Russell Bryant [Wed, 21 Jul 2010 13:02:46 +0000 (13:02 +0000)]
Update documentation for 'comebacktoorigin' in featuers.conf.
The documentation for this option did not match the code. Fix that along with
some minor cleanups to the code along the way. Document a slight change in
behavior (to something that was previously undocumented) in UPGRADE.txt.
David Vossel [Tue, 20 Jul 2010 21:41:21 +0000 (21:41 +0000)]
fixes sip CANCEL race condition
If Asterisk sends a 4xx error and the other side sends a CANCEl
before receiving the 4xx and responding with the ACK, Asterisk
will process the CANCEL and send a 487 Request Terminated as
a new final response to the INVITE. Since we are issuing a new
final response to the INVITE, the old one must be pretend_acked
else it will keep retransmitting.
This commit contains several changes to the way output channel variables are handled.
FAX output channel variables will now match the values reported by FAXOPT() and should be set in all failure and success cases.
This commit also contains a few modifications to the way FAXOPT() variables are populated in a few spots and fixes for some reference count leaks of the session details structure in some failure cases.
Also found and fixed more cases where FAXOPT(status) may not have gotten set.
Avoid trying to pickup a parked extension before the park operation is completed.
A crash could occur if the extension is picked up while the parking extension is
being announced. Testing pu->notquiteyet while searching for a parked extension
resolves this crash.
Mark Michelson [Mon, 19 Jul 2010 17:16:23 +0000 (17:16 +0000)]
Fix port setting of external address in SIP.
There are two changes here:
1. Since the externip setting can now have a port attached
to it, calling it "externip" is misleading. The option is now
documented and parsed as "externaddr." This also extends to the
"matchexterniplocally" setting. It is now documented and parsed
as "matchexternaddrlocally." The old names for the options may
still be used, but they are no longer used in the sip.conf.sample
file.
2. If no port is set for the externaddr, and UDP is the transport
to be used, then we will set the port of the externaddr to that of
the udpbindaddr. This was how things worked prior to the IPv6 merge,
so this is a regression fix.
Mark Michelson [Mon, 19 Jul 2010 17:10:00 +0000 (17:10 +0000)]
Remove the fe80:1234::1234 test case from test_acl.c
The ACL test was failing on Mac OS X because it would
convert the above invalid link-local address into
fe80::1234 while reporting no error from getaddrinfo().
Linux does not do this.
Jeff Peeler [Mon, 19 Jul 2010 14:39:07 +0000 (14:39 +0000)]
Fix regression with distinctive ring detection.
The issue here is that passing an array to a function prohibits the ARRAY_LEN
macro from returning the real size. To avoid this the size is now defined and
use of ARRAY_LEN is avoided.
Mark Michelson [Mon, 19 Jul 2010 14:17:16 +0000 (14:17 +0000)]
Make ACLs IPv6-capable.
ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer.
ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended
transfer, ast_bridge_call() is called for a second bridge on the same channel,
and it clears that flag, which still needs to get set for when the original
ast_bridge_call() gets control back and checks it.
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
priexclusive in chan_dahdi.conf ignored when reloading dahdi module
During a reload, the priexclusive and outsignalling parameters are not
read in from the config file as intended. Unfortunately, they get set to
defaults as a result. This patch makes sure that they do not get set to
defaults during a reload.
(closes issue #17441)
Reported by: mtryfoss
Patches:
issue17441_v1.4.patch uploaded by rmudgett (license 664)
issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
issue17441_trunk.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
........
For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation.
Olle Johansson [Fri, 16 Jul 2010 10:00:58 +0000 (10:00 +0000)]
Add ability to configure the Max-Forwards header in the dialplan, as well as in
sip.conf configuration for the channel and for devices.
The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.
Jeff Peeler [Thu, 15 Jul 2010 20:21:03 +0000 (20:21 +0000)]
Correct not setting the bindport before attempting to open the socket.
Related to changes from 276571, I was accidentally testing with a port set in
my configuration causing me to miss this. Also moved the TCP handling as well
to occur before build_peer is called.
Jeff Peeler [Wed, 14 Jul 2010 22:58:24 +0000 (22:58 +0000)]
Fix MWI notification transmission problems over SIP.
MWI updates were not being sent if no messages were found in the event cache.
This was corrected since a phone may need to clear its MWI status configured
previously from another mailbox.
Upon module or sip reload, MWI updates could not be sent due to the sipsock
socket not being set early enough in reload_config. The code handling the
descriptor assignment and such has simply been moved before the call to
build_peer.
Issuing a sip reload cleared the IP address of the peer, but skipped checking
the database for registration information. The database is now checked both
for sip reload and actually reloading the module.
If a transmission occurs before the do_monitor thread has started, do not
attempt to send a signal to it.
Mark Michelson [Wed, 14 Jul 2010 22:32:29 +0000 (22:32 +0000)]
Fix errors where incorrect address information was printed.
ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions)
uses thread-local storage for storing the string that it creates. In cases where
ast_sockaddr_stringify_fmt was being called twice within the same statement, the
result of one call would be overwritten by the result of the other call. This
usually was happening in printf-like statements and was resulting in the same
stringified addressed being printed twice instead of two separate addresses.
I have fixed this by using ast_strdupa on the result of stringify functions if
they are used twice within the same statement. As far as I could tell, there were
no instances where a pointer to the result of such a call were saved anywhere, so
this is the only situation I could see where this error could occur.
David Vossel [Wed, 14 Jul 2010 19:51:08 +0000 (19:51 +0000)]
handle special case were "200 Ok" to pending INVITE never receives ACK
Unlike most responses, the 200 Ok to a pending INVITE Request is
acknowledged by an ACK Request. If the ACK Request for this Response is not received
the previous behavior was to immediately destroy the dialog and hangup
the channel. Now in an effort to be more RFC compliant, instead of immediately
destroying the dialog during this special case, termination is done with a BYE Request
as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is
never received. The behavior of immediately hanging up the channel remains.
This only affects how dialog termination proceeds for this one special case.
RFC 3261 section 13.3.1.4
"If the server retransmits the 2xx response for 64*T1 seconds without receiving
an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is
accomplished with a BYE, as described in Section 15."
Tim Ringenbach [Wed, 14 Jul 2010 16:09:11 +0000 (16:09 +0000)]
Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.
Change the documented pgsql schema to use "timestamp" instead of "time",
as the latter is only a time without a date.
Added some missing columns for cel's pgsql schema, and corrected spelling
on some others. Updated cel's uniqueid size to be the same as the cdr.
Added id column to cel's pgsql schema and updated code to allow unknown
columns to get their default value instead of forcing 0 or empty string.
Added microseconds to the timestamp cel logs to pgsql.
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
David Vossel [Tue, 13 Jul 2010 22:18:38 +0000 (22:18 +0000)]
chan_sip: RFC compliant retransmission timeout
Retransmission of packets should not be based on how many packets were
sent, but instead on a timeout period. Depending on whether or not the
packet is for a INVITE or NON-INVITE transaction, the number of packets
sent during the retransmission timeout period will be different, so
timing out based on the number of packets sent is not accurate.
This patch fixes this by removing the retransmit limit and only stopping
retransmission after a timeout period is reached. By default this
timeout period is 64*(Timer T1) for both INVITE and non-INVITE
transactions. For more information on sip timer values refer to
RFC3261 Appendix A.
Make user removals and traversals thread safe in meetme.
Race conditions present in meetme involving the user list where a lack of
locking has the potential for a user to be removed during a traversal or as in
the case of the reporter after checking if the list is empty could cause a
crash. Fixing this was done by convering the userlist to an ao2 container.
Terry Wilson [Tue, 13 Jul 2010 17:11:37 +0000 (17:11 +0000)]
Destroy RTP fds when we schedule final dialog destruction
Since we are only keeping the dialog around for retransmissions at this point
and there is no possibility that we are still handling RTP, go ahead and
destroy the RTP sessions. Keeping them alive for 32 past when they are used
is unnecessary and can lead to problems with having too many open file
descriptors, etc.
Access peer->cdr directly instead of through a saved off reference.
At this point in the code, it is possible that peer_cdr may be invalid.
Specifically, in the blind transfer code, CDRs are swapped between channels.
So, peer_cdr is no longer == peer->cdr.
The scenario that exposed a crash in this code was a blind transfer that hit
the system call limit, causing the transferee channel to get destroyed after
the transfer attempt failed. Even if it succeeds and this code doesn't crash,
this code was still trying to reset a CDR on a channel that was now owned by
a different thread, which is a BadThing(tm).
Change ast_write to not stop generator when called from ast_prod.
For SIP channels configured with the progressinband option on, the ringback was
being immediately stopped. This problem was due to ast_prod being moved for a
deadlock fix in 259858. Prodding the channel after setting up the generator
triggered the check in ast_write to stop the generator. The fix here should
write the frame the same as was done before the call to ast_prod was moved.
This change adds an ERROR message to let you know when a failure exists to
get the columns from the pgsql database, which typically means that the
table does not exist.
When creating a conference for a unit test, it is not mandatory to open a
dahdi pseudo channel, so if we fail doing it, continue creating the conference.
Russell Bryant [Fri, 9 Jul 2010 21:57:21 +0000 (21:57 +0000)]
Fix some issues related to dynamic feature groups in features.conf.
The bridge handling code did not properly consider feature groups when setting
parameters that would affect whether or not a native bridge would be attempted.
If DYNAMIC_FEATURES only include a feature group, a native bridge would occur
that may prevent features from working.
Fix a bug in verbose output that would show the key mapping as empty if it was
using the default mapping and not a custom mapping in the feature group.
Add feature groups to the output of "features show".
Adjust the feature execution logic to match that of the logic when executing
a feature that was not configured through a feature group.
Update features.conf.sample to show that an '=' is still required if using
the default key mapping from [applicationmap].
Finally, clean up a little bit of formatting to better coform to coding
guidelines while in the area.
(closes issue #17589)
Reported by: lmadsen
Patches:
issue_17589.rev4.txt uploaded by russell (license 2)
Tested by: russell, lmadsen
Mark Michelson [Fri, 9 Jul 2010 20:58:52 +0000 (20:58 +0000)]
Fix error in parsing SIP registry strings from ASTdb.
It was essentially an off-by-one error. The easiest way
to fix this was to use the handy-dandy AST_NONSTANDARD_RAW_ARGS
macro to parse the pieces of the registration string out. Tested
and it works wonderfully.
projects / thirdparty / asterisk.git / log
