* Eliminated RAII_VAR in get_outbound_endpoint().
* Simplify update_to() coding. However, this function can only be a NoOp
because the To string can only be a URI and not a name-address formatted
string.
* Simplify update_from() coding. Also fixed a code path modifying the
from string when the caller could still want to use the original string.
* Fixed msg_data_create() incompletely removing the "pjsip:" to then add
back the "sip:" string if needed. The code didn't handle the "pjsip:sip:"
case because it left the colon after pjsip in the string.
Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db
Currently when you add global headers from the dialplan both
the header in the dialplan and the globally configured header
are added to the resulting SIP INVITE. This change makes it
so the headers in the dialplan take precedence and are the
only ones added.
Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad
Confbridge announcements tend to block a channel while they are being
played. In some circumstances, this is warranted since you want that
particular channel not to hear the announcement (Example: "John Doe has
entered the conference"). For others it makes less sense.
This change first introduces methods for playing sounds asynchronously
into the conference. This is very similar to how synchronous sounds are
played, except the channel initiating the playback does not wait for the
sound to complete before moving on.
Asynchronous announcements are used for two circumstances:
* Sounds played for a user after they have left the bridge
* Sounds that play first to a single user and then the rest of the
conference (if the channel and conference use the same language)
ASTERISK-26289 #close
Reported by Mark Michelson
Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a
Following the Encrypt-all-the-things paradigm:
The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone
determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts
the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile
(sRTP is preferred aka optional; not mandatory). If the VoIP server does not
support sRTP and TLS, the phone shows an open padlock icon.
This paradigm is supported by several VoIP/SIP clients on default. Some
implementations even cannot be changed to RTP/sAVP. Therefore here, this
change allows Preferred sRTP for ingress. For egress, please, create a dial
plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP.
ASTERISK-20234 #close
Reported by: tootai
Tested by: tootai, Alexander Traud
patches:
srtp_patches.diff submitted by Matt Jordan
Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd
The code was referencing the config section as 'globals'
instead of 'general'. This change swaps it over to 'general'.
Change-Id: I9dfe7788f41c4a6754c77e103880dc1a747de7fe
Prior to this patch, a stop issued by a delete of a Playback resource
(indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop
the current media URI playing. Subsequent URIs specified by a playback
operation would then proceed on, even though we had just indicated to
the User that the Playback was finished *and* after they had just
'deleted' the resource. Whoops.
This patch corrects it by bailing out of the sequence of URIs to play if
one of them is terminated with an AST_CONTROL_STREAM_STOP indication.
ASTERISK-26341 #close
Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42
The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
http://downloads.digium.com/pub/telephony/ are now listed in the
"External" sections of the "Resource Modules" and "Codec Translators"
pages in menuselect. Any that are selected will automatically be
downloaded and installed when "make install" is run. Their LICENSE and
README (if avaialble) files will be installed to
ASTVARLIBDIR/documentation/thirdparty/<product_name>.
Example use with codecs:
The codecs/codecs.xml file is a menuselect style xml file that lists
the codecs to be included. Their support levels are 'external', which
triggers the download and install, and defaultenabled is no. Also
because codec_g729a is actually in a directory named codec_g729 on the
download server, the newly added 'member_data' element is used to
override the default of the directory name being the package name. You
can use the 'directory_name' attribute to keep default base URL
(http://downloads.digium.com/pub/telephony/) but use the new directory,
or you use the 'remote_url' attribute to specify a full URL to the
download directory. In this case, you must still follow the same
subdirectory naming conventions as that used for the packages located
at 'http://downloads.digium.com/pub/telephony'.
A new configure option '--with-externals-cache' was added and like
'--with-sounds-cache' it allows the installer to cache tarballs so
they're not downloaded every time.
To assist with the download and install process, each external package
now has a manifest.xml file that, among other things, contains a package
version and checksums for each file in the tarball. The manifest is
saved to both the cache directory and ASTMODDIR and together with the
manifest.xml on the downloads site, tells the install scripts whether
a download and/or update is needed.
bash and xmlstarlet are required for downloader operation. If they're
not installed, the external items in menuselect will be unavailable.
Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a
On heavy loaded system the TCP/TLS incoming calls could be
disconnected by pjproject while these calls are being
processed by asterisk which could use the session's memory pools.
If the session in the disconnected state then the session memory
pools were already freed, so we get segfault.
This patch adds a lifetime control on an INVITE session to pjproject.
The lifetime of the session is manipulated by calling
pjsip_inv_add_ref/pjsip_inv_dec_ref.
This patch uses these functions to inform pjproject that the
session is in use.
This patch adds check if the session state is not disconnected
and also checks if the memory pool is not NULL.
This patch also places tasks 'session_end' and 'session_end_completion'
into session's serializer to avoid race condition.
ASTERISK-26291 #close
Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7
Certain SNOM phones send so-called "optional crypto" in their SDP body.
Regular SRTP setup looks like this:
m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...
SNOM-style "optional crypto" looks like this:
m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...
A crypto line is supplied, but the m-line does not have SAVP.
When res_srtp.so is *not* loaded, then chan_sip.so treats the optional
crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the
incoming call with the following message:
WARNING: process_sdp: Failed to receive SDP offer/answer with
required SRTP crypto attributes for audio
For platforms that want to start providing SRTP this presents a
compatibility problem.
This changeset lets chan_sip handle the SDP as if no crypto-line was
supplied: i.e. accept the call as regular RTP, just like it did before
res_srtp was loaded.
Now you'll get this informative warning instead:
WARNING: Ignoring crypto attribute in SDP because RTP transport is
insecure
ASTERISK-23989 #close
Reported by: Olle Johansson
Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2
In any scenario in which the callee is not connected to the caller, the
current code in app_dial will crash due to raising a Dial End Stasis
Message after the callee channel has been hung up. This patch corrects
the error by simply moving the explicit hangup of the callee (peer)
channel until after the dial end message.
ASTERISK-25691 #close
Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d
If the callee selects option '5' using the Dial application's privacy
(P) option, the DIALSTATUS is erroneously set to ANSWER. This option
reflects the callee sending the caller to VoiceMail one time; the call
is definitely *not* ANSWERed in such a scenario. With this patch, the
DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that
is set when the 'send to VoiceMail every time' option is set.
ASTERISK-25691
Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358
Create an alternative to ast_sorcery_generic_alloc which uses astobj2
shared locking. Use this new method for the 'struct ast_sip_aor' allocator.
Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f
This allows standard ao2 functions to be used to release references to
an ast_named_lock. This change can cause less frequent locking of the
global named_locks container. The container is no longer locked when a
named_lock reference is being release except when this causes the
named_lock to be destroyed.
Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6
Previously, the buffer used for MP3 streamed from HTTP servers had a size of
1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1
minute. Only when the buffer is full does audio start to play.
For MP3 files streamed from a server, that is usually not a big deal as long as
the connection to the server is fast enough to supply that much data within a
second or two. For MP3 live streams however, it takes 1 minute to download 1
minute of audio, so without this change, app_mp3 wasn't really usable for MP3
live streams.
This commit changes the buffer size so that it covers 6 seconds of an MP3 file
streamed from a server and 0.5 seconds of an MP3 live stream. The latter is
identified by the use of a .m3u file extension.
app_mp3 so far only supported 8 kHz audio.
Now it always runs at the sample rate of the channel.
ASTERISK-26085 #close
Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0
In ARI, the channels API allows to hangup a channel with a hangup reason.
This commit adds a new reason "answered_elsewhere".
When using a SIP channel, this will eventually allow Asterisk to add a proper
"Reason" header to a CANCEL message.
ASTERISK-26321
Change-Id: Ia97675bd4acd6a7f58eb467953dfb94559f6583d
If the PJSIP endpoint's AOR with the permanent contact
was deleted from the realtime storage the res_pjsip module
continues trying to qualify this contact.
The error 'Unable to find an endpoint to qualify contact'
appeares every 'qualify_frequency' seconds.
This patch deletes this contact in this case.
The PJSIP endpoint's AOR with the permanent contact
is never qualified if it is added to realtime storage
after asterisk started.
This patch adds qualifying for the AOR's permanent contacts
on the first handling of this AOR.
ASTERISK-26319 #close
Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe
zuul
Richard Mudgett
Joshua Colp
Mark Michelson
Alexander Traud
Matt Jordan
George Joseph
Alexei Gradinari
Walter Doekes
Corey Farrell
Michael Kuron
Jean Aunis