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: http://svn.asterisk.org/svn/asterisk/trunk@370055 f38db490-d61c-443f-a65b-d21fe96a405b
Reduce memory consumption and add the H.264 and H.263 modules I shamefully neglected to add.


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370060 f38db490-d61c-443f-a65b-d21fe96a405b

Add French translation for chan_unistim phones on-screen menus. 



git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370066 f38db490-d61c-443f-a65b-d21fe96a405b


Deactivate timer for dialing entered number on hook switch hang up.

(closes issue ASTERISK-19554)
Reported by: Stefano Villani



git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370067 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@370072 f38db490-d61c-443f-a65b-d21fe96a405b
Fix an issue where specifying the resource in the username would cause authentication to fail.


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370073 f38db490-d61c-443f-a65b-d21fe96a405b
Add comments about the BUILD_NATIVE change

This is a significant change and mention of it should have gone into
UPGRADE.txt and CHANGES.
........

Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370082 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370083 f38db490-d61c-443f-a65b-d21fe96a405b
Fix a bug exposed by the testsuite where text streams would no longer be parsed correctly.


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370111 f38db490-d61c-443f-a65b-d21fe96a405b
Add missing namespace for old non-SASL based authentication.


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370116 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@370121 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@370126 f38db490-d61c-443f-a65b-d21fe96a405b
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
........

Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370132 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370133 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@370152 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@370157 f38db490-d61c-443f-a65b-d21fe96a405b

Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant.



git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370165 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@370171 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.asterisk.org/svn/asterisk/trunk@370177 f38db490-d61c-443f-a65b-d21fe96a405b
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.
........

Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370187 f38db490-d61c-443f-a65b-d21fe96a405b
Resolve severe memory leak in CEL logging modules.

A customer reported a significant memory leak using Asterisk 1.8. They
have tracked it down to ast_cel_fabricate_channel_from_event() in
main/cel.c, which is called by both in-tree CEL logging modules
(cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event
that they log.

The cause was an incorrect assumption about how data attached to an
ast_channel would be handled when the channel is destroyed; the data
is now stored in a datastore attached to the channel, which is
destroyed along with the channel at the proper time.

(closes issue AST-916)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2053/
........

Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370206 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370211 f38db490-d61c-443f-a65b-d21fe96a405b
callid logging: Issue test events when the callid is changed for a channel

Review: https://reviewboard.asterisk.org/r/2054/


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@370225 f38db490-d61c-443f-a65b-d21fe96a405b