Commit Graph

30480 Commits

Author SHA1 Message Date
lvl
eda1af091e 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
2018-09-06 16:13:00 -05:00
George Joseph
60b703d9ca Merge "http.c: Give HTTP error response when received lines are too long." into 13 2018-09-06 11:49:46 -05:00
George Joseph
8936a264bb Merge "app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done" into 13 2018-09-05 10:59:45 -05:00
George Joseph
5d4cf16c17 Merge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch" into 13 2018-09-05 09:55:06 -05:00
Rodrigo Ramírez Norambuena
17040d1ce3 app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done
Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8
2018-09-04 07:51:31 -05:00
Chris-Savinovich
7b898dd8e1 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
2018-08-31 18:02:35 -04:00
Richard Mudgett
4e20fe9f06 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
2018-08-30 16:40:31 -05:00
Jenkins2
c0e47ee2be Merge "make config: os-release output error." into 13 2018-08-30 14:37:43 -05:00
Joshua Colp
d0da68bfc9 Merge "res_pjsip_transport_websocket: Properly set src_name for IPv6" into 13 2018-08-30 11:09:16 -05:00
Joshua Colp
6bf0943ecb Merge "res_fax: Handle fax gateway being started more than once." into 13 2018-08-30 05:04:29 -05:00
Richard Mudgett
f2266602a1 res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch
ASTERISK-27988

Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843
2018-08-29 09:45:22 -05:00
Rodrigo Ramírez Norambuena
00e5c7d0ff 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
2018-08-29 08:26:02 -05:00
Joshua Colp
cfe8ef917d Merge "Create --disable-binary-modules option." into 13 2018-08-29 06:09:42 -05:00
Joshua Colp
fa71f498ce 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
2018-08-29 07:18:08 -03:00
George Joseph
36a03b673a Merge "alembic: increase uri column size" into 13 2018-08-28 09:16:39 -05:00
George Joseph
7ede0ca5b3 Merge "res/res_rtp_asterisk: remove debug traces generated by an empty frame" into 13 2018-08-28 09:15:21 -05:00
Sean Bright
f3188a634d 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
2018-08-28 09:01:19 -04:00
Corey Farrell
abb04622f7 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
2018-08-27 13:51:29 -05:00
Jenkins2
37102eb449 Merge "chan_sip: improved ip:port finding of peers for non-UDP transports." into 13 2018-08-27 07:43:03 -05:00
George Joseph
8e904d22cf Merge "sample_configs: noload res_hep.so by default" into 13 2018-08-27 06:30:13 -05:00
Joshua Colp
c2cafddaf3 Merge "AMI: be less verbose when adding HTTP headers to AMI/HTTP messages." into 13 2018-08-24 10:12:29 -05:00
Jaco Kroon
2a5d408733 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 b2c4e8660a (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
2018-08-24 02:31:41 -05:00
Jaco Kroon
b528715fa0 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
2018-08-23 12:19:46 -05:00
Matthew Fredrickson
adb3195697 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
(cherry picked from commit c8bacd45f1)
2018-08-23 10:12:08 -05:00
Jenkins2
6c4a029bbe Merge "chan_sip: remove unnecessary ast_channel_unlock(peer) as RAII looks after it" into 13 2018-08-23 08:06:25 -05:00
Florian Floimair
1928e00d90 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: I08e17b37af5a446c36cc7d1f903861f81fd41cca
2018-08-23 14:06:45 +02:00
Joshua Colp
0bc1366e15 Merge "app_queue: Silence GCC 8 compiler warning" into 13 2018-08-22 11:17:20 -05:00
Sean Bright
b9d9c0a8b9 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
2018-08-22 08:52:46 -05:00
Joshua Colp
1cb87da69c Merge "pbx_dundi.c: Handle thread shutdown better." into 13 2018-08-21 18:53:06 -05:00
alecdavis
6964bc37e0 chan_sip: remove unnecessary ast_channel_unlock(peer) as RAII looks after it
Otherwise console output

        (get_refer_info): mutex 'peer' freed more times than we've locked!
        (get_refer_info): Error releasing mutex: Operation not permitted

    or
        (get_refer_info): attempted unlock mutex 'peer' without owning it!
        (__ast_read): 'peer' was locked here.
        ...dump_backtrace

        (get_refer_info): Error releasing mutex: Operation not permitted
        (__ast_read): mutex 'chan' freed more times than we've locked!

ASTERISK-28011 #close

Change-Id: I6e45f2764ba4f3273a943300f91ac9b461ac2893
2018-08-22 11:46:32 +12:00
neutrino88
49d388bfa8 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
2018-08-21 18:24:48 -05:00
Joshua Colp
ef061d1854 Merge "AMI: Remove docs for nonexistent AMI ContactStatus event headers" into 13 2018-08-21 08:07:28 -05:00
Joshua Colp
a339cc1240 Merge "pbx_dundi.c: Misc memory management fixes when destroying peers" into 13 2018-08-21 06:27:34 -05:00
Joshua Colp
d195767b07 Merge "pbx_dundi: Fix debug frame decode string." into 13 2018-08-21 05:52:26 -05:00
Richard Mudgett
9b9dee2d2c AMI: Remove docs for nonexistent AMI ContactStatus event headers
Change-Id: I5736965c64c44338f7330e85a24bb46818607f19
2018-08-20 12:31:58 -05:00
George Joseph
3146666b71 Merge "res_sorcery_realtime.c: Fix unqualified fetch warning." into 13 2018-08-20 10:56:48 -05:00
Richard Mudgett
a66fa4db24 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
2018-08-17 16:33:00 -05:00
Richard Mudgett
f3924b576a 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
2018-08-17 14:44:35 -05:00
Richard Mudgett
fbc53412f6 pbx_dundi.c: Handle thread shutdown better.
Change-Id: Id52f99bd6a948fe6dd82acc0a28b2447a224fe87
2018-08-17 14:42:48 -05:00
Richard Mudgett
ad2dfb07b0 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
2018-08-17 14:41:04 -05:00
Richard Mudgett
0a7dab8904 pbx_dundi: Update sample config documentation.
Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849
2018-08-17 14:39:39 -05:00
Joshua Colp
9ce4708da9 Merge "CI: Disable res_odbc if REF_DEBUG is requested." into 13 2018-08-17 06:16:14 -05:00
Joshua Colp
40d2d317ae Merge "res_pjsip: Resolve transport management leak at shutdown." into 13 2018-08-17 05:38:47 -05:00
George Joseph
33a1593fba CI: Final version of setting correct gerrit creds
Change-Id: I7729ecceedceb12f52bf18dae259846aa1d993b3
2018-08-16 12:28:03 -06:00
George Joseph
ab095ca4c6 CI: Yet Another stab at creds
Change-Id: I768850780d39151c5dd8e2bb1a4b24776727958e
2018-08-16 12:23:05 -06:00
George Joseph
b1e294c670 CI: Another stab at creds
Change-Id: I5b1dc3b354789e676c27a4e1fc42f5c3343cc994
2018-08-16 12:11:11 -06:00
George Joseph
45d096fe8d 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
2018-08-16 11:08:21 -06:00
George Joseph
0c4d49c5a9 Merge "res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered" into 13 2018-08-16 09:46:00 -05:00
Corey Farrell
db367ddbbf res_pjsip: Resolve transport management leak at shutdown.
Cleanup idle check scheduled events at shutdown.

Change-Id: I61bfbb56bac69fe840c3242927d31ff3593be461
2018-08-15 13:55:55 -05:00
Corey Farrell
7f4420a222 CI: Disable res_odbc if REF_DEBUG is requested.
This is for 13/15 only, res_odbc never unloads so it is impossible to do
REF_DEBUG testing with that module.

Change-Id: I2c1e32b80564e8fb08b6b5900ede6b5d304ebd10
2018-08-15 11:46:46 -05:00