodestruct Handling of the relatedpeer structure associated with a sip_pvt should be done during the final sip_destruction function, not in sip_autodestruct. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	172741bfcf	Merged revisions 284666 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284666 \| tilghman \| 2010-09-02 11:11:15 -0500 (Thu, 02 Sep 2010) \| 9 lines Merged revisions 284665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284665 \| tilghman \| 2010-09-02 11:07:19 -0500 (Thu, 02 Sep 2010) \| 2 lines Fixing build. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8190e96fad	Merged revisions 284610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 \| tilghman \| 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) \| 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5eae9f44f7	Merged revisions 284597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284597 \| tilghman \| 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) \| 29 lines Merged revisions 284593,284595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284593 \| tilghman \| 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) \| 18 lines Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 \| tilghman \| 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) \| 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ ................ r284595 \| tilghman \| 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) \| 2 lines Failed to rerun bootstrap.sh after last commit ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c28c620936	Merged revisions 284561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284561 \| dvossel \| 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) \| 9 lines During request to dialog matching, verify init_ruri is present before comparing. During request to dialog matching, we attempt a best effort routine for fork detection which requires several elements to be in place. The dialog's initial request uri is one of those elements. Since it is best effort, if the init_ruri is not present for some reason we can not proceed with that routine. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	920f5ea8b7	Merged revisions 284477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284477 \| twilson \| 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) \| 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP lines Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d99e8609de	Merged revisions 284415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284415 \| tilghman \| 2010-08-31 15:22:10 -0500 (Tue, 31 Aug 2010) \| 21 lines Merged revisions 284399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284399 \| tilghman \| 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) \| 14 lines Merged revisions 284393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284393 \| tilghman \| 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) \| 7 lines Don't send a devstate change on poke_noanswer if the state did not change. (closes issue #17741) Reported by: schmidts Patches: chan_sip.c.patch uploaded by schmidts (license 1077) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	7e718275a5	Add trustrpid and sendrpid global values to 'sip show settings' (closes issue #17860) Reported by: jtodd Patches: __20100816-chan_sip-sip-show-settings.txt uploaded by lmadsen (license 10) Tested by: lmadsen, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	22c5c7c437	Merged revisions 284032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284032 \| dvossel \| 2010-08-27 17:37:11 -0500 (Fri, 27 Aug 2010) \| 21 lines Merged revisions 284002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284002 \| dvossel \| 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) \| 14 lines Merged revisions 283960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283960 \| dvossel \| 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) \| 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. (closes issue #17758) Reported by: ibc Patches: multiple_accept_headers_1.4.diff uploaded by dvossel (license 671) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	522806df97	Merged revisions 283692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283692 \| dvossel \| 2010-08-26 10:26:37 -0500 (Thu, 26 Aug 2010) \| 32 lines Merged revisions 283691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283691 \| dvossel \| 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) \| 25 lines Merged revisions 283690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283690 \| dvossel \| 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) \| 19 lines Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response. If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc compliant and results in confusion at the other endpoint. sip_pretend_ack will ack and remove all the packets in the retransmit queue. This means that the INVITE will stop retransmitting, and that any response to that INVITE that comes after the pretend_ack occurs will be ignored. Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal hangup, we should let the protocol stack process the INVITE transaction and terminate the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag is used, once the dialog proceeds to an escapable state the transaction will either be canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do this any other way is incorrect. If the endpoint is not responding to the INVITE request, the INVITE must continue to be retransmitted until it times out which will result in the dialog being destroyed. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	75232687f4	Merged revisions 283595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283595 \| dvossel \| 2010-08-25 17:57:56 -0500 (Wed, 25 Aug 2010) \| 14 lines Merged revisions 283594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283594 \| dvossel \| 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) \| 7 lines Add to and from tags to NOTIFY dialog-info xml body so pickup can occur. When pedantic mode is used, the dialog-info xml generated during a ringing event must contain the to and from tag values. Otherwise if a pickup occurs using INVITE with replaces, Astrisk will not be able to locate the subscription. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	848135748f	Merged revisions 283559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283559 \| dvossel \| 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) \| 16 lines Merged revisions 283558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283558 \| dvossel \| 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) \| 10 lines Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2e4c877542	Merged revisions 283527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283527 \| russell \| 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) \| 2 lines Convert ast_log(LOG_DEBUG, ...) to ast_debug(...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Damien Wedhorn	179ba271d0	Ignore redial hard button when no previous number. (closes issue #17887) Reported by: salecha Patches: skinny.redial.diff uploaded by wedhorn (license 30) Tested by: wedhorn, salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bcf5988caf	Merged revisions 283493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 \| dvossel \| 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) \| 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	ea7ddb38fc	Merged revisions 283457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283457 \| lmadsen \| 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) \| 9 lines Fix issue where TOS is no longer set on RTP packets. Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk. (closes issue #17890) Reported by: elguero Patches: qos_18.diff uploaded by elguero (license 37) Review: https://reviewboard.asterisk.org/r/868 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bb9be59671	Merged revisions 283382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283382 \| dvossel \| 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) \| 25 lines Merged revisions 283381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283381 \| dvossel \| 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) \| 18 lines Merged revisions 283380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 \| dvossel \| 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) \| 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Damien Wedhorn	db994dbc6c	Hack to allow easy debugging of skinny in trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Damien Wedhorn	530be85aad	Add additional AST_CONTROL_ states to control2str. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Damien Wedhorn	101673347f	Fixes display issues on 7910 and older phones. Also correct the callinfo provided in skinny_answer. (closes issue #17876) Reported by: salecha Patches: skinny_cnd3.diff uploaded by wedhorn (license 30) Tested by: salecha, wedhorn Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	e91caf9b07	Merged revisions 283050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283050 \| rmudgett \| 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) \| 36 lines Merged revisions 283049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r283049 \| rmudgett \| 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) \| 29 lines Merged revisions 283048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 \| rmudgett \| 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) \| 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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