Age | Commit message (Collapse) | Author |
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
Stop sending IPv6 link-local scope-ids in SIP messages
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
Also fixed a reference leak in an error path in sip_msg_send().
........
r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines
Timout or error on INFO or MESSAGE transaction causes call to be lost.
When exchanging INFO messages within a call, 4xx error causes the call to
be disconnected although RFC 2976 explicitly states that such transactions
do not modify the state of the dialog.
When exchanging MESSAGE messages within a call, 4xx error causes the call
to be disconnected. To provide least surprise, we should not disconnect
the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428
Section 2)
(closes issue ASTERISK-17901)
Reported by: neutrino88
Review: https://reviewboard.asterisk.org/r/1257/
Review: https://reviewboard.asterisk.org/r/1258/
JIRA SWP-3486
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line
Comments and whitespace in chan_sip.c
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines
Ignore media offers with a port of 0
Section 5.1 of RFC3264 states:
A port number of zero in the offer indicates that the stream is offered
but MUST NOT be used.
(closes issue ASTERISK-17845)
Reported by: jacco
Patches:
issue19281_2.patch uploaded by jacco (license 1277)
Tested by: jacco, twilson
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines
Add header string to libpri debug output.
Add header string to libpri debug output so the libpri output can be
found/extracted easier from huge debug trace files.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines
Lock the channel before calling the setoption callback
The channel needs to be locked before calling these callback functions. Also,
sip_setoption needs to lock the pvt and a check p->rtp is non-null before using
it.
Review: https://reviewboard.asterisk.org/r/1220/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines
Add rtpkeepalives back to 1.8
The RTP-engine conversion left out support for handling rtpkeepalives.
This patch adds them back.
(closes issue ASTERISK-17304)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/1226/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines
Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT
It turned out that this was causing NAT=Yes to always use rport when present which was
against 1.6.2 behavior and the check itself was redundant since the only way this
segment of code could be reached was if RPORT_PRESENT was already evaluated as true
earlier.
(closes issue ASTERISK-17789)
Reported by: byronclark
Patches:
use_sip_nat_force_rport.patch uploaded by byronclark (license 1200)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Review: https://reviewboard.asterisk.org/r/1265/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines
Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop.
(closes issue ASTERISK-17798)
tested by mnicholson
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun 2011) | 5 lines
don't drop any voice frames when checking for T.38 during early media
(closes issue ASTERISK-17705)
Review: https://reviewboard.asterisk.org/r/1186/
patch by oej
reported by oej
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER.
Initial value must be the time to autoanswer in ms, then optionally :BEEP
to play a tone when answered and :MUTE to mute the mic when answering.
eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and
play a beep. just 3000 would answer afer 3 secs of ringing with no
beep and full two way audio.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything.
Review: https://reviewboard.asterisk.org/r/1256/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines
Make handle_request_publish do dialog expiration and destruction.
This patch fixes handle_request_publish so that it does dialog expiration and destruction.
Without this patch the incoming PUBLISH requests will get stuck in the dialog list.
Restarting asterisk is the only way to remove them.
Personal observation on one system the server hung up while looping through the channels
rendering asterisk unusable and all sip phones unregisterd when they try reregister
more requests are added.
(closes issue #18898)
Reported by: gareth
Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot
Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915
Review: https://reviewboard.asterisk.org/r/1253
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line
Correct IAX2 and SIP event subscription description string.
........
r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line
Constify subscription description parameter string.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Don't send all messages to 's'. Get the destination from the request URI.
(Found using automated test cases).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines
Chan_local locking cleanup.
This patch removes all of the unnecessary deadlock
avoidance loops that occur in chan_local. It also
resolves an issue with a deadlock triggered by
local channel optimizations.
(issue #18028)
Review: https://reviewboard.asterisk.org/r/1231/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines
Enhance NOTICE message to know who couldn't access the dialplan.
(closes issue #19390)
Reported by: lmadsen
Patches:
__20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10)
Tested by: russell
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines
markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines
Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic.
Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails.
(closes issue #19346)
Reported by: kobaz
Tested by: kobaz,JonathanRose
Review: [full review board URL with trailing slash]
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines
ast_sockaddr_resolve() in netsock2.c may deref a null pointer
Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables
(closes issue #19346)
Reported by: kobaz
Patches:
netsock2.patch uploaded by kobaz (license 834)
Tested by: kobaz, Marquis
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 May 2011) | 2 lines
Remove some variables that were set but unused.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines
Native SIP CCSS sends bad CC cancel SUBSCRIBE message.
The SUBSCRIBE message used to cancel a CC request has incorrect To/From
SIP headers. They are reversed and the dialog tags are the same when they
should not be. If pedantic mode was disabled, then the cancel would have
succeeded despite the incorrect message.
* The SIP_OUTGOING flag was not set correctly for the dialog and I had to
move some CC subscribe handling code as a result.
* Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a
CC request SUBSCRIBE message comes in and the CC instance is not found,
the 404 response was duplicated.
JIRA AST-568
JIRA SWP-3493
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | 10 lines
Fixes segfault occuring in chan_sip.c at __set_address_from_contact
Checks to see if domain contains anything before sending it off to ast_sockaddr_resolve
which is where the segfault was occuring due to null str.
(closes issue #18857)
Reported by: sybasesql
Review: https://reviewboard.asterisk.org/r/1225/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines
This commit modifies the way polling is done on TLS sockets.
Because of the buffering the TLS layer does, polling is unreliable. If poll is
called while there is data waiting to be read in the TLS layer but not at the
network layer, the messaging processing engine will not proceed until something
else writes data to the socket, which may not occur. This change modifies the
logic around TLS sockets to only poll after a failed read on a non-blocking
socket. This way we know that there is no data waiting to be read from the
buffering layer.
(closes issue #19182)
Reported by: st
Patches:
ssl-poll-fix3.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines
Adds legacy_useroption_parsing to address interoperability concerns.
With the new option engaged, Asterisk should interpret user fields with useroptions
contained within the userfield of the uri by stripping them out of the original message
whenever a semicolon is encountered in the userfield string.
(closes issue #18344)
Reported by: danimal
Tested by: jrose
Review: https://reviewboard.asterisk.org/r/1223/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines
Merged revisions 319653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines
Merged revisions 319652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines
Make sure everyone gets an unhold when a transfer succeeds
Some phones, like the Snom phones, send a hold to the transfer target after
before sending the REFER. We need to make sure that we unhold the parties
that are being connected after the masquerade. If Local channels with the /nm
option are used when dialing the parties, hold music would still be playing on
the transfer target, even after being connected with the transferee.
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines
Unbreak the storing of registrations for restart
The fix for issue 18882 broke retrieving non-realtime peers from the ast_db
on restart/reload. This patch tries to unbreak things while leaving the intent
of the original fix intact.
(closes issue #19318)
Reported by: remiq
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: lmadsen, remiq
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines
Merged revision 319468 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines
The mISDN HDLC mode is prevented on dialed channels.
The use of mISDN HDLC mode is prevented if the mISDN dial technology
option 'h1' is used when config option astdtmf=yes.
There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC
mode. Instead of setting the channel to HDLC mode it is set to
transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the
logging message is correct, but the if condition is not.
Make check the nodsp and hdlc flags.
JIRA ABE-2787
JIRA SWP-3437
..........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The vars were either explicitly or implicitly not used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.
Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.
(closes issue #19221)
Reported by: kenner
JIRA SWP-3396
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and
changes to the transmit_* functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r319204 | twilson | 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) | 11 lines
Merged revisions 319202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) | 4 lines
Unlink a peer from peers_by_ip when expiring a registration
Review: https://reviewboard.asterisk.org/r/1218/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r319145 | dvossel | 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) | 9 lines
Merged revisions 319144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) | 2 lines
Fixes issue with peer ref-counting during handle_request_subscribe.
(closes issue #19293)
Reported by: irroot
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) | 8 lines
Make sure tcptls_session exists before dereferencing it.
(closes issue #19192)
Reported by: stknob
Patches:
10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723)
Tested by: vois, Chainsaw
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
state of the channel reverts to unknown this should be rejected.
this is important for negotiating T.38 gateway see #13405
This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected.
Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states.
(closes issue #18889)
Reported by: irroot
Tested by: irroot, darkbasic, mnicholson
Review: https://reviewboard.asterisk.org/r/1115
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) | 11 lines
This patch allows TCP peers into the ast_db where they were previously
restricted.
(closes issue #18882)
Reported by: cmaj
Patches:
patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
uploaded by cmaj (license 830)
Tested by: cmaj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) | 4 lines
Handle ipv6 addresses in the sent-by Via: field.
This change fixes a regression in via header parsing and ipv6 handling.
(closes issue #18951)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines
PRI early media won't ring.
And another way to pass early media. Don't indicate that there is inband
information present, just assume that the B channel is connected.
* Restore clearing the dialing flag Rx squelch unconditionally when a
PROCEEDING message comes in.
(closes issue #19268)
Reported by: tbsky
Patches:
issue19268_v1.8.patch uploaded by rmudgett (license 664)
Tested by: tbsky
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines
Fix directed group pickup feature code *8 with pickupsounds enabled
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.
1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
2). dialplan applications for directed_pickups shouldn't beep.
3). feature code for directed pickup should beep on success/failure if configured.
Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
Moved app_directed:pickup_do() to features:ast_do_pickup().
Functions below, all now use the new ast_do_pickup()
app_directed_pickup.c:
pickup_by_channel()
pickup_by_exten()
pickup_by_mark()
pickup_by_part()
features.c:
ast_pickup_call()
(closes issue #18654)
Reported by: Docent
Patches:
ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett
Review: https://reviewboard.asterisk.org/r/1185/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Consolidate the functions and add some debugging info. Allows to be
able to set a substate without explicitly knowing what the state is.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Add the setsubstate_onhook to complete the initial substate handling
procedures. Added dumpsub(sub, forcehangup) which is the common way of
calling setsubstate_onhook. Dumpsub attempts to activate another sub
after setting the current one onhook.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) | 2 lines
Comment out the REF_DEBUG that slipped in during debugging
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines
Merged revisions 318548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines
Clean up several chan_sip reference leaks
Several situations in the code could lead to peers or sip_pvt references
being leaked. This would cause RTP ports to never be destroyed (leading
to exhaustion of all available RTP ports) and memory leaks.
The original patch for this issue from rgagnon was the result of an
obscene amount of testing and hard work, for which I am very grateful. I
did some cleanup and added a few additional refcount fixes that I found.
(closes issue #17255)
Reported by: kvveltho
Patches:
tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202)
Tested by: rgagnon, twilson, wdoekes, loloski
Review: https://reviewboard.asterisk.org/r/1101/
Review: https://reviewboard.asterisk.org/r/1207/
Review: https://reviewboard.asterisk.org/r/1210/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|