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into 13
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RTP." into 13
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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
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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
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channel" into 13
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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
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Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09
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Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838
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Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3
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Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2
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Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15
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Change-Id: If35174614545727817d329c60ba4456c028941b5
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Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
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* Make ast_format_cap_get_names() NULL tolerant.
ASTERISK-26331 #close
Reported by: CGI.NET
Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3
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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
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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
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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
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into 13
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into 13
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As described in issue ASTERISK-26282 the AEL parser creates macros with
extension '~~s~~'. app_macro searches only for extension 's' so the
created extension cannot be found. with this patch app_macro searches for
both extensions and performs the right extension.
ASTERISK-26282 #close
Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb
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Previously, if context A was including context B and context B was including
context A, i.e. if there was a circular dependency between contexts, then
calling manager_show_dialplan_helper could lead to an infinite recursion,
resulting in a crash.
This commit applies the same solution as the one implemented in the
show_dialplan_helper function. The manager_show_dialplan_helper and
show_dialplan_helper functions contain lots of code in common, but the former
was missing the "infinite recursion avoidance" code.
ASTERISK-26226 #close
Change-Id: I1aea85133c21787226f4f8442253a93000aa0897
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The reason for the disable is that while Asterisk works fine with older
libsrtp versions, newer versions of pjproject won't compile with them.
Debian 6 for instance, has libsrtp 1.4.4 which is older than what
pjproject is expecting.
We don't use most of pjmedia but we DO use it for SDP negotiation.
Luckily disabling srtp in pjmedia doesn't interfere with it's ability
to negitiate a secure channel. The proper crypto attributes are
negotiated in both directions.
ASTERISK-26279 #close
Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2
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In your Diaplan, if you specify
same => n,Set(CHANNEL(secure_bridge_media)=1)
same => n,Set(CHANNEL(secure_bridge_signaling)=1)
only the SIP channel driver chan_sip supports this. All other channels drivers
like res_pjsip fail. In case of failure, the original sRTP source code released
the whole channel, even if not hung-up, yet. This change does not release the
channel but instead hangs-up the channel.
ASTERISK-26306
Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db
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When using the migration script sip_to_pjsip.py, and your sip.conf is
configured with bindaddr=::, two transports are written to pjsip.conf, one for
0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4
and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface
like in chan_sip.
Furthermore, the script internal functions "build_host" and "split_hostport"
did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change
makes sure, even such addresses are parsed correctly.
ASTERISK-26309
Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48
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When dialing channels it is possible that they may not ever
leave the not in use state (Local channels in particular) by
the time we cancel them. If this occurs but we know they were
dialed we explicitly remove them from the pending members
container so that subsequent call attempts occur.
ASTERISK-26299 #close
Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65
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We may check a global config option hundreds of times a second or more.
Asking sorcery for the global configuration from the config files backend
involves several allocations and container traversals. Using realtime
without a memory cache is a lot worse because you have to lookup in the
realtime database each time to reconstitute the sorcery object. With a
memory cache for realtime, there is about the same amount of overhead as
for config files. Either way, it is still fairly expensive to access the
sorcery object that much.
* Cache the global config options so we can access them faster. You must
now always perform a res_pjsip reload to change the global options.
Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7
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ast_channel_get_t38_state() calls ast_channel_queryoption() with
AST_OPTION_T38_STATE. If the passed in channel is a local channel then a
deadlock can happen if a channel lock is held when called.
* Made ast_channel_get_t38_state() callers not hold a channel lock before
calling.
* Update ast_channel_get_t38_state() doxygen to note that no channel locks
can be held when calling the function.
ASTERISK-26203 #close
Reported by: Etienne Lessard
ASTERISK-24822 #close
Reported by: David Brillert
ASTERISK-22732 #close
Reported by: Richard Mudgett
Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214
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ASTERISK-25980 added the FAXMODE channel variable to res_fax.c.
Unfortunately, it also introduced a deadlock potential because
set_channel_variables() which sets FAXMODE can be called during a
masquerade. The ast_channel_get_t38_state() which gets the value used to
set FAXMODE cannot be called with the channel locked. As a result, local
channels can deadlock because of how they must acquire the locks necessary
to operate.
The intent of FAXMODE is for dialplan to know how a fax was transferred
after the fax completes. However, the previous patch sets FAXMODE to the
channel's current T.38 state AFTER the fax has completed and where T.38
may have already disconnected.
* Set FAXMODE based upon T.38 negotiations exchanged either with the fax
applications or the fax framehooks.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1
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fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be
called with any channel locks already held. A deadlock can happen if the
function is operating on a local channel.
* Made fax_gateway_indicate_t38() unlock the channel before calling
ast_indicate_data() since fax_gateway_indicate_t38() is always called with
the channel locked.
* Made fax_gateway_indicate_t38() return void since nothing cared about
its return value.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407
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Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7
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The framehook container could become corrupted if the channel lock is not
held before calling.
Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584
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The framehook container could become corrupted if the channel lock is not
held before calling.
Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438
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ast_multicast_rtp_create_options now checks for NULL or empty options
Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362
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