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A patch I did back in 2014 modified ast_config_text_file_save2 to check the
writability of the main file and include files before truncating and re-writing
them. An unintended side-effect of this was that if a file doesn't exist,
the check fails and the write is aborted.
This patch causes ast_config_text_file_save2 to check the writability of the
parent directory of missing files instead of checking the file itself. This
allows missing files to be created again. A unit test was also added to
test_config to test saving of config files.
The regression was discovered when app_voicemail's passwordlocation=spooldir
feature stopped working.
ASTERISK-25917 #close
Reported-by: Jonathan Rose
Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80
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The run_agi function is eating control frames when it shouldn't be. This is
causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond
transfer.
Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie
answers.
Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE
and is left thinking he's connected to Bob.
In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls
an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on
Charlie's channel.
The fix was to accumulate deferrable frames in the "forever" loop instead of
dropping them, and re-queue them just before running the actual agi command
or exiting.
ASTERISK-25951 #close
Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645
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res_odbc was changed in Asterisk 13.8.0 to remove connection management,
opting instead to let unixodbc maintain open connections and return
those to Asterisk as requested.
This was a boon for realtime, since it meant that multiple threads could
potentially run parallel queries since they could each be using their
own database connections.
However, on the user-facing side, func_odbc, there were some inherent
behaviors being relied on that no longer hold true after the change.
One such reported behavior was that MySQL's LAST_INSERTED_ID() works
per-connection. This means that if Asterisk uses separate connections
for every database operation, whereas before it used one connection for
everything, we have broken expectations and functionality.
The fix provided in this patch is to make func_odbc use a single
database connection per DSN. This way, user-facing database usage will
have the same behavior as it did pre-13.8.0. However, realtime, which is
the real workhorse of database interaction, will continue to let
unixodbc manage connections.
ASTERISK-25938 #close
Reported by Edwin Vandamme
Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc
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Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f
ASTERISK-25927 #close
Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8
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*lt is NULL if t->tracking == 0
ASTERISK-25948 #close
Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba
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We lose the fact that there is a swap channel if there is one. We
currently wind up rejoining the stasis bridge as a normal join after the
swap channel has already been kicked from the bridge.
This patch preserves the swap channel so the AMI/ARI events can note that
the channel joining the bridge is swapping with another channel. Another
benefit to swaqpping in one operation is if there are any channels that
get lonely (MOH, bridge playback, and bridge record channels). The lonely
channels won't leave before the joining channel has a chance to come back
in under stasis if the swap channel is the only reason the lonely channels
are staying in the bridge.
ASTERISK-25947 #close
Reported by: Richard Mudgett
ASTERISK-24649
Reported by: John Bigelow
ASTERISK-24782
Reported by: John Bigelow
Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee
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An earlier patch blocked the ast_bridge_impart() call until the channel
either entered the target bridge or it failed. Unfortuantely, if the
target bridge is stasis and the imprted channel is not a stasis channel,
stasis bounces the channel out of the bridge to come back into the bridge
as a proper stasis channel. When the channel is bounced out, that
released the block on ast_bridge_impart() to continue. If the impart was
a result of a transfer, then it became a race to see if the swap channel
would get hung up before the imparted channel could come back into the
stasis bridge. If the imparted channel won then everything is fine. If
the swap channel gets hung up first then the transfer will fail because
the swap channel is leaving the bridge.
* Allow a chain of ast_bridge_impart()'s to happen before any are
unblocked to prevent the race condition described above. When the channel
finally joins the bridge or completely fails to join the bridge then the
ast_bridge_impart() instances are unblocked.
ASTERISK-25947
Reported by: Richard Mudgett
ASTERISK-24649
Reported by: John Bigelow
ASTERISK-24782
Reported by: John Bigelow
Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1
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When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning
the From header, then it overwrites the display name and uri from the channel's
connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was
leaving the display name from the From header in the new RPID or PAI header.
On an attended transfer where the originator had a caller id number set but not
a display name, the re-INVITE to the final transferee had the number of the
originator but the display name of the transferer.
Added a check to clear out the display name in the new header if
connected.id.name was invalid.
ASTERISK-25942 #close
Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b
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This module is used as part of testsuite tests to confirm
stuff works. I'm accordingly marking it as core as it is
required by those tests.
Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88
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The PJSIP parsing functions provide a nice concise way to check the
length of a hostname in a SIP URI. The problem is that in order to use
those parsing functions, it's required to use them from a thread that
has registered with PJLib.
On startup, when parsing AOR configuration, the permanent URI handler
may not be run from a PJLib-registered thread. Specifically, this could
happen when Asterisk was started in daemon mode rather than
console-mode. If PJProject were compiled with assertions enabled, then
this would cause Asterisk to crash on startup.
The solution presented here is to do our own parsing of the contact URI
in order to ensure that the hostname in the URI is not too long. The
parsing does not attempt to perform a full SIP URI parse/validation,
since the hostname in the URI is what is important.
ASTERISK-25928 #close
Reported by Joshua Colp
Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60
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Recent changes to the PJSIP registrar resulted in tests failing due to
missing AOR_CONTACT_ADDED test events. The reason for this was that the
user_agent string had junk values in it, resulting in being unable to
generate the event.
I'm going to be honest here, I have no idea why this was happening. Here
are the steps needed for the user_agent variable to get messed up:
* REGISTER is received
* First contact in the REGISTER results in a contact being removed
* Second contact in the REGISTER results in a contact being added
* The contact, AOR, expiration, and user agent all have to be passed as
format parameters to the creation of a string. Any subset of those
parameters would not be enough to cause the problem.
Looking into what was happening, the thing that struck me as odd was
that the user_agent variable was meant to be set to the value of the
User-Agent SIP header in the incoming REGISTER. However, when removing a
contact, the user_agent variable would be set (via ast_strdupa inside a
loop) to the stored contact's user_agent. This means that the
user_agent's value would be incorrect when attempting to process further
contacts in the incoming REGISTER.
The fix here is to use a different variable for the stored user agent
when removing a contact. Correcting the behavior to be correct also
means the memory usage is less weird, and the issue no longer occurs.
ASTERISK-25929 #close
Reported by Joshua Colp
Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08
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At shutdown it is possible for modules to be unloaded that wouldn't
normally be unloaded. This allows the environment to be cleaned up.
The res_pjsip_transport_management module did not have the unload
logic in it to clean itself up causing the res_pjsip module to not
get unloaded. As a result the res_pjsip monitor thread kept going
processing traffic and timers when it shouldn't.
Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a
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We have to setup the channel roles after the bridge class push is called
because the bridge class push callback may have set roles on the incoming
channel. Since we have already partially pushed the channel into the
bridge and reversing what we have already done could be problematic, the
only thing we can do is press on to complete pushing the channel into the
bridge.
* Ignore any channel role setup errors after pushing the channel into a
bridge. The channel may behave incorrectly in the bridge but we can no
longer abort the push at this time.
Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00
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If rtupdate=no do not verify sipregs/peers table has updatable fields.
ASTERISK-25934 #close
Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d
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ASTERISK-25888 #close
Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117
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Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a
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There is a good amount of repetition in the two frame handling routines
in the Dial API. This commit combines the two functions into one.
This is in preparation for an upcoming commit that adds the ability to
handle frames for a channel in a bridge.
ASTERISK-25925
Reported by Mark Michelson
Change-Id: Iaae2f174e3058e774cb44e10659fcdfb85345c58
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Failed registration using PJSIP/Realtime if one of the codec name
in allow/disallow option is wrong or contains space.
This patch strip codec name.
ASTERISK-25914
Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d
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The scheduler thread that kills idle TCP connections was not registering
with PJProject properly and causing assertions if PJProject was built in
debug mode.
This change registers the thread with PJProject the first time that the
scheduler callback executes.
AST-2016-005
Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283
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"Idle" here means that someone connects to us and does not send a SIP
request. PJProject will not automatically time out such connections, so
it's up to Asterisk to do it instead.
When we receive an incoming TCP connection, we will start a timer
(equivalent to transaction timer D) waiting to receive an incoming
request. If we do not receive a request in that timeframe, then we will
shut down the TCP connection.
ASTERISK-25796 #close
Reported by George Joseph
AST-2016-005
Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6
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ASTERISK-25796
Reported by George Joseph
AST-2016-005
Change-Id: Id322a05f927392293570599730050bc677d99433
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Due to some ignored return values, Asterisk could crash if processing an
incoming REGISTER whose contact URI was above a certain length.
ASTERISK-25707 #close
Reported by George Joseph
Patches:
0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch
AST-2016-004
Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d
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Fix off nominal crash where we could not setup the channel to process
frames for the softmix bridge technology because of allocation failure.
Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372
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In 13, the new ast_string_field_header structure had to be dynamically
allocated and assigned to a pointer in ast_string_field_mgr to preserve ABI
compatability. In master, it can be converted to being a structure-in-place in
ast_string_field_mgr to eliminate the extra alloc and free calls.
Change-Id: Ia97c5345eec68717a15dc16fe2e6746ff2a926f4
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From the patch submitted to Teluu on 4/12/2016
<<<<<<<<<
The wholesale stripping of '[]' from header parameters causes issues if
something (like a port) occurs after the final ']'.
'[2001:a::b]' will correctly parse to '2001:a::b'
'[2001:a::b]:8080' will correctly parse to '2001:a::b' but the scanner is left
with ':8080' and parsing stops with a syntax error.
I can't even find a case where stripping the '[]' is a good thing anyway. Even
if you continued to parse and resulted in a string that looks like this...
'2001:a::b:8080', it's not valid.
This came up in Asterisk because Kamailio sends us a Contact with an alias
URI parameter that has an IPv6 address in it like this:
Contact: <sip:1171@127.0.0.1:5080;alias=[2001:1:2::3]~43691~6>
which should be legal but causes a syntax error because of the characters
after the final ']'. Even if it didn't, the '[]' should still not be stripped.
I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6
enabled. No issues were caused by removing the code that strips the '[]'.
>>>>>>>>>>>
ASTERISK-25123 #close
Reported-by: Anthony Messina
Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a
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The test_voicemail_notify_endl test checks the end-of-line
characters of an email message to confirm that they are consistent.
The test wrongfully assumed that reading from the email message
into a buffer will always result in more than 1 character being
read. This is incorrect. If only 1 character was read the test
would go outside of the buffer and access other memory causing
a crash.
The test now checks to ensure that 2 or more characters are read
in ensuring the test stays within the buffer.
ASTERISK-25874 #close
Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710
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If try to move message to Cust1 (number 5)
the function 'save_to_folder' tries to create Greeting folder instead of Cust1.
This patch fixed it by setting GREETINGS_FOLDER = -1
ASTERISK-24927 #close
Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51
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* Added Useragent and RegExpire headers to AMI Event
ContactStatusDetail with associated documentation.
ASTERISK-25903 #close
Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239
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