If a BLF subscription exists for long enough, using %d may print negative version numbers.
Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative.
Tested with Asterisk 1.8.8.2 with Grandstream phones.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1694/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix corner cases in get_calleridname() parsing and ensure that the
output buffer is nul terminated.
* Make get_calleridname() truncate the name it parses if the given buffer
is too small rather than abandoning the parse and not returning anything
for the name. Adjusted get_calleridname_test() unit test to handle the
truncation change.
* Fix get_in_brackets_test() unit test to check the results of
get_in_brackets() correctly.
* Fix parse_name_andor_addr() to not return the address of a local buffer.
This function is currently not used.
* Fix potential NULL pointer dereference in sip_sendtext().
* No need to memset(calleridname) in check_user_full() or tmp_name in
get_name_and_number() because get_calleridname() ensures that it is nul
terminated.
* Reply with an accurate response if get_msg_text() fails in
receive_message(). This is academic in v1.8 because get_msg_text() can
never fail.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347058 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The issue mentioned in the bug report had been fixed recently by
twilson. The reporter included this documentation fix.
(closes issue ASTERISK-18572)
Reported by: Richard Miller
Patch by: Richard Miller (modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345160 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344385 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original REGISTERTRYING flag, in addition to being impossible to
check, also encroached on the space for the flag above it. This
patch moves the flags that were below REGISTERTRYING back to where
they were as though we had just removed the REGISTERTRYING option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343276 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343220 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341254 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/branches/1.8@326291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/branches/1.8@321155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/branches/1.8@319938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This function isn't used anywhere yet, but we definitely don't want
to keep the same value for buf between calls to the function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies the way chan_sip.c does transaction to dialog
matching. Asterisk now stores information in the top most Via header
of the initial incoming request and compares that against other Requests
that have the same call-id. This results in Asterisk being able to
detect a forked call in which it has received multiple legs of the
fork. I completely stripped out the previous matching code and made
the comparisons a little more explicit and easier to understand. My
comments in the code should offer all the details involving this patch.
This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
find multiple dialogs with the same call-id. Since the callback
function was returning (CMP_MATCH | CMP_STOP) only the first item
found was being returned. I fixed this by making a new callback
function for finding multiple dialogs that only returns (CMP_MATCH)
on a match allowing for multiple items to be returned.
Review: https://reviewboard.asterisk.org/r/776/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This initially was created to work around the issue of
using a string comparison instead of a binary comparison
for IP addresses. It evolved a bit when test cases were
created and it was discovered that comparison of URI
parameters was not working exactly as it should.
sip_uri_cmp() and its helpers have been moved to reqresp_parser.c
and a new test has been added.
(closes issue #17662)
Reported by: oej
Review: https://reviewboard.asterisk.org/r/792
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are two changes here:
1. Since the externip setting can now have a port attached
to it, calling it "externip" is misleading. The option is now
documented and parsed as "externaddr." This also extends to the
"matchexterniplocally" setting. It is now documented and parsed
as "matchexternaddrlocally." The old names for the options may
still be used, but they are no longer used in the sip.conf.sample
file.
2. If no port is set for the externaddr, and UDP is the transport
to be used, then we will set the port of the externaddr to that of
the udpbindaddr. This was how things worked prior to the IPv6 merge,
so this is a regression fix.
(closes issue #17665)
Reported by: mmichelson
Patches:
17665.diff#2 uploaded by pprindeville (license 347)
Tested by: pprindeville
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
sip.conf configuration for the channel and for devices.
The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.
Review: https://reviewboard.asterisk.org/r/778/
Thanks to dvossel for the review and good advice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Retransmission of packets should not be based on how many packets were
sent, but instead on a timeout period. Depending on whether or not the
packet is for a INVITE or NON-INVITE transaction, the number of packets
sent during the retransmission timeout period will be different, so
timing out based on the number of packets sent is not accurate.
This patch fixes this by removing the retransmit limit and only stopping
retransmission after a timeout period is reached. By default this
timeout period is 64*(Timer T1) for both INVITE and non-INVITE
transactions. For more information on sip timer values refer to
RFC3261 Appendix A.
Review: https://reviewboard.asterisk.org/r/749/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some code improperly assumes that the sessions are still there, so revert the
change until I can find all of them and fix them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since we are only keeping the dialog around for retransmissions at this point
and there is no possibility that we are still handling RTP, go ahead and
destroy the RTP sessions. Keeping them alive for 32 past when they are used
is unnecessary and can lead to problems with having too many open file
descriptors, etc.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Part of the change with the IPv6 changes is to treat a host:port as
a single 'domain' entity. This test was not updated to have the correct
expectation after calling parse_uri().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
