Commit Graph

22741 Commits

Author SHA1 Message Date
Kevin P. Fleming 79087cbbd5 Ensure that all ast_datastore_info structures are 'const'.
While addressing a bug, I came across a instance of 'struct ast_datastore_info'
that was not declared 'const'. Since the API already expects them to be
'const', this patch changes the declarations of all existing instances
that were not already declared that way.
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Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18 17:18:20 +00:00
Joshua Colp 8401e81383 Fix a crash in pjnath when starting an ICE connectivity check and immediately destroying the ICE session.
The initial ICE connectivity check is scheduled as a timer item that is to be executed immediately. It is possible for this timer item to start executing while the ICE session it is working on is destroyed. To reduce the chance of this any timer items that need to be immediately executed will be executed within the thread that has started the initial ICE connectivity check.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18 15:15:41 +00:00
Joshua Colp cbdb2dbb0e Fix a crash occurring as a result of excess stack usage.
This fix involves moving the allocation of some temporary codec structures to the heap and also reduces the number of maximum payloads to something more sane for both regular and low memory builds.

(closes issue ASTERISK-20140)
Reported by: jonnt


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18 11:38:05 +00:00
Igor Goncharovskiy 9278b5e51e Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18 07:17:00 +00:00
Joshua Colp cd91570bc6 Add pubsub unsubscription support so subscriptions do not linger for MWI and device state progatation.
The pubsub code did not attempt to remove subscriptions at all. This has now changed so that if a client is being disconnected it will unsubscribe. It will also unsubscribe at connection time so if it unexpectedly disconnected duplicate subscriptions will not occur.

(closes issue ASTERISK-19882)
Reported by: mattvryan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-17 19:05:36 +00:00
Joshua Colp 44345b0973 Fix a crash as a result of propagating MWI or device state over XMPP when the client is disconnected.
The MWI and device state propagation code wrongly assumes that an XMPP client connection will remain established at all times. This fix corrects that by making the lifetime of the subscription the same as the lifetime of the connection itself. As the connection is established and disconnected the subscription itself is created and destroyed.

(closes issue ASTERISK-18078)
Reported by: elguero


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-17 16:32:10 +00:00
Walter Doekes 6027b26fa7 Code cleanup and bugfix in chan_sip outboundproxy parsing.
The bug was clearing the global outboundproxy when a peer-specific
outboundproxy was bad. The cleanup reduces duplicate code.

Review: https://reviewboard.asterisk.org/r/2034/
Reviewed by: Mark Michelson
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Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370132 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 19:58:00 +00:00
Joshua Colp fdd39eae58 Fix an issue where a service discovery request could crash Asterisk.
A server sending a service discovery request to us may or may not put a from attribute in the message. If the from attribute is present use it in the to attribute for the result. If the from attribute is not present do not add a to attribute.

(issue ASTERISK-16203)
Reported by: wubbla


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 19:14:29 +00:00
Joshua Colp 3b59ab1c77 Fix a bug where some XMPP servers would reject authentication. We need to use the user portion of the JID and not the full configured username.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 17:26:40 +00:00
Joshua Colp 7a78aa39d1 Add missing namespace for old non-SASL based authentication.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 16:54:55 +00:00
Joshua Colp f234eae9ee Fix a bug exposed by the testsuite where text streams would no longer be parsed correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 15:08:53 +00:00
Kinsey Moore 25e721ee9b Add comments about the BUILD_NATIVE change
This is a significant change and mention of it should have gone into
UPGRADE.txt and CHANGES.
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Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370082 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 14:02:10 +00:00
Joshua Colp 5d20f60337 Fix an issue where specifying the resource in the username would cause authentication to fail.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 12:58:18 +00:00
Joshua Colp e938737570 Add support for SIP over WebSocket.
This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb.

Review: https://reviewboard.asterisk.org/r/2008


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 12:35:04 +00:00
Igor Goncharovskiy f9c3585d73 Deactivate timer for dialing entered number on hook switch hang up.
(closes issue ASTERISK-19554)
Reported by: Stefano Villani



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 07:38:18 +00:00
Igor Goncharovskiy 95ac8f4743 Add French translation for chan_unistim phones on-screen menus.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 07:34:12 +00:00
Joshua Colp acb5f5f824 Reduce memory consumption and add the H.264 and H.263 modules I shamefully neglected to add.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13 18:41:07 +00:00
Joshua Colp a693fd1d87 Add support for parsing SDP attributes, generating SDP attributes, and passing it through.
This support includes codecs such as H.263, H.264, SILK, and CELT. You are able to set up a call and have attribute information pass. This should help considerably with video calls.

Review: https://reviewboard.asterisk.org/r/2005/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13 16:49:40 +00:00
Tzafrir Cohen 2603707f30 live_ast: don't set working directory
contrib/scripts/live_ast currently assumes that it is being run from the
top-level directory of the source tree. It creates a script that will
also set the working directory.

This fix avoids the need to set the working directory if the caller sets
LIVE_AST_BASE_DIR instead.

It relies on realpath for that. If realpath is not available, it will
fall back to the original behaviour.

Review: https://reviewboard.asterisk.org/r/2027/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13 00:05:46 +00:00
Terry Wilson a7dfafdc56 Handle deprecated (aliased) option names with the config options api
Add a simple way to register "deprecated" option names that alias to a
different "current" name.

Review: https://reviewboard.asterisk.org/r/2026/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 21:43:09 +00:00
Richard Mudgett 9773d2351b Add missing ast_hangup() calls on some analog exception paths.
Make starting analog_ss_thread() or __analog_ss_thread() failure paths
hangup the channel.
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Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370025 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 20:28:07 +00:00
Kinsey Moore c1354af599 Include Expires header for SIP PUBLISH requests
RFC3903 requres SIP PUBLISH requests to have Expires headers, so add
them.

Review: https://reviewboard.asterisk.org/r/2003/
Patch-by: gareth
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Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 370015 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 20:06:23 +00:00
Kinsey Moore 65fe6976ae Prevent double uri_escaping in chan_sip when pedantic is enabled
If pedantic mode is enabled, outbound invites will have double-escaped
contacts.  This avoids setting an already-escaped string into a field
where it is expected to be unescaped.

(closes issue ASTERISK-20023)
Reported by: Walter Doekes
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Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369994 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 19:05:11 +00:00
Michael L. Young 6761812566 Correct Documentation For DEC Function
The documentation for DEC in func_math.c was incorrect.  Looks like a copy and
paste error.

(Closes issue ASTERISK-20095)
Reported by: Billy Chia
Tested by: Michael L. Young
Patches:
    func_math.patch uploaded by Billy Chia (license 6381)
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Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369971 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 14:38:44 +00:00
Michael L. Young 9bd9eb809c Reverting last merge since it wasn't completed properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 14:36:44 +00:00
Michael L. Young a8c12c6e67 Correct Documentation For DEC Function
The documentation for DEC in func_math.c was incorrect.  Looks like a copy and
paste error.

(Closes issue ASTERISK-20095)
Reported by: Billy Chia
Tested by: Michael L. Young
Patches:
    func_math.patch uploaded by Billy Chia (license 6381)
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Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12 14:27:56 +00:00
Jonathan Rose 10afdf3a2a Named ACLs: Introduces a system for creating and sharing ACLs
This patch adds Named ACL functionality to Asterisk. This allows system
administrators to define an ACL and refer to it by a unique name. Configurable
items can then refer to that name when specifying access control lists.
It also includes updates to all core supported consumers of ACLs. That includes
manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk
by Olle E. Johansson and provides a subset of the Named ACL functionality
implemented in that branch. For more information on this feature, see acl.conf
and/or the Asterisk wiki.

Review: https://reviewboard.asterisk.org/r/1978/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 18:33:36 +00:00
Tilghman Lesher 6190ae4430 Allow the REALTIME() function to report errors back to the caller.
Also, do more error checking on the arguments specified to the REALTIME()
function and clarify the documentation.  While I was editing the file, a
few coding guidelines fixups, as well.

Review: https://reviewboard.asterisk.org/r/2031/
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Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369938 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 17:16:50 +00:00
Matthew Jordan 92a65de048 Don't perform an XInclude to a document node that may not always be present
Because some of the manager events are defined in the top of the source, due
to the macro calls not containing all necessary information to have the
documentation colocated with the call itself, several include statements were
failing when built with 'make'.  While this did not cause any problems in
compilation or validation, it did result in a number of warnings being dumped
to stderr.

This patch changes those references such that they always resolve, regardless
of the documentation build options.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 17:14:45 +00:00
Joshua Colp a25b4b7457 Do not consider failure to read the configuration file in chan_motif to be a show stopper for loading Asterisk by returning decline instead of failure.
(closes issue ASTERISK-20103)
Reported by: Terry Wilson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 16:42:01 +00:00
Matthew Jordan 9bc2127d7b Fix validation errors when producing documentation using default build script
The awk script parses out the first instance of the DOCUMENTATION tag that it
finds within a file.  If a file did not previously have a DOCUMENTATION tag
but received one due to it having an AMI event, then the XML fragment
associated with the AMI event was erroneously placed in the resulting XML
file.  Without the python scripts, these XML fragments will not validate.

This patch adds DOCUMENTATION tags at the top of those files that did
not previously have them to prevent the awk script from pulling AMI event
documentation.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11 02:06:05 +00:00
Matthew Jordan 2ffae5745d Add some additional documentation for core AMI events
This patch adds some basic documentation for a number of modules.  This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri.  The DTD
has also been updated to allow referencing of AMI commands.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 22:26:27 +00:00
Kinsey Moore 3805e2ae4d Fix failing SDP_offer_answer test
Asterisk now generates image stream declinations with the same
transport case that it used to before the stream declination
improvements. (udptl vs UDPTL)

(closes issue SWP-4736)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 15:36:37 +00:00
Joshua Colp 55871d3a67 Add additional description stanza names from the old Google Talk protocol which is used with Google Voice.
(closes issue ASTERISK-20114)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 15:25:12 +00:00
Joshua Colp 74ebe6d5ab Respect codec preference order when adding codecs to a media description.
This change allows an endpoint in motif.conf to be configured with a preference of G.722 and fallback of ulaw. With Google this allows communication with Google Talk clients to use G.722 while when using Google Voice ulaw will be used.

(closes issue ASTERISK-20114)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 14:00:05 +00:00
Kinsey Moore 6416a246ed Improve Goto and GotoIf related documentation
Correct documentation on labeliftrue and labeliffalse parameters of
GotoIf() and update several other locations that use the same syntax.

(closes issue ASTERISK-20007)
Patch-by: Leif Madsen
Reported-by: WIMPy
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Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369871 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 13:40:32 +00:00
Matthew Jordan b1bb826350 Fix initial loading problem with res_curl
When the OpenSSL duplicate initialization issues were resolved in r351447,
res_curl could fail to load if it checked SSL_library_init after SSL
initialization completed.  This is due to the SSL_library_init stub returning
a value of 0 for success, as opposed to a value of 1.  OpenSSL uses a value of
1 to indicate success - in fact, SSL_library_init is documented to always return
1.  Interestingly, the CURL libraries actually checked the return value - the fact
that nothing else that depends on OpenSSL was having problems loading probably means
they don't check the return value.

(closes issue AST-924)
Reported by: Guenther Kelleter
patches:
  (AST-924.patch license #6372 uploaded by Guenther Kelleter)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 13:34:15 +00:00
Joshua Colp 7296b670d4 Add required items for Google video support.
This adds legacy STUN support for RTCP sockets, adds RTCP candidates to the Google transport information, and adds required codec parameters.

(closes issue ASTERISK-20106)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10 11:49:18 +00:00
Joshua Colp 8f162be802 When receiving a STUN binding request send one out as the Google Talk client uses this as a method to determine if the remote party is still reachable or not.
Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis.

(closes issue ASTERISK-20107)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 22:38:25 +00:00
Joshua Colp 7baa8bf43d Add support for exposing the received contact URI and also for setting the request URI in messages.
(closes issue AST-911)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 19:51:37 +00:00
Joshua Colp b46e1b45e4 Force the clock rate of G.722 to be 16000 when using the Google transports as it is 8000 elsewhere.
(closes issue ASTERISK-20105)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 19:05:25 +00:00
Joshua Colp 04504e80a3 Document that multiple endpoints using the same connection is not supported.
(closes issue ASTERISK-20104)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 18:54:43 +00:00
Jason Parker b1cd995273 Add Digium phones context to sip_notify sample config.
This makes it so that they can be reconfigured remotely.

(closes issue ASTERISK-19910)
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Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369819 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 17:07:06 +00:00
Joshua Colp 31beb35f47 Fix an issue where media would not flow for situations where the legacy STUN code is in use.
The STUN packets should *not* be blocked by strict RTP.

(closes issue ASTERISK-20102)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 16:44:24 +00:00
Joshua Colp 540f4b81f9 Add additional namespaces for Google Talk which are used for the gmail client.
(closes issue ASTERISK-20101)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 16:27:47 +00:00
Joshua Colp fa0bcb6c70 Fix dependency to be on res_xmpp. Long ago in a galaxy far far away it used to use res_jabber.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 15:58:36 +00:00
Jonathan Rose 60bc927579 chan_sip: Fix small behavioral change accidentally introduced in r369750
When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also
inadvertently changed the return value, which would likely make the indication
not be sent in audio. This fixes that while still removing the warning message.
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Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369793 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09 14:54:22 +00:00
Joshua Colp a3fa37b8cf Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif.
This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either.
These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold,
unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications.
The original Google Talk protocol is also supported for Google Voice interoperability.

You may ask yourself though where the name motif comes from... and I would say to you... music!

motif: a perceivable or salient recurring fragment or succession of notes

Sorta like a jingle!

Review: https://reviewboard.asterisk.org/r/1917/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-07 17:06:51 +00:00
Kinsey Moore db59a3f123 Remove unnecessary generation of informational cause frames
It is not necessary to generate information cause code frames on every
protocol event that occurs.  This removes all the instances where the
frame was not conveying a cause code and was instead just conveying a
protocol-specific message.  This also corrects the generation of the
message associated with disconnects for MFC/R2 to use the MFC/R2
specific text for the disconnect cause.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06 22:03:44 +00:00
Jonathan Rose 49aa47171b chan_sip: Add case for FLASH control frames so that we don't display a warning.
chan_sip channels can receive flash control frames when connected to analog
phones and possibly for other reasons. There really isn't a reason to warn when
these frames are received, we can safely ignore them.

Patches:
    dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)
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Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369751 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-06 21:28:26 +00:00