Don't duplicate variables on the sip_pvt. Just reset the variable list each
time.
(closes issue #19202)
Reported by: wdoekes
Patches:
issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Don't block doing silly deadlock avoidance. Just return and try again later.
The funciton gets called often enough that it's fine. Also, this change was
already made in trunk.
(closes issue #18791)
Reported by: irroot
Patches:
chan_sip.rtptimeout.patch uploaded by irroot (license 52)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If this change causes any problems, we will need to backport the more extensive
uri encoding and decoding handling changes that are in trunk/1.10.
(closes issue #18686)
Reported by: wolfgang
Patches:
quick-and-dirty.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, devellow, wolfgang, mav3rick
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes a couple SIP connected line update problems:
1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured. Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.
2) The connected line should not be updated on initial connect if there is
no connected line information. Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.
(closes issue #18367)
Reported by: GeorgeKonopacki
Patches:
issue18367_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1199/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
(closes issue #19083)
Reported by: rgagnon
Patches:
issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is based on an uncommitted patch by jpeeler for the issue. Instead of
relocking and then unlocking the channel though, we keep the lock on the channel
until we are finished doing what we need to the channel.
(closes issue #18441)
Reported by: Alric
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r317255 | russell | 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) | 22 lines
Merged revisions 317211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines
chan_sip: fix broken realtime peer count, fix memory leak
This patch addresses two bugs in chan_sip:
1) The count of realtime peers and users was off. The increment checked the
value of the caching option, while the decrement did not.
2) Add a missing regfree() for a regex.
(closes issue #19108)
Reported by: vrban
Patches:
missing_regfree.patch uploaded by vrban (license 756)
sip_object_counter.patch uploaded by vrban (license 756)
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r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) | 12 lines
Fixes session-timers=refuse not being enforced for *caller*
During handle_request_invite, the session timer mode was retrieved from
a cached variable. This patch forces a peer lookup of the session timer
mode in the case of an incoming invite.
(closes issue #18804)
Reported by: wdoekes
Patches:
issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717)
issue_18804_v2.diff uploaded by dvossel (license 671)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This has already been discussed and should have been resolved earlier. View
revsion 285565's log for more information about why it is important to not
put timer in the Require header.
(closes issue #18704)
Reported by: mfrager
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
Merged revisions 315891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
Fix our compliance with RFC 3261 section 18.2.2.
This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.
(closes issue #18951)
Reported by: jmls
Patches:
issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines
Merged revisions 315671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines
Make sure unregistering a peer unlinks it from the peer container
Instead of mostly copying the code from expire_register, just use the function
that "does the right thing".
(closes issue #16033)
Reported by: kkm
Patches:
016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
Tested by: kkm, tilghman, twilson
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just allocate
enough for the sip_pkt.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Deadlock avoidance between the sip pvt and the pvt->owner is
very difficult. Now that channel's are ao2 objects, this complication
is no longer necessary. It turns out the pvt's msg queue only
exists because of deadlock avoidance (when deadlock avoidance fails
msgs were added to a queue to be processed later), so this goes away as well.
The technique used in the new sip_lock_pvt_full() function should
be used as a template for replacing all locations where deadlock
avoidance occurs between a channel tech_pvt and the pvt's owner.
My hope is that this will begin a reversal of the invalid channel
driver locking architecture we have been using for so long.
This patch also resolves an issue where the pvt->owner gets
unlocked during processing the msg queue.
(closes issue #18690)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/1182/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes how chan_sip handles dynamic rtp payload types
it does not understand. At the moment if a dynamic payload's mime
type does not match one we understand, the payload does not get
removed from our payload table. As a result of this, the payload
is set to whatever dynamic codec we use internally for that payload
number on outgoing INVITES. This is incorrect.
This patch fixes this by properly checking the rtpmap set function's
return code to make sure it was found. The function can return both
-1 and -2 depending on the source of the mismatch. We were just
checking -1 explicitly.
Review: https://reviewboard.asterisk.org/r/1169/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The get_destination() function was not using the "s" extension when the
request URI did not specify an extension. This is a regression caused
when the URI parsing code was extracted into parse_uri().
Made get_destination() substitute the "s" extension when the parsed URI
results in an empty string.
(closes issue #18348)
Reported by: shmaize
Patches:
issue18348_v1.8.patch uploaded by rmudgett (license 664)
Tested by: shmaize
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings.
(closes issue #18759)
Reported by: bklang
Patches:
null-strings.patch uploaded by bklang (license 919)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines
Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.
Since it's a duplicate, nothing is going to be done, so delme doesn't need to
be set at all. Strangely, when this was added, this was being set to 1 in 1.6,
and 0 in trunk.
(issue AST-439)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Call path
sip_set_rtp_peer (locks chan then pvt)
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper (locks p->owner)
But by the time p->owner lock was attempted, seems as though chan and p->owner were different.
So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.
(closes issue #18837)
Reported by: alecdavis
Patches:
bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, Irontec, ZX81, cmaj
Review: [https://reviewboard.asterisk.org/r/1126/]
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only subscribe
for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its call-completion state
is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it
trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
Because this is an invalid state change, it just ignores the message. The
only state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state.
Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
the call by sending a CANCEL.
Asterisk should always send a response. Even if its a negative one.
The fix is to allow for the CCSS core to notify a CC agent that a failure
has occurred when CC is requested. The "ack" callback is replaced with a
"respond" callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that may need
to be communicated to the requester.
(closes issue #18336)
Reported by: GeorgeKonopacki
Tested by: mmichelson, rmudgett
JIRA SWP-2633
(closes issue #18337)
Reported by: GeorgeKonopacki
Tested by: mmichelson
JIRA SWP-2634
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines
Merged revisions 306617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines
Don't allow a REFER w/replaces to replace its own dialog
Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces
header that matches the dialog of the REFER. This would be a situation like A
calls B, A calls C, A transfers B to A, which is just silly. This patch makes
the transfer fail instead of making Asterisk freak out and forget to hang other
channels up.
Review: https://reviewboard.asterisk.org/r/1093/
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Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper)
has caused locking problems. Both of these functions lock the channel when
the channel argument is passed in!
In this case, the suspected problem (the backtrace makes it impossible to tell)
was the private being locked in sip_set_rtp_peer and then:
transmit_reinvite_with_sdp
try_suggested_sip_codec
pbx_builtin_getvar_helper
(Traced to verify that the fix was only required in 1.8 and later.)
(closes issue #18491)
Reported by: cmaj
Patches:
chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830)
Tested by: cmaj
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines
Merged revisions 305888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines
Minor AST_FRAME_TEXT related issues.
* Include the null terminator in the buffer length. When the frame is
queued it is copied. If the null terminator is not part of the frame
buffer length, the receiver could see garbage appended onto it.
* Add channel lock protection with ast_sendtext().
* Fixed AMI SendText action ast_sendtext() return value check.
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r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines
Merged revisions 305252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines
Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))
chan_iax2 and other channel drivers already had code to prevent this. The
attempt that app_dial was making to prevent it was not correct, so I fixed that.
(closes issue #18371)
Reported by: gbour
Patches:
18371.patch uploaded by gbour (license 1162)
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r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines
Merged revisions 303906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines
Guard against retransmitting BYEs indefinitely
In the case of an attended transfer (A calls B, A atxfers to C) where
A becomes unreachable before replying to Asterisk's BYE, Asterisk can
sometimes retransmit the BYE indefinitely. This is because
__sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out,
it will not ever be marked as ALREADYGONE, so when __sip_autodestruct
is called again, we end up starting the cycle over.
This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt
in the case of a BYE that has timed out. This should prevent Asterisk
from trying to transmit new BYE messages in the future.
Review: https://reviewboard.asterisk.org/r/1077/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Two fixes:
1) One must always have the private unlocked before calling
pbx_builtin_setvar_helper to not invalidate locking order since it locks the
channel.
2) Unlock the channel before calling pbx_find_extension, which starts and stops
autoservice during the lookup. The problem scenario as illustrated by the
reporter:
Thread: do_monitor
-----------------------
handle_request_do
handle_incoming
handle_request_refer
ast_parking_ext_valid
pbx_find_extension
ast_autoservice_stop
while (chan_list_state == as_chan_list_state) { usleep(1000); }
Thread: autoservice_run
-----------------------
autoservice_run
chan = ast_waitfor_n
ast_waitfor_nandfds
ast_waitfor_nandfds_classic / simple / complex (depending on your system)
ast_channel_lock(c[x]);
handle_request_do and schedule_process_request_queue locks the owner
if it exists. The autoservice thread is waiting for the channel lock, which
wasn't ever released since the do_monitor thread was waiting for autoservice
operations to complete. Solved by unlocking the channel but keeping a reference
to guarantee safety.
(closes issue #18403)
Reported by: jthurman
Patches:
20110103-blind_deadlock.diff uploaded by jthurman (license 614)
issue18403.patch uploaded by jpeeler (license 325)
Tested by: jthurman
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@301790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r300298 | twilson | 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) | 22 lines
Merged revisions 300216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines
Don't authenticate SUBSCRIBE re-transmissions
This only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer...
(closes issue #18075)
Reported by: mdu113
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: twilson, mdu113
Review: https://reviewboard.asterisk.org/r/1005/
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r299242 | mnicholson | 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) | 23 lines
Merged revisions 299194,299198,299220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines
Respond as soon as possible with a 202 Accepted to refer requests.
This change also plugs a few memory leaks that can occur when parking sip calls.
ABE-2656
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r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines
Remove changes to via processing that were not supposed to go into the last commit.
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r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines
Use ast_free() instead of free()
ABE-2656
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* Make sure to allocate a cc_params structure
when creating autopeers.
* Use sip_uri_cmp when retrieving SIP CC agents
and monitors in case parameters appear in the
URI.
(closes issue #18504)
Reported by: kkm
(closes issue #18338)
Reported by: GeorgeKonopacki
Patches:
18338.diff uploaded by mmichelson (license 60)
Tested by: GeorgeKonopacki
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@299248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r297960 | twilson | 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) | 21 lines
Merged revisions 297959 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines
Ignore spurious REGISTER requests
If a REGISTER request with a Call-ID matching an existing transaction is received
it was possible that the REGISTER request would overwrite the initreq of the
private structure. This info is used to generate messages for other responses in
the transaction. This patch ignores REGISTER requests that match non-REGISTER
transactions.
(closes issue #18051)
Reported by: eeman
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/1050/
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r297605 | jpeeler | 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) | 18 lines
Merged revisions 297603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines
Improve handling of REGISTER requests with multiple contact headers.
The changes here attempt to more strictly follow RFC 3261 section 10.3.
Basically the following will now cause a 400 Bad Response to be returned, if:
- multiple Contact headers are present with one set to expire all bindings ("*")
- wildcard parameter is specified for Contact without Expires header or Expires
header is not set to zero.
ABE-2442
ABE-2443
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines
Merged revisions 297072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines
Fix not stopping MOH when transfered local channel queue member is answered.
The problem here is only present when local channels are used with the MOH
passthru option as well as no optimization (/nm). I will describe the slightly
bizarre scenario that was used to test, where phones B and C are queue members:
Phone A dials into a queue with two members using local channels and the above
options. Phone B answers. Phone A blind transfers phone B into the same queue.
Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH.
In this scenario, the unhold frame that should have gotten to phone B never
arrived due to the masquerade from the blind transfer. This is usually fine
since app_queue manages the starting and stopping of MOH. However, with the
passthrough option enabled when app_queue attempts to stop MOH it tries to do
so on the local channel rather than the real channel. The easiest solution
was to just make sure to send an unhold frame during the transfer since it
wouldn't make sense to have MOH playing after a transfer anyway. This only
modifies SIP transfers, but the other transfers did not seem to be a problem.
If DTMF based transfers were a problem it might be okay to add ast_moh_stop
to finishup, but I didn't want to have to add that unless required.
ABE-2624
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
