asterisk/asterisk.git
2 years agoinstall_prereq: Remove unpackaged version of jansson.
Corey Farrell [Mon, 10 Sep 2018 15:12:55 +0000 (11:12 -0400)]
install_prereq: Remove unpackaged version of jansson.

This is removed in favor of ./configure --with-jansson-bundled.  The
install-unpackaged command would only install jansson once, so once
installed it would never update, where the bundled copy will be kept up
to date.

Change-Id: Ideab1f65419608d3795aa608e9da873823cc42d3

2 years agoMerge "res_srtp.c: Show linked version of libsrtp on module init"
George Joseph [Mon, 17 Sep 2018 14:23:52 +0000 (09:23 -0500)]
Merge "res_srtp.c: Show linked version of libsrtp on module init"

2 years agoMerge "res_pjsip: Log IPv6 addresses correctly"
George Joseph [Mon, 17 Sep 2018 13:34:02 +0000 (08:34 -0500)]
Merge "res_pjsip: Log IPv6 addresses correctly"

2 years agoCI: Fix typo in testsuite git checkout
George Joseph [Mon, 17 Sep 2018 12:10:18 +0000 (06:10 -0600)]
CI: Fix typo in testsuite git checkout

Change-Id: I30024515e5b00a5044fd39fbff27d818f016b719

2 years agores_srtp.c: Show linked version of libsrtp on module init
Sean Bright [Sun, 16 Sep 2018 11:08:29 +0000 (07:08 -0400)]
res_srtp.c: Show linked version of libsrtp on module init

Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342

2 years agores_pjsip: Log IPv6 addresses correctly
Sean Bright [Fri, 7 Sep 2018 14:40:05 +0000 (10:40 -0400)]
res_pjsip: Log IPv6 addresses correctly

Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name
store IPv6 addresses without enclosing brackets. This causes some log
output to be confusing because it is difficult to separate the IPv6
address from a port specification.

* Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and
  pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6
  output.

* When a pj_sockaddr is not available, explicitly wrap IPv6 addresses
  in brackets.

* When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure
  to also set pjsip_rx_data.pkt_info.src_addr.

Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8

2 years agoCI: Use proper credentials for Security testsuite checkout
George Joseph [Fri, 14 Sep 2018 17:31:28 +0000 (11:31 -0600)]
CI: Use proper credentials for Security testsuite checkout

Can't do anonymous http checkout from Security-testsuite.
Need to use same credentials as the gerrit review checkout.

Change-Id: I87af68c995cb8926f5e87f9af245600d76984f05

2 years agoMerge "res_musiconhold.c: Restart MOH if previous hold just reached end-of-file"
George Joseph [Fri, 14 Sep 2018 16:11:47 +0000 (11:11 -0500)]
Merge "res_musiconhold.c: Restart MOH if previous hold just reached end-of-file"

2 years agoMerge "optional_api: Remove unused nonoptreq fields"
Jenkins2 [Thu, 13 Sep 2018 18:08:10 +0000 (13:08 -0500)]
Merge "optional_api: Remove unused nonoptreq fields"

2 years agoMerge "CI: Use .gitreview to default BRANCH_NAME."
Joshua Colp [Thu, 13 Sep 2018 14:00:15 +0000 (09:00 -0500)]
Merge "CI: Use .gitreview to default BRANCH_NAME."

2 years agoMerge "res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP"
Joshua Colp [Thu, 13 Sep 2018 12:11:40 +0000 (07:11 -0500)]
Merge "res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP"

2 years agoMerge "Build System: Resolve conflict between DESTDIR and bundled jansson."
Joshua Colp [Wed, 12 Sep 2018 22:19:24 +0000 (17:19 -0500)]
Merge "Build System: Resolve conflict between DESTDIR and bundled jansson."

2 years agoCI: Use .gitreview to default BRANCH_NAME.
Corey Farrell [Wed, 12 Sep 2018 17:39:23 +0000 (13:39 -0400)]
CI: Use .gitreview to default BRANCH_NAME.

This ensures that binary modules are avoided in the master branch even
if BRANCH_NAME is not set.

Change-Id: I79162d2063f22fa9d6b31fde4827ace2dd5bf0da

2 years agooptional_api: Remove unused nonoptreq fields
Walter Doekes [Tue, 11 Sep 2018 12:22:18 +0000 (14:22 +0200)]
optional_api: Remove unused nonoptreq fields

As they're not actively used, they only grow stale. The moduleinfo field itself
is kept in Asterisk 13/15 for ABI compatibility.

ASTERISK-28046 #close

Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc

2 years agoMerge "manager: Set AMI event "Newexten" to the EVENT_FLAG_DIALPLAN class"
Joshua Colp [Wed, 12 Sep 2018 16:01:25 +0000 (11:01 -0500)]
Merge "manager: Set AMI event "Newexten" to the EVENT_FLAG_DIALPLAN class"

2 years agomanager: Set AMI event "Newexten" to the EVENT_FLAG_DIALPLAN class
lvl [Mon, 3 Sep 2018 11:50:07 +0000 (13:50 +0200)]
manager: Set AMI event "Newexten" to the EVENT_FLAG_DIALPLAN class

The documentation already specified EVENT_FLAG_DIALPLAN for this
event, but the implementation was using EVENT_FLAG_CALL.

Using EVENT_FLAG_DIALPLAN allows AMI clients to opt out of receiving
this highly verbose event.

ASTERISK-28033

Change-Id: I45b3119f30e4dbc17b49831f2b1a4f2c1beadafe

2 years agores_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP
Sean Bright [Wed, 12 Sep 2018 12:18:07 +0000 (08:18 -0400)]
res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP

The bundled version of pjproject has a patch for Solaris compatability
that changes the definition of various socket structures which we need
to account for when compiling against a non-bundled version.

ASTERISK-28049 #close

Change-Id: Ia1ea47c433fc2d915115193ee889a752373925f0

2 years agoBuild System: Resolve conflict between DESTDIR and bundled jansson.
Corey Farrell [Tue, 11 Sep 2018 03:28:04 +0000 (23:28 -0400)]
Build System: Resolve conflict between DESTDIR and bundled jansson.

If Asterisk is built using a DESTDIR this will cause the bundled jansson
to be installed to an unexpected location and we will fail to find it.

Change-Id: Id033e2813261e0d45232383d44c6391122169548

2 years agores_musiconhold.c: Restart MOH if previous hold just reached end-of-file
Frederic LE FOLL [Thu, 30 Aug 2018 08:42:18 +0000 (10:42 +0200)]
res_musiconhold.c: Restart MOH if previous hold just reached end-of-file

On MOH activation, moh_files_readframe() is called while the current
stream attached to the channel is NULL and it calls ast_moh_files_next()
immediately.  However, it won't call ast_moh_files_next() again if sample
reading fails.  The failure may occur because res_musiconhold retains the
last sample reading position in the channel data and MOH during the
previous hold/retrieve just reached EOF.  Obviously, a bit of bad luck is
required here.

* Restructured moh_files_readframe() to try a second time to start MOH if
there was no stream setup and the saved position was at EOF.  Also added
comments describing what is going on for each step.

ASTERISK-28029

Change-Id: I1508cf2c094f8feca22d6f76deaa9fdfa9944860

2 years 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."

2 years 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"

2 years 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"

2 years 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

2 years 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

2 years 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

2 years 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."

2 years 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."

2 years 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

2 years 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"

2 years 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"

2 years 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

2 years 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

2 years 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."

2 years 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

2 years 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

2 years 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."

2 years 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"

2 years 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

2 years 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."

2 years 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

2 years 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"

2 years 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

2 years 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

2 years 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

2 years 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"

2 years 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."

2 years 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

2 years 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

2 years 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"

2 years 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

2 years 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

2 years 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."

2 years 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

2 years 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"

2 years 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."

2 years 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."

2 years 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"

2 years 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

2 years 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"

2 years 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."

2 years 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."

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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."

2 years 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."

2 years 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."

2 years 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

2 years 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

2 years 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

2 years 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"

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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

2 years 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."

2 years 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."

2 years 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"

2 years 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."

2 years 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

2 years 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

2 years 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."

2 years 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

2 years 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

2 years 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

2 years 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."

2 years 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

2 years 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."

2 years 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."

2 years 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."

2 years 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."

2 years 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