asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	fcd2b0e2c7	Merged revisions 288500 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288500 \| twilson \| 2010-09-22 16:10:09 -0700 (Wed, 22 Sep 2010) \| 15 lines Merged revisions 288499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288499 \| twilson \| 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) \| 8 lines Don't let a Local channel get bridged to itself If a local channel gets bridged to itself, it becomes orphaned with no devices left to actually tell it to hang up. This patch modifies local_fixup() to detect this case and deny it. Review: https://reviewboard.asterisk.org/r/934 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	0f4fa2300a	Merged revisions 288417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288417 \| dvossel \| 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) \| 11 lines Merged revisions 288416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 \| dvossel \| 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) \| 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4cb567b461	Merged revisions 288344 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288344 \| dvossel \| 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) \| 9 lines Merged revisions 288343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 \| dvossel \| 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) \| 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e2ceeb5789	Merged revisions 288193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288193 \| rmudgett \| 2010-09-21 19:03:37 -0500 (Tue, 21 Sep 2010) \| 33 lines Merged revisions 288192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288192 \| rmudgett \| 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) \| 26 lines In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel. Near the beginning of schedule_delivery(), ast_bridged_channel() is called on iaxs[fr->callno]->owner. However, the channel is not locked, which can result in ast_bridged_channel() crashing should owner->tech change to a technology that doesn't implement bridged_channel. I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since the owner lock was not held there either. Converted the existing channel deadlock avoidance to use iax2_lock_owner(). Using the new function simplified some awkward code. In the process of fixing the locking on ast_bridged_channel(), I also found a memory leak in socket_process() for v1.6.2 and v1.8. The local struct variable ies.vars is not freed on early/abnormal function exits. (closes issue #17919) Reported by: rain Patches: issue17919_v1.4.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/926/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	913c6b39b4	Merged revisions 288113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288113 \| tilghman \| 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) \| 22 lines Merged revisions 288112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 \| tilghman \| 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) \| 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5542bb8e90	Merged revisions 288147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r288147 \| pabelanger \| 2010-09-21 18:22:43 -0400 (Tue, 21 Sep 2010) \| 9 lines Setup timer before set_config(). (closes issue #18019) Reported by: Netview Patches: issue_0018019.patch uploaded by pabelanger (license 224) Tested by: Netview ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	35d4d7fb48	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/branches/1.8@287929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	d0581b8bbd	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/branches/1.8@287895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a39b2f5ed2	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/branches/1.8@287893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8909fde226	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/branches/1.8@287683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	9cffa9cb3f	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/branches/1.8@287645 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	3ffb314312	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/branches/1.8@287643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ca5be12f8c	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/branches/1.8@287017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	c9bfde6afd	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/branches/1.8@286931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cbb2327e6d	Simplify some code in sig_analog. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a6c9f0aadc	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	ebe189365e	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/branches/1.8@286868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	50d114dcd5	Sets subscribed type for outgoing MWI subscriptions so correct Event header is used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	d028e9839e	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/branches/1.8@286758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	67c20662b7	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/branches/1.8@286457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	e72b55f3cf	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/branches/1.8@286189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	f08842e846	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/branches/1.8@286120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c54833f35d	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/branches/1.8@286118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	006435cc1f	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/branches/1.8@285568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	b452a0fc01	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/branches/1.8@285564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	7e6f798329	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/branches/1.8@285455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	de7ee06771	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/branches/1.8@285369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	292ef559d9	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/branches/1.8@285195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4b9b342078	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/branches/1.8@285017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4c42713010	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/branches/1.8@285006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	a9d85157ed	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/branches/1.8@284967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	677c54d1f2	During OPTIONS authentication, the authpeer does not need to be returned for any reason. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	125f089394	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/branches/1.8@284950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	5f2b40c699	Simplified pri_dchannel() poll timeout duration code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c271acacda	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	b5f428dee5	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/branches/1.8@284705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	c69a826812	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/branches/1.8@284666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	7e3f95e00a	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/branches/1.8@284610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6c61e312c6	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/branches/1.8@284597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	ed423183d6	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/branches/1.8@284561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	8a112de270	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/branches/1.8@284477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b8dbf411e8	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/branches/1.8@284415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	962f12b524	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/branches/1.8@284032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	9bb986156a	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/branches/1.8@283692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	e781f27150	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/branches/1.8@283595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	8ae2b6a612	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/branches/1.8@283559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	abca511f03	Convert ast_log(LOG_DEBUG, ...) to ast_debug(...) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2787a14001	Changes the default behavior for sip.conf's pedantic option from "no" to "yes". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	5c82781efe	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/branches/1.8@283457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	6f3a4b0511	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/branches/1.8@283382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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6723 Commits (a08abc1dca23b832b055f288797038a10d98e17f)