asterisk/asterisk.git
14 months agoMerge "core: Don't stop generators when writing RTCP frames."
Jenkins2 [Fri, 7 Sep 2018 12:02:38 +0000 (07:02 -0500)]
Merge "core: Don't stop generators when writing RTCP frames."

14 months agoMerge "stasis_cache: Prune stasis_subscription_change messages"
Joshua Colp [Fri, 7 Sep 2018 10:40:36 +0000 (05:40 -0500)]
Merge "stasis_cache: Prune stasis_subscription_change messages"

14 months agoMerge "app_queue: Update realtime queuemembers after wait_a_bit(), not before"
Joshua Colp [Fri, 7 Sep 2018 09:48:30 +0000 (04:48 -0500)]
Merge "app_queue: Update realtime queuemembers after wait_a_bit(), not before"

14 months agocore: Don't stop generators when writing RTCP frames.
Joshua Colp [Wed, 5 Sep 2018 11:39:40 +0000 (11:39 +0000)]
core: Don't stop generators when writing RTCP frames.

Generators provide such functionality as tone generation or
silence generation. RTCP frames provide RTCP information and
should not stop generators from operating.

ASTERISK-28005

Change-Id: Ieadada07b068a7aa426e8763f1b73a18e1ac34a9

14 months agoapp_queue: Update realtime queuemembers after wait_a_bit(), not before
lvl [Mon, 3 Sep 2018 11:28:26 +0000 (13:28 +0200)]
app_queue: Update realtime queuemembers after wait_a_bit(), not before

This ensures the most up-to-date information is used for the next
call attempt.

ASTERISK-28032

Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce

14 months agores_pjproject: Add utility functions to convert between socket structures
Sean Bright [Tue, 28 Aug 2018 13:42:13 +0000 (09:42 -0400)]
res_pjproject: Add utility functions to convert between socket structures

Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address
needs to be rendered to a string and then parsed into the correct
structure. This also involves a call to getaddrinfo(3). The same is true
for the inverse operation.

Instead, because we know the internal structure of both ast_sockaddr and
pj_sockaddr, we can translate directly between the two without the
need for an intermediate string.

Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761

14 months agoMerge "http.c: Give HTTP error response when received lines are too long."
George Joseph [Thu, 6 Sep 2018 16:49:25 +0000 (11:49 -0500)]
Merge "http.c: Give HTTP error response when received lines are too long."

14 months agoMerge "iostream.c: Fix ast_iostream_gets() needlessly returning failure."
Jenkins2 [Wed, 5 Sep 2018 19:29:13 +0000 (14:29 -0500)]
Merge "iostream.c: Fix ast_iostream_gets() needlessly returning failure."

14 months agostasis_cache: Prune stasis_subscription_change messages
George Joseph [Thu, 30 Aug 2018 18:08:05 +0000 (12:08 -0600)]
stasis_cache: Prune stasis_subscription_change messages

The stasis cache provides a way to reconstruct the current state
of topic subscribers.  Unfortunately, since every subscribe and
unsubscribe is cached, the cache continues to grow unabated while
asterisk is running.  This patch removes subscribe messages from
the cache when the corresponding unsubscribe is received.

This patch also registers the cache containers with ao2 so that if
AO2_DEBUG is turned on, you can list the container and get its
stats from the CLI.

ASTERISK-27121

Change-Id: I3d18905e477f3721815da91f30da8d3fbb2d4f56

14 months agoMerge "app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done"
George Joseph [Wed, 5 Sep 2018 16:00:11 +0000 (11:00 -0500)]
Merge "app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done"

14 months agoMerge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch"
George Joseph [Wed, 5 Sep 2018 14:56:21 +0000 (09:56 -0500)]
Merge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch"

14 months agoapp_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done
Rodrigo Ramírez Norambuena [Mon, 3 Sep 2018 14:27:07 +0000 (11:27 -0300)]
app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done

Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8

14 months agopbx_config.c: Fix reloading module if initially declined to load
Chris-Savinovich [Wed, 15 Aug 2018 19:27:52 +0000 (15:27 -0400)]
pbx_config.c: Fix reloading module if initially declined to load

Added decline if extensions.conf file not available
when loading pbx_config, and also made sure everything
gets properly unregistered and/or destroyed on unload.

Change-Id: Ib00665106043b1be5148ffa7a477396038915854

14 months agoMerge "make config: os-release output error."
Joshua Colp [Fri, 31 Aug 2018 09:55:01 +0000 (04:55 -0500)]
Merge "make config: os-release output error."

14 months agohttp.c: Give HTTP error response when received lines are too long.
Richard Mudgett [Thu, 30 Aug 2018 19:42:06 +0000 (14:42 -0500)]
http.c: Give HTTP error response when received lines are too long.

Added a check when we receive a HTTP request line or header line that is
too long.  We now return an error response to the sender because we are
not able to process the request.

Change-Id: I6df2705435fd7dde4d5d3bdf7acec859cfb7c12d

14 months agoiostream.c: Fix ast_iostream_gets() needlessly returning failure.
Richard Mudgett [Wed, 29 Aug 2018 21:14:46 +0000 (16:14 -0500)]
iostream.c: Fix ast_iostream_gets() needlessly returning failure.

Providing a buffer larger than the internal buffer of ast_iostream_gets()
fails to get lines longer than the internal buffer.

* Made ast_iostream_gets() fill the supplied buffer with read data until
either a '\n' is found or the supplied buffer is filled just like fgets().

Change-Id: If18b3f6ee500e22f0633a68779ed09f7e0f305ed

14 months agoMerge "res_fax: Handle fax gateway being started more than once."
Joshua Colp [Thu, 30 Aug 2018 10:44:02 +0000 (05:44 -0500)]
Merge "res_fax: Handle fax gateway being started more than once."

14 months agoMerge "res_pjsip_transport_websocket: Properly set src_name for IPv6"
Joshua Colp [Thu, 30 Aug 2018 10:08:34 +0000 (05:08 -0500)]
Merge "res_pjsip_transport_websocket: Properly set src_name for IPv6"

14 months agores_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch
Richard Mudgett [Mon, 6 Aug 2018 20:37:05 +0000 (15:37 -0500)]
res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch

ASTERISK-27988

Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843

14 months agoMerge "Create --disable-binary-modules option."
George Joseph [Wed, 29 Aug 2018 11:31:54 +0000 (06:31 -0500)]
Merge "Create --disable-binary-modules option."

14 months agores_fax: Handle fax gateway being started more than once.
Joshua Colp [Wed, 29 Aug 2018 10:18:08 +0000 (07:18 -0300)]
res_fax: Handle fax gateway being started more than once.

The T.38 fax gateway state machine can cause the fax gateway
to be started more than once on a channel depending on the
responses of the remote endpoint. This would previously leak
the channel name, channel unique id, and underlying fax engine
state. This change instead makes it so that if the fax gateway
session is already present and not reserved the fax gateway
is not started again.

ASTERISK-27981

Change-Id: I552d95086860cb18f2522ee40ef47b13b6da2e0e

14 months agoMerge "alembic: increase uri column size"
Joshua Colp [Wed, 29 Aug 2018 10:20:01 +0000 (05:20 -0500)]
Merge "alembic: increase uri column size"

14 months agores_pjsip_transport_websocket: Properly set src_name for IPv6
Sean Bright [Tue, 28 Aug 2018 13:01:19 +0000 (09:01 -0400)]
res_pjsip_transport_websocket: Properly set src_name for IPv6

SIP responses over WebSockets when the client is using IPv6 have invalid
Via headers according to RFC 3261. The 'received' header parameter
should not be wrapped in brackets if it is an IPv6 address.

When src_name is populated by the built-in PJSIP transports, the code
uses pj_sockaddr_print() with 'flags' set to 0, meaning that the
brackets are not rendered around IPv6 addresses.

This may be related to ASTERISK~27101.

See also: https://github.com/onsip/SIP.js/pull/594

ASTERISK-28020 #close

Change-Id: I8ea9d289901b837512bee2ca2535e3dc14f04d77

14 months agoCreate --disable-binary-modules option.
Corey Farrell [Sun, 26 Aug 2018 18:18:42 +0000 (14:18 -0400)]
Create --disable-binary-modules option.

This new option can be passed for ./configure or
./tests/CI/buildAsterisk.sh to prevent download/install of binary
modules.

Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES
will result in binary modules being enabled even if the build target is
incompatible with those modules.  This includes CI scripts which enable
categories before disabling specific modules.

If more binary modules are offered in the future this will help avoid
accidentally downloading them if unwanted or incompatible.  Adding a
binary module will only require creating a new menuselect entry similar
to the existing ones, it will not be necessary to modify the CI scripts.

Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166

14 months agores/res_rtp_asterisk: remove debug traces generated by an empty frame
neutrino88 [Tue, 21 Aug 2018 12:59:08 +0000 (08:59 -0400)]
res/res_rtp_asterisk: remove debug traces generated by an empty frame

The realtime text timer pops regularly and sends text frames even if
the buffer is empty. This causes a lot of unecessary debug logging.

* Made red_write() test if we need to send a frame before calling
ast_rtp_write()

ASTERISK-28002
Reported by: Emmanuel BUU
Tested by: Emmanuel BUU

Change-Id: Icf81310c3b8080b615a42060afc02ab41f9523dd

14 months agoMerge "pbx_dundi: Added IPv6 support for dundi"
Jenkins2 [Mon, 27 Aug 2018 14:38:15 +0000 (09:38 -0500)]
Merge "pbx_dundi: Added IPv6 support for dundi"

14 months agoMerge "chan_sip: improved ip:port finding of peers for non-UDP transports."
George Joseph [Mon, 27 Aug 2018 12:17:39 +0000 (07:17 -0500)]
Merge "chan_sip: improved ip:port finding of peers for non-UDP transports."

14 months agochan_sip: improved ip:port finding of peers for non-UDP transports.
Jaco Kroon [Mon, 13 Aug 2018 13:12:21 +0000 (15:12 +0200)]
chan_sip: improved ip:port finding of peers for non-UDP transports.

Also remove function peer_ipcmp_cb since it's not used (according to
rmudgett).

Prior to b2c4e8660a9c89d07041271371151779b7ec75f6 (ASTERISK_27457)
insecure=port was the defacto standard.  That commit also prevented
insecure=port from being applied for sip/tcp or sip/tls.

Into consideration there are three sets of behaviour:

1.  "previous" - before the above commit.
2.  "current" - post above commit, pre this one.
3.  "new" - post this commit.

The problem that the above commit tried to address was guests over TCP.
It succeeded in doing that but broke transport!=udp with host!=dynamic.

This commit attempts to restore sane behaviour with respect to
transport!=udp for host!=dynamic whilst still retaining the guest users
over tcp.

It should be noted that when looking for a peer, two passes are made, the
first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer,
thus looking for full matches (IP + Port), the second pass sets
SIP_INSECURE_PORT, thus expecting matches on IP only where the matched
peer allows for that (in the author's opinion:  UDP with insecure=port,
or any TCP based, non-dynamic host).

In previous behaviour there was special handling for transport=tcp|tls
whereby a peer would match during the first pass if the utilized
transport was TCP|TLS (and the peer allowed that specific transport).

This behaviour was wrong, or dubious at best.  Consider two dynamic tcp
peers, both registering from the same IP (NAT), in this case either peer
could match for connections from an IP.  It's also this behaviour that
prevented SIP guests over tcp.

The above referenced commit removed this behaviour, but kept applying
the SIP_INSECURE_PORT only to WS|WSS|UDP.  Since WS and WSS is also TCP
based, the logic here should fall into the TCP category.

This patch updates things such that the previously non-explicit (TCP
behaviour) transport test gets performed explicitly (ie, matched peer
must allow for the used transport), as well as the indeterministic
source-port nature of the TCP protocol is taken into account.  The new
match algorithm now looks like:

1.  As per previous behaviour, IP address is matched first.

2.  Explicit filter with respect to transport protocol, previous
    behaviour was semi-implied in the test for TCP pure IP match - this now
    made explicit.

3.  During first pass (without SIP_INSECURE_PORT), always match on port.

4.  If doing UDP, match if matched against peer also has
    SIP_INSECURE_PORT, else don't match.

5.  Match if not a dynamic host (for non-UDP protocols)

6.  Don't match if this is WS|WSS, or we can't trust the Contact address
    (presumably due to NAT)

7.  Match (we have a valid Contact thus if the IP matches we have no
    choice, this will likely only apply to non-NAT).

To logic-test this we need a few different scenarios.  Towards this end,
I work with a set number of peers defined in sip.conf:

[peer1]
host=1.1.1.1
transport=tcp

[peer2]
host=1.1.1.1
transport=udp

[peer3]
host=1.1.1.1
port=5061
insecure=port
transport=udp

[peer4]
host=1.1.1.2
transport=udp,tcp

[peer5]
host=dynamic
transport=udp,tcp

Test cases for UDP:

1 - incoming UDP request from 1.1.1.1:
  - previous:
    - pass 1:
      * peer1 or peer2 if from port 5060 (indeterminate, depends on peer
        ordering)
      * peer3 if from port 5061
      * peer5 if registered from 1.1.1.1 and source port matches
    - pass 2:
      * peer3
  - current: as per previous.
  - new:
    - pass 1:
      * peer2 if from port 5060
      * peer3 if from port 5061
      * peer5 if registered from 1.1.1.1 and source port matches
    - pass 2:
      * peer3

2 - incoming UDP request from 1.1.1.2:
  - previous:
    - pass 1:
      * peer5 if registered from 1.1.1.2 and port matches
      * peer4 if source port is 5060
    - pass 2:
      * no match (guest)
  - current: as previous.
  - new as previous (with the variation that if peer5 didn't have udp as
          allowed transport it would not match peer5 whereas previous
          and current code could).

3 - incoming UDP request from anywhere else:
  - previous:
    - pass 1:
      * peer5 if registered from that address and source port matches.
    - pass 2:
      * peer5 if insecure=port is additionally set.
      * no match (guest)
  - current - as per previous
  - new - as per previous

Test cases for TCP based transports:

4 - incoming TCP request from 1.1.1.1
  - previous:
    - pass 1 (indeterministic, depends on ordering of peers in memory):
      * peer1; or
      * peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or
      * peer2 if the source port happens to be 5060; or
      * peer3 if the source port happens to be 5061.
    - pass 2: cannot happen since pass 1 will always find a peer.
  - current:
    - pass 1:
      * peer1 or peer2 if from source port 5060
      * peer3 if from source port 5060
      * peer5 if registered as 1.1.1.1 and source port matches
    - pass 2:
      * no match (guest)
  - new:
    - pass 1:
      * peer 1 if from port 5060
      * peer 5 if registered and source port matches
    - pass 2:
      * peer 1

5 - incoming TCP request from 1.1.1.2
  - previous (indeterminate, depends on ordering):
    - pass 1:
      * peer4; or
      * peer5 if peer5 registered from 1.1.1.2
    - pass 2: cannot happen since pass 1 will always find a peer.
  - current:
    - pass 1:
      * peer4 if source port is 5060
      * peer5 if peer5 registered as 1.1.1.2 and source port matches
    - pass 2:
      * no match (guest).
  - new:
    - pass 1:
      * peer4 if source port is 5060
      * peer5 if peer5 registered as 1.1.1.2 and source port matches
    - pass 2:
      * peer4

6 - incoming TCP request from anywhere else:
  - previous:
    - pass 1:
      * peer5 if registered from that address
    - pass 2: cannot happen since pass 1 will always find a peer.
  - current:
    - pass 1:
      * peer5 if registered from that address and port matches.
    - pass 2:
      * no match (guest)
  - new: as per current.

It should be noted the test cases don't make explicit mention of TLS, WS
or WSS.  WS and WSS previously followed UDP semantics, they will now
enforce source port matching.  TLS follow TCP semantics.

The previous commit specifically tried to address test-case 6, but broke
test-cases 4 and 5 in the process.

ASTERISK-27881 #close

Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2

14 months agoAMI: be less verbose when adding HTTP headers to AMI/HTTP messages.
Jaco Kroon [Mon, 20 Aug 2018 12:23:38 +0000 (14:23 +0200)]
AMI: be less verbose when adding HTTP headers to AMI/HTTP messages.

All HTTP/AMI message headers are being sent to the verbose channel.
There are multiple places this is happening.  Consolidate the loop into
a function.  Drop the debug/verbose message.

Convert to using ast_asprintf to perform the length calculation, memory
allocation and snprintf all in one step.

Change-Id: Ic45e673fde05bd544be95ad5cdbc69518207c1a1

14 months agoMerge "sample_configs: noload res_hep.so by default"
Jenkins2 [Thu, 23 Aug 2018 13:55:44 +0000 (08:55 -0500)]
Merge "sample_configs: noload res_hep.so by default"

14 months agoalembic: increase uri column size
Florian Floimair [Thu, 23 Aug 2018 11:57:31 +0000 (13:57 +0200)]
alembic: increase uri column size

When mobile SIP clients register with Asterisk that use some sort of
push notifications, the URI can get quite lengthy due to the
additional push-service annotations (things like tokens, pn-type, etc.)
contained in it.

ASTERISK-28022 #close

Change-Id: I4c7ceadc3bb405f3daf722641c8cd5ca4188cc37

14 months agosample_configs: noload res_hep.so by default
Matthew Fredrickson [Wed, 22 Aug 2018 15:50:55 +0000 (10:50 -0500)]
sample_configs: noload res_hep.so by default

Change disables loading of res_hep.so in default installation.  Loading
res_hep has a performance impact whether it's used or not.  This disables
loading of it in sample config files.

Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0

14 months agoMerge "res_pjsip: Reduce processing when a Contact is updated."
Joshua Colp [Wed, 22 Aug 2018 17:42:46 +0000 (12:42 -0500)]
Merge "res_pjsip: Reduce processing when a Contact is updated."

14 months agoapp_queue: Silence GCC 8 compiler warning
Sean Bright [Tue, 21 Aug 2018 18:50:33 +0000 (14:50 -0400)]
app_queue: Silence GCC 8 compiler warning

I'm only seeing an error in 14+, so I assume it is due to different
compiler options:

app_queue.c: In function ‘handle_queue_add_member’:
app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11
    bytes into a region of size 3 [-Werror=format-overflow=]
     sprintf(num, "%d", state);
                   ^~
app_queue.c:10234:18: note: directive argument in the range
    [-2147483648, 99]
     sprintf(num, "%d", state);
                  ^~~~

Compiler: gcc version 8.0.1 20180414 (experimental)
    [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2)

Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10

14 months agoMerge "AMI: Remove docs for nonexistent AMI ContactStatus event headers"
Joshua Colp [Tue, 21 Aug 2018 23:53:11 +0000 (18:53 -0500)]
Merge "AMI: Remove docs for nonexistent AMI ContactStatus event headers"

14 months agoMerge "pbx_dundi: Fix debug frame decode string."
Joshua Colp [Tue, 21 Aug 2018 23:52:46 +0000 (18:52 -0500)]
Merge "pbx_dundi: Fix debug frame decode string."

14 months agoMerge "pbx_dundi.c: Handle thread shutdown better."
George Joseph [Tue, 21 Aug 2018 12:26:01 +0000 (07:26 -0500)]
Merge "pbx_dundi.c: Handle thread shutdown better."

14 months agoMerge "pbx_dundi.c: Misc memory management fixes when destroying peers"
Joshua Colp [Tue, 21 Aug 2018 11:27:23 +0000 (06:27 -0500)]
Merge "pbx_dundi.c: Misc memory management fixes when destroying peers"

14 months agoAMI: Remove docs for nonexistent AMI ContactStatus event headers
Richard Mudgett [Mon, 20 Aug 2018 16:23:21 +0000 (11:23 -0500)]
AMI: Remove docs for nonexistent AMI ContactStatus event headers

Change-Id: I5736965c64c44338f7330e85a24bb46818607f19

14 months agoMerge "res_rtp_asterisk.c: Fix unused variable warnings"
George Joseph [Mon, 20 Aug 2018 16:31:20 +0000 (11:31 -0500)]
Merge "res_rtp_asterisk.c: Fix unused variable warnings"

14 months agoMerge "res_sorcery_realtime.c: Fix unqualified fetch warning."
George Joseph [Mon, 20 Aug 2018 15:57:05 +0000 (10:57 -0500)]
Merge "res_sorcery_realtime.c: Fix unqualified fetch warning."

14 months agoMerge "res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response."
George Joseph [Mon, 20 Aug 2018 15:55:01 +0000 (10:55 -0500)]
Merge "res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response."

14 months agores_pjsip: Reduce processing when a Contact is updated.
Joshua Colp [Mon, 6 Aug 2018 11:22:22 +0000 (11:22 +0000)]
res_pjsip: Reduce processing when a Contact is updated.

When a Contact is updated the only material change that qualify
support cares about is the underlying configuration for the AOR.
In this case we will update things with the new AOR information but
otherwise the callback to indicate the Contact has changed can be
ignored.

This is because it is only when a Contact is added or deleted that
material changes occur within the qualify support. An update can't
change the URI since it would result in a new Contact so it can be
ignored.

Change-Id: I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d

15 months agores_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response.
Richard Mudgett [Sat, 11 Aug 2018 00:28:45 +0000 (19:28 -0500)]
res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response.

We were still getting crashes after the first fix.  Somehow we receive a
non-2xx final response before we get a 200 final response.  With the
failure response we had already cleaned up and destroyed some data
structures.  When the unexpected 200 response comes in we crash.

* Add protection code to prevent processing another final T.38 reINVITE
response.

ASTERISK-27944

Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74

15 months agores_sorcery_realtime.c: Fix unqualified fetch warning.
Richard Mudgett [Thu, 9 Aug 2018 23:46:19 +0000 (18:46 -0500)]
res_sorcery_realtime.c: Fix unqualified fetch warning.

The allow_unqualified_fetch option for the sorcery realtime backend
blocked actually fetching all rows when the option is set to warn.

* Made issue a warning and actually do the request when
allow_unqualified_fetch=warn is set.

Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312

15 months agopbx_dundi: Added IPv6 support for dundi
Kirsty Tyerman [Mon, 11 Jun 2018 05:07:17 +0000 (15:07 +1000)]
pbx_dundi: Added IPv6 support for dundi

Change includes move to netsock2 library.

ASTERISK-27164
Reported-by: Adam Secombe

Change-Id: Ia9e8dc3d153de7a291dbda4bd87fc827dd2bb846

15 months agopbx_dundi.c: Misc memory management fixes when destroying peers
Richard Mudgett [Thu, 16 Aug 2018 02:31:45 +0000 (21:31 -0500)]
pbx_dundi.c: Misc memory management fixes when destroying peers

* In destroy_peer(), fixed memory leaks of lookup history strings and
qualify transactions when destroying peers.

* In destroy_peer(), fixed leaving the registerexpire scheduled callback
active when a peer is destroyed on a reload.  The reload marks and sweeps
peers so any peers not explicitly configured get destroyed.  Peers created
dynamically from the '*' peer will not exist until they re-register after
the reload.  These destroyed peers caused memory corruption when the
registerexpire timer expired.

* Made build_peer() not schedule any callbacks on the '*' peer
(empty_eid).  It is a special peer that is cloned to dynamically created
peers so it doesn't actually get involved in any message transactions.

* Made do_register_expire() remove the dundi/dpeers AstDB entry when a
peer registration expires.

* Fix deep_copy_peer() to not copy some things that cannot be copied to
the cloned peer structure.  Timers, message transactions, and lookup
history are specific to a peer instance.

* Made set_config() lock around processing the mappings configuration.

* Reordered unload_module() to handle load_module() declining the load due
to error.

Change-Id: Ib846b2b60d027f3a2c2b3b563d9a83a357dce1d6

15 months agopbx_dundi.c: Handle thread shutdown better.
Richard Mudgett [Thu, 16 Aug 2018 04:49:19 +0000 (23:49 -0500)]
pbx_dundi.c: Handle thread shutdown better.

Change-Id: Id52f99bd6a948fe6dd82acc0a28b2447a224fe87

15 months agopbx_dundi: Fix debug frame decode string.
Richard Mudgett [Wed, 15 Aug 2018 23:14:52 +0000 (18:14 -0500)]
pbx_dundi: Fix debug frame decode string.

* Fixed a typo in the name of the REGREQ frame decode string array.
* Fixed off by one range check indexing into the frame decode string
array.
* Removed some unneeded casts associated with the decode string array.

Change-Id: I77435e81cd284bab6209d545919bf236ad7933c2

15 months agopbx_dundi: Update sample config documentation.
Richard Mudgett [Thu, 16 Aug 2018 21:21:07 +0000 (16:21 -0500)]
pbx_dundi: Update sample config documentation.

Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849

15 months agores_rtp_asterisk.c: Fix unused variable warnings
Richard Mudgett [Wed, 15 Aug 2018 19:44:48 +0000 (14:44 -0500)]
res_rtp_asterisk.c: Fix unused variable warnings

Compiling without SRTP support installed resulted in some unused variable
warnings.  These warnings also showed that the srtp variable was obtained
and passed around some functions but not really used even when a system
has SRTP installed.

Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc

15 months agoMerge "res_resolver_unbound: Fix leak of config nameserver strings."
Joshua Colp [Fri, 17 Aug 2018 10:40:01 +0000 (05:40 -0500)]
Merge "res_resolver_unbound: Fix leak of config nameserver strings."

15 months agoMerge "res_pjsip: Resolve transport management leak at shutdown."
Joshua Colp [Fri, 17 Aug 2018 10:38:56 +0000 (05:38 -0500)]
Merge "res_pjsip: Resolve transport management leak at shutdown."

15 months agoMerge "res_odbc: Allow unload at shutdown."
Kevin Harwell [Thu, 16 Aug 2018 22:48:01 +0000 (17:48 -0500)]
Merge "res_odbc: Allow unload at shutdown."

15 months agoCI: Fixup for non-13 branches
George Joseph [Thu, 16 Aug 2018 18:51:51 +0000 (12:51 -0600)]
CI: Fixup for non-13 branches

Change-Id: I5e1d4a09e58b92b541bc8ed6f9e10e54c4e5101f

15 months agoCI: Final version of setting correct gerrit creds
George Joseph [Thu, 16 Aug 2018 18:28:03 +0000 (12:28 -0600)]
CI:  Final version of setting correct gerrit creds

Change-Id: I7729ecceedceb12f52bf18dae259846aa1d993b3

15 months agoCI: Add https credentials to gerrit checkouts
George Joseph [Thu, 16 Aug 2018 17:08:21 +0000 (11:08 -0600)]
CI:  Add https credentials to gerrit checkouts

If the review to be tested is in a project with restricted access,
we need to use the jenkins user's gerrit https credentials when we
do the checkout or the checkout will fail.

Change-Id: I9dc9994763c5ebfeb9f1cff60fb53f6902b7fd5f

15 months agoMerge "res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered"
George Joseph [Thu, 16 Aug 2018 14:45:33 +0000 (09:45 -0500)]
Merge "res/res_pjsip_sdp_rtp:  put rtcp-mux in answer only if offered"

15 months agomake config: os-release output error.
Rodrigo Ramírez Norambuena [Thu, 16 Aug 2018 14:04:36 +0000 (11:04 -0300)]
make config: os-release output error.

Fix not show the error
"/bin/sh: /etc/os-release: No such file or directory" when the command
'make config' is run in a System without systemv.

The instruction 'make config' pre execute the syntax
"$(shell . /etc/os-release && echo $$ID)" to identified if system is a
Slackware and Opensuse.

This change prevent show the message and is send to the /dev/null

Change-Id: I7f43e281a8d9405b2519fc653de82d9b8b645fdf

15 months agores/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered
Torrey Searle [Thu, 9 Aug 2018 07:34:17 +0000 (09:34 +0200)]
res/res_pjsip_sdp_rtp:  put rtcp-mux in answer only if offered

If in the initial sdp the caller doesn't include the line
a=rtcp-mux

Then asterisk shoud not include rtcp-mux in the response regardless
of rtcp-mux being enabled on the endpoint

ASTERISK-28007 #close

Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7

15 months agores_resolver_unbound: Fix leak of config nameserver strings.
Corey Farrell [Wed, 15 Aug 2018 19:49:01 +0000 (15:49 -0400)]
res_resolver_unbound: Fix leak of config nameserver strings.

Change-Id: I3f396316bb40d1ae6e91f5f688042420f1a540ed

15 months agores_pjsip: Resolve transport management leak at shutdown.
Corey Farrell [Wed, 15 Aug 2018 18:51:36 +0000 (14:51 -0400)]
res_pjsip: Resolve transport management leak at shutdown.

Cleanup idle check scheduled events at shutdown.

Change-Id: I61bfbb56bac69fe840c3242927d31ff3593be461

15 months agores_odbc: Allow unload at shutdown.
Corey Farrell [Wed, 15 Aug 2018 16:31:00 +0000 (12:31 -0400)]
res_odbc: Allow unload at shutdown.

This makes it possible for REF_DEBUG to report no leaks when loading
res_odbc.

Change-Id: I1a3dea786bd6e7f4820a6dd5cbaa197fa783ce93

15 months agores_pjsip: Fix leak in pjsip_options.
Corey Farrell [Wed, 15 Aug 2018 16:12:49 +0000 (12:12 -0400)]
res_pjsip: Fix leak in pjsip_options.

sip_options_get_endpoint_state_compositor_state leaked a reference to
the first available endpoint state compositor that was found.

Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7

15 months agoMerge "res_pjsip_caller_id: Add "party" parameter to RPID header."
George Joseph [Wed, 15 Aug 2018 14:44:43 +0000 (09:44 -0500)]
Merge "res_pjsip_caller_id: Add "party" parameter to RPID header."

15 months agoMerge "res_pjsip/rtp: No joint capabilities between streams."
Jenkins2 [Wed, 15 Aug 2018 14:38:12 +0000 (09:38 -0500)]
Merge "res_pjsip/rtp: No joint capabilities between streams."

15 months agoMerge "contrib/scripts: Make astgenkey executable"
George Joseph [Wed, 15 Aug 2018 12:50:48 +0000 (07:50 -0500)]
Merge "contrib/scripts: Make astgenkey executable"

15 months agoMerge "Build System: Improve ccache matching for different menuselect options."
Jenkins2 [Tue, 14 Aug 2018 18:41:32 +0000 (13:41 -0500)]
Merge "Build System: Improve ccache matching for different menuselect options."

15 months agocontrib/scripts: Make astgenkey executable
Richard Mudgett [Tue, 14 Aug 2018 16:55:42 +0000 (11:55 -0500)]
contrib/scripts: Make astgenkey executable

Change-Id: I11641d65592536dea9cbca5aa94a24c25d24dd5f

15 months agores_pjsip_caller_id: Add "party" parameter to RPID header.
Joshua Colp [Tue, 14 Aug 2018 12:29:18 +0000 (09:29 -0300)]
res_pjsip_caller_id: Add "party" parameter to RPID header.

This change adds the "party" parameter to the Remote-Party-ID header
which indicates which party the header information is applicable
to. In Asterisk this is determined on whether we are the calling
or called party. This is added to improve interoperability with some
implementations.

ASTERISK-28006

Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca

15 months agoMerge "CI: Add support for coverage processing."
Jenkins2 [Tue, 14 Aug 2018 12:34:13 +0000 (07:34 -0500)]
Merge "CI: Add support for coverage processing."

15 months agores_pjsip/rtp: No joint capabilities between streams.
Ben Ford [Tue, 7 Aug 2018 15:57:29 +0000 (10:57 -0500)]
res_pjsip/rtp: No joint capabilities between streams.

When a conference contained a mixture of audio/video and audio-only
users, a NOTICE message would pop up stating there are no joint
capabilities between streams. This happens because streams can never be
removed, but they can be in a REMOVED state. If we have the scenario
where user A joins with audio/video, user B joins with audio-only, and
user C joins with audio/video, then user A leaves, the message would
be triggered. That removed stream is still in the SDP, but Asterisk
would pass it through, causing it to be seen as a ulaw stream. A check
has been added for removed streams, setting their status to REMOVED when
handling negotiated SDPs.

Also addressed an issue where user A joins, then user B joins but does
not receive video until much later. Full frames were not being sent,
causing some PLI from the browser. Because the video was flowing in one
direction, the browser sets the SSRC to 1, but Asterisk was dropping the
PLI because of that. Added a check to see if the SSRC is 1 or not, which
sends full frames and allows video to flow between user A and user B.
This should only happen when dealing with PSFB or FUR, and in the case
of PSFB, only for PLI.

ASTERISK-27398

Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e

15 months agoapp_queue: set QUEUESTATUS to LEAVEEMPTY instead of CONTINUE
Ivan Poddubny [Sun, 12 Aug 2018 16:04:42 +0000 (18:04 +0200)]
app_queue: set QUEUESTATUS to LEAVEEMPTY instead of CONTINUE

When a call leaves a queue on leaveempty condition, QUEUESTATUS
must be set to LEAVEEMPTY, no matter whether Queue was executed with or
without the "c" (continue) option.

The regression was introduced in the fix for ASTERISK_25665.
The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was
overwritten in case when "c" is set, regardless of what was the cause
for leaving the queue.

ASTERISK-27973 #close
Reported-by: Valentin Safonov

Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c

15 months agoSample configs: Fix pjsip.conf syntax error.
Corey Farrell [Thu, 9 Aug 2018 20:25:41 +0000 (16:25 -0400)]
Sample configs: Fix pjsip.conf syntax error.

It is valid for a config file to be empty or contain only comments, but
not valid for a config value to be set when no uncommented context
exists.  This caused an error to be loged numerous times during start
when loading the default pjsip.conf.

Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6

15 months agoMerge "res_pjsip_registrar: Improve performance on inbound handling."
Joshua Colp [Wed, 8 Aug 2018 17:08:49 +0000 (12:08 -0500)]
Merge "res_pjsip_registrar: Improve performance on inbound handling."

15 months agoCI: Add support for coverage processing.
Corey Farrell [Fri, 20 Jul 2018 03:28:14 +0000 (23:28 -0400)]
CI: Add support for coverage processing.

Enable coverage with `./tests/CI/buildAsterisk.sh --coverage`.  This
will cause Asterisk to be compiled with coverage support.  It also
initializes 'before' coverage data for all sources.  Accept
--tested-only to disable modules which are not run by any test.
Enabling coverage also sets tested-only true by default.  To build
everything with coverage enabled use `--coverage --tested-only=0`.

./tests/CI/processCoverage.sh is used to process the coverage and
generate HTML reports.

Fix utils/check_expr2 which failed to compiled with coverage enabled.

Add status output 5 times per stage of astobj2_test_perf to ensure
remote CLI does not timeout when compiled with coverage.  Remote CLI
disconnects if no output is received for 60 seconds.  When coverage is
enabled it takes about 70 seconds for my laptop to run the stages of
this test, so with the change a message is printed every 14 seconds.

Change-Id: I890f7d5665087426ad7d3e363187691b9afc2222

15 months agoMerge "stasis: Reduce calculation of stasis message type hash."
Joshua Colp [Wed, 8 Aug 2018 10:54:02 +0000 (05:54 -0500)]
Merge "stasis: Reduce calculation of stasis message type hash."

15 months agoMerge "res_pjsip: Make pjlib.h consistently included."
Joshua Colp [Wed, 8 Aug 2018 10:53:53 +0000 (05:53 -0500)]
Merge "res_pjsip: Make pjlib.h consistently included."

15 months agoMerge "res_pjsip.h: Fix doxygen comments."
Joshua Colp [Wed, 8 Aug 2018 10:18:44 +0000 (05:18 -0500)]
Merge "res_pjsip.h: Fix doxygen comments."

15 months agoMerge "pjproject_bundled: Fix for Solaris builds. Do not undef s_addr."
Joshua Colp [Wed, 8 Aug 2018 10:10:32 +0000 (05:10 -0500)]
Merge "pjproject_bundled: Fix for Solaris builds. Do not undef s_addr."

15 months agores_pjsip.h: Fix doxygen comments.
Richard Mudgett [Mon, 6 Aug 2018 17:19:12 +0000 (12:19 -0500)]
res_pjsip.h: Fix doxygen comments.

Change-Id: I9cf97bdc756012d1f552ab007f4aa85e0ddb4e62

15 months agostasis: Reduce calculation of stasis message type hash.
Joshua Colp [Mon, 6 Aug 2018 11:36:22 +0000 (08:36 -0300)]
stasis: Reduce calculation of stasis message type hash.

When the stasis cache is used a hash is calculated for
retrieving or inserting messages. This change calculates
a hash when the message type is initialized that is then
used each time needed. This ensures that the hash is
calculated only once for the message type.

Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37

15 months agoMerge "dialplan_functions: wrong srtp use status report of a dialplan function"
Joshua Colp [Mon, 6 Aug 2018 13:34:20 +0000 (08:34 -0500)]
Merge "dialplan_functions: wrong srtp use status report of a dialplan function"

15 months agoMerge "pjproject_bundled: Find shared libraries in root --with-ssl=PATH."
Joshua Colp [Mon, 6 Aug 2018 10:28:47 +0000 (05:28 -0500)]
Merge "pjproject_bundled: Find shared libraries in root --with-ssl=PATH."

15 months agopjproject_bundled: Fix for Solaris builds. Do not undef s_addr.
Alexander Traud [Mon, 30 Jul 2018 12:49:08 +0000 (14:49 +0200)]
pjproject_bundled: Fix for Solaris builds. Do not undef s_addr.

The authors of PJProject undef s_addr because of some issue in Microsoft
Windows. However in Oracle Solaris, s_addr is not a structure member, but
defined to map to the real structure member.

Updates the patch from ASTERISK_20366

ASTERISK-27997

Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11

15 months agores_pjsip: Make pjlib.h consistently included.
Richard Mudgett [Fri, 3 Aug 2018 20:59:06 +0000 (15:59 -0500)]
res_pjsip: Make pjlib.h consistently included.

* Don't include pjlib.h twice in res_pjsip.h
* Consistently use #include <> form for pjproject includes.
(pjsip.h and pjlib.h)

Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7

15 months agodialplan_functions: wrong srtp use status report of a dialplan function
Salah Ahmed [Thu, 2 Aug 2018 19:37:16 +0000 (14:37 -0500)]
dialplan_functions: wrong srtp use status report of a dialplan function

If asterisk offer an endpoint with SRTP and that endpoint respond
with non srtp, in that case channel(rtp,secure,audio) reply wrong
status.

Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key:
Currently this flag has being set redundantly. In either case identical
or different remote_key this flag has being set. So if we
don't set it while we receive identical remote_key or non SRTP SDP
response then we can take decision of srtp use by using that flag.

ASTERISK-27999

Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7

15 months agoMerge "BuildSystem: Enable ncurses for menuselect in Solaris 11."
Kevin Harwell [Fri, 3 Aug 2018 18:29:11 +0000 (13:29 -0500)]
Merge "BuildSystem: Enable ncurses for menuselect in Solaris 11."

15 months agoMerge "pjsip_wizard.conf.sample: Update remote_hosts description."
Kevin Harwell [Fri, 3 Aug 2018 18:27:24 +0000 (13:27 -0500)]
Merge "pjsip_wizard.conf.sample: Update remote_hosts description."

15 months agoMerge "res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header"
Kevin Harwell [Fri, 3 Aug 2018 18:26:30 +0000 (13:26 -0500)]
Merge "res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header"

15 months agopjproject_bundled: Find shared libraries in root --with-ssl=PATH.
Alexander Traud [Mon, 30 Jul 2018 11:05:34 +0000 (13:05 +0200)]
pjproject_bundled: Find shared libraries in root --with-ssl=PATH.

The script configure from Teluu expects shared libraries (.so) in a subfolder
called 'lib', when --with-xyz=PATH is specified. However for OpenSSL, the
default location is the root of the source folder = PATH. Furthermore, Asterisk
supports both, 'lib' and root. For consistency and because Asterisk is using
(only) OpenSSL in PJProject, it is enhanced to support both locations, just
like Asterisk.

ASTERISK-27995

Change-Id: I8eb916a88b6b8c22e29bb40bee8faaca6c73406f

15 months agores_pjsip_registrar: Improve performance on inbound handling.
Joshua Colp [Wed, 1 Aug 2018 14:45:04 +0000 (14:45 +0000)]
res_pjsip_registrar: Improve performance on inbound handling.

This change removes a sorcery lookup for retrieving all
contacts at the end of the registration process by keeping
track of the contacts that are added/updated/deleted.

This ensures at the end of the process the container of
contacts we have is the current state.

Pool usage has also been reduced by allocating one for
usage throughout the handling of a REGISTER and resetting
it to a clean state. This ensures that in most cases
we allocate once and just reuse it.

ASTERISK-28001

Change-Id: I1a78b2d46f9a2045dbbff1a3fd6dba84b612b3cb

15 months agoMerge "thirdparty/pjproject: fix deadlock in response retransmissions"
Joshua Colp [Thu, 2 Aug 2018 12:12:35 +0000 (07:12 -0500)]
Merge "thirdparty/pjproject: fix deadlock in response retransmissions"

15 months agoMerge "BuildSystem: Enable Jansson in Solaris 11."
Joshua Colp [Thu, 2 Aug 2018 11:39:07 +0000 (06:39 -0500)]
Merge "BuildSystem: Enable Jansson in Solaris 11."

15 months agothirdparty/pjproject: fix deadlock in response retransmissions
Torrey Searle [Tue, 17 Jul 2018 12:13:43 +0000 (14:13 +0200)]
thirdparty/pjproject: fix deadlock in response retransmissions

The tdata containing the response can be shared by both the dialog
object and the tsx object.  In order to prevent the race condition
between the tsx retransmission and the dialog sending a response,
clone the tdata before modifying it for the dialog send response.

ASTERISK-27966 #close

Change-Id: Ic381004a3a212fe1d8eca0e707fe09dba4a6ab4e

15 months agoBuild System: Improve ccache matching for different menuselect options.
Corey Farrell [Wed, 1 Aug 2018 04:54:11 +0000 (00:54 -0400)]
Build System: Improve ccache matching for different menuselect options.

Changing any Menuselect option in the `Compiler Flags` section causes a
full rebuild of the Asterisk source tree.  Every enabled option causes
a #define to be added to buildopts.h, thus breaking ccache caching for
every source file that includes "asterisk.h".  In most cases each option
only applies to one or two files.  Now we only define those options for
the specific sources which use them, this causes much better cache
matching when working with multiple builds.  For example testing code
with an without MALLOC_DEBUG will now use just over half the ccache
size, only main/astmm.o will have two builds cached instead of every
file.

Reorder main/Makefile so _ASTCFLAGS set on specific object files are all
together, sorted by filename.  Stop adding -DMALLOC_DEBUG to CFLAGS of
bundled pjproject, this define is no longer used by any header so only
serves to break cache.

The only code change is a slight adjustment to how main/astmm.c is
initialized.  Initialization functions always exist so main/asterisk.c
can call them unconditionally.  Additionally rename the astmm
initialization functions so they are not exported.

Change-Id: Ie2085237a964f6e1e6fff55ed046e2afff83c027

15 months agoMerge "res_rtp_asterisk: In Developer Mode, do not require OpenSSL."
Joshua Colp [Wed, 1 Aug 2018 09:23:06 +0000 (04:23 -0500)]
Merge "res_rtp_asterisk: In Developer Mode, do not require OpenSSL."

15 months agoMerge "res_pjsip_pubsub: Use ast_true for "prune_on_boot"."
Joshua Colp [Tue, 31 Jul 2018 21:11:32 +0000 (16:11 -0500)]
Merge "res_pjsip_pubsub: Use ast_true for "prune_on_boot"."

15 months agopjsip_wizard.conf.sample: Update remote_hosts description.
Richard Mudgett [Tue, 31 Jul 2018 16:24:08 +0000 (11:24 -0500)]
pjsip_wizard.conf.sample: Update remote_hosts description.

Remove the note that SRV records are not supported as that is no longer
true.

ASTERISK-27993

Change-Id: Id0dd6ef40e52702be9727a2b6122216cb00bb4ca

15 months agoCI: Add optional uninstall step before installing asterisk
George Joseph [Fri, 27 Jul 2018 18:23:02 +0000 (12:23 -0600)]
CI: Add optional uninstall step before installing asterisk

Change-Id: I7dedf1e925eafc3a0adf01dd9dfbe44eb642aab7