This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid. When an MWI event would occur, this would cause a seg fault.
(closes issue ASTERISK-18663)
Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan
Patches:
blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
Review: https://reviewboard.asterisk.org/r/1610/
........
Merged revisions 347058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347068 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code that allowed admins to create users with domain-only uri's had
stopped to work in 1.8 because of the reqresp parser rewrites. This is
fixed now: if you have a [mydomain.com] sip user, you can register with
useraddr sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in the
allow list as well.
Reviewboard r1606 shows a list of registration combinations and which
SIP response codes are returned.
Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson
(closes issue ASTERISK-18389)
(closes issue ASTERISK-18741)
........
Merged revisions 346899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 346900 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous patch (r346040) incorrectly parsed the URI in the presence
of a port, e.g., user@hostname:port would fail as the port would be
double appended to the SIP message. This patch uses the parse_uri function
to correctly parse the URI into its username and hostname parts, and places
them in the correct fields in the sip_pvt structure.
(issue ASTERISK-18903)
Review: https://reviewboard.asterisk.org/r/1597/
........
Merged revisions 346856 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
........
Merged revisions 346564 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 346565 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using the MessageSend application to send a SIP MESSAGE to a non-peer,
chan_sip attempted to validate the hostname or IP Address. In the process,
it stripped off the extension and failed to add it back to the sip_pvt
structure before transmitting. This patch adds the full URI passed in
from the message core to the sip_pvt structure.
(closes issue ASTERISK-18903)
Reported by: Shaun Clark
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1597/
........
Merged revisions 346040 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.
See ASTERISK-18702 it has a very good description of the issue.
I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.
* Added 'dtmf' enum value to sip.conf allowoverlap config option. The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.
* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.
* Fixed get_destination() inconsistency with the pickup extension
matching.
* Fixed initialization of PAGE3 of global_flags in reload_config().
(closes issue ASTERISK-18702)
Reported by: Pavel Troller
Review: https://reviewboard.asterisk.org/r/1517/
Review: https://reviewboard.asterisk.org/r/1582/
........
Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix bug ASTERISK-16558 which dealt with the order of responses to incoming
streams defined by SDP.
Fix unreported bug where offering multiple same-type streams would cause
Asterisk to reply with an incorrect SDP response missing one or more streams
without a proper declination.
Fix bugs related to a single non-audio stream being offered with responses
requesting codecs that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Review: https://reviewboard.asterisk.org/r/1516/
........
Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 344386 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Another deadlock between the conlock/hints and channels/channel locking
orders.
* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().
(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
........
Merged revisions 344268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 344271 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The domain matching code prior to 1.8 used to manually remove the port
from the host:port string when determining if an incoming request
matched the list of domains. When switching to the new parsing
functions, the documentation implied that the "domain" was being
returned by these functions, when instead it was returning the
"hostport" as defined by RFC 3261. This led to confusion and resulted
in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when
domain=x.x.x.x was set in sip.conf.
This patch renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport strings
when dealing with domain matching.
Review: https://reviewboard.asterisk.org/r/1574/
........
Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 344216 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
........
r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines
Fixes regression caused by r343635
There was a missing unlock for a function return that is only
present in Asterisk 10 and Asterisk Trunk.
(closes issue ASTERISK-18839)
Reported by: Michael L. Young
Patches:
asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
........
Merged revisions 343851 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 343852 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A dialog cannot be destroyed by the ao2_callback dialog_needdestroy
because of a deadlock between the dialogs container lock and the RWLOCK of
the events subscription list.
* Create dialogs_to_destroy container to hold dialogs that will be
destroyed.
* Ensure that the event subscription callback will never happen with an
invalid peer pointer by making the event callback removal the first thing
in the peer destructor callback.
NOTE: This particular deadlock will not happen with Asterisk 10, but some
of the changes still apply.
(closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky
Review: https://reviewboard.asterisk.org/r/1564/
........
Merged revisions 343577 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 343578 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.
Review: https://reviewboard.asterisk.org/r/1562/
........
Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were several problems with the dynamic realtime peer/user lookup
code. The lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during the
addition of the sipregs functionality, several possibilities for memory
leaks had been introduced. The insecure=port matching has always been
broken for anyone using the sipregs family. And, related, the broken
implementation forced those using sipregs to *still* have an ipaddr
column on their sippeers table.
Thanks Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which caused
the realtime_peer to have a completely unused code path.
This changeset fixes the leaks, the lookup inconsistenties and that
you won't need an ipaddr column on your sippeers table anymore (when
you're using sipregs). Beware that when you're using sipregs, peers
with insecure=port will now start matching!
(closes issue ASTERISK-17792)
(closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1395
........
Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 342929 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
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
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
* 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
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
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
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
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
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
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
https://origsvn.digium.com/svn/asterisk/branches/10
................
r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines
Merged revisions 335319 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines
Lock the peer->mvipvt to avoid crashes with SIP history enabled
After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
which cause issues with SIP history additions in combination with the max limit for
number of history entries.
Review: https://reviewboard.asterisk.org/r/1373/
(closes issue ASTERISK-18288)
Thanks to irrot for peer review. Work with this bug funded by IPvision AS
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Make check_rtp_timeout() remember the values returned by
ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling them.
(closes issue ASTERISK-18319)
Reported by: Rob Gagnon
Patches:
issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1377/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r334007 | kmoore | 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) | 14 lines
Merged revisions 334006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines
Correct an AMI protocol violation with SIPshowpeer
The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are
ended by using \r\n this confuses any interfacing script.
(closes issue ASTERISK-17486)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r333837 | twilson | 2011-08-29 16:41:13 -0500 (Mon, 29 Aug 2011) | 22 lines
Merged revisions 333836 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines
Refresh peer address if DNS unavailable at peer creation
If Asterisk starts and no DNS is available, outbound registrations will fail
indefinitely. This patch copies the address from the sip_registry struct, which
will be updated, to the peer->addr when necessary.
If dnsmgr is enabled, the registration fails without the patch because even
though the address on the registry is updated via dnsmgr, the address is just
copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update
the address that is copied to the sip_pvt or peers.
Closes issue ASTERISK-18000
Review: https://reviewboard.asterisk.org/r/1335/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r332119 | jrose | 2011-08-16 12:45:38 -0500 (Tue, 16 Aug 2011) | 23 lines
Merged revisions 332118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines
ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs
Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on whichever mailbox
triggered the mwi event. Now all of them get counted regardless. Also fixes a bug
involving parsing of the mailbox option in sip.conf so that trailing and leading
spaces before/after commas are trimmed.
(closes issue ASTERISK-18067)
Reported by: aragon
(closes issue ASTERISK-15479)
Reported by: Ben Winslow
Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines
In 10 and trunk this option is disabled by default.
Merged revisions 332021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
AST-580
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r331518 | kmoore | 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) | 17 lines
Merged revisions 331517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines
SIP Notify via AMI or CLI leaks SIP PVTs
Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref following it
corrects the issue as seen via REF_DEBUG. The unref existed in a distant
revision and it appears as though the wrong ref operation was removed.
(closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to address a compatability issue with certain features on certain devices
which rely on display name content to change behavior, initreqprep in chan_sip.c
has been changed to no longer substitute cid_number into the display name when
cid_name isn't present. Instead, it will send no display name in that case.
(closes issue ASTERISK-16198)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/1341/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines
Merged revisions 331248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines
Misc minor items found in code.
* Add some reentrancy protection in pbx.c when creating the contexts_table
hash table.
* Fix inverted test in chan_sip.c conditional code.
* Fix uninitialized variable and use of the wrong variable in chan_iax2.c.
* Fix test of return value in app_parkandannounce.c. Explicitly testing
for -1 is bad if the function does not actually return that value when it
fails.
* Fixup some comments and add some curly braces in features.c.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/2.0
................
r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
Merged revisions 328935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
Inband DTMF regression
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
Merged revisions 328823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
RTP bridge away with inband DTMF and feature detection
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines
INVITE 403 Forbidden response always retransmits the maximum times.
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds pass-through support for CELT. CELT
formats are defined in codecs.conf and can be configured
to any sample rate a CELT endpoint supports. This patch also
addresses a crash in channel.c resulting from a frame list being
freed incorrectly. This crash was discovered while testing a CELT
translator which had to split encoded audio into multiple frames.
The codec translator is not a part of this patch, but may be
contributed in the future.
Review: https://reviewboard.asterisk.org/r/1294/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
Add the attribute "type" to each "<use>" for menuselect.
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines
Used auth= parameter freed during "sip reload" causes crash.
If you use the auth= parameter and do a "sip reload" while there is an
ongoing call. The peer->auth data points to free'd memory.
The patch does several things:
1) Puts the authentication list into an ao2 object for reference counting
to fix the reported crash during a SIP reload.
2) Converts the authentication list from open coding to AST list macros.
3) Adds display of the global authentication list in "sip show settings".
(closes issue ASTERISK-17939)
Reported by: wdoekes
Patches:
jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/1303/
JIRA SWP-3526
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines
Better way to get chan and pvt lock for issue ASTERISK-17431.
Redoes -r308945 for issue ASTERISK-17431 deadlock fix for
sip_set_udptl_peer() and sip_set_rtp_peer().
* Lock the channels in the defined order and avoid the need for a deadlock
avoidance loop.
* Lock the channel before getting the pointer to the private structure to
be sure that the pointer will not change due to a masquerade or channel
hangup.
* To preserve sanity, check that chan and p->owner are the same. (Pointer
rearangements should not happen without the protection of locks because
bad things tend to happen otherwise.)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines
Misc minor changes in chan_sip.
* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().
* Removed a redundant static prototype.
* Some typos.
* Some whitespace.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Walter Doekes
Matthew Jordan
Jonathan Rose
Tilghman Lesher
Terry Wilson
Kinsey Moore
David Vossel
Matthew Nicholson
Stefan Schmidt
Paul Belanger
Leif Madsen
Olle Johansson
Gregory Nietsky
Jason Parker
Sean Bright
Mark Murawki
Kevin P. Fleming