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r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
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r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines
Merged revisions 225243 via svnmerge from
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r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines
IAX2: VNAK loop caused by signaling frames with no destination call number
It is possible for the PBX thread to queue up signaling frames before
a destination call number is received. This can result in signaling
frames being sent out with no destination call number. Since recent
versions of Asterisk require accurate destination callnumbers for all
Full Frames, this can cause a VNAK loop to occur. To resolve this
no signaling frames are sent until a destination callnumber is received,
and destination call numbers are now only required for iax_pvt matching
when the frame is an ACK.
Review: https://reviewboard.asterisk.org/r/413/
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r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines
Merged revisions 225032 via svnmerge from
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r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines
IAX/SIP shrinkcallerid option
The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
and '-' from the string. This means values such as 555.5555 and
test-test result in 555555 and testtest. There are instances,
such as Skype integration, where a specific value is passed via
caller id that must be preserved unmodified. This patch makes
the shrinking of caller id optional in chan_sip and chan_iax in
order to support such cases. By default this option is on to
preserve previous expected behavior.
(closes issue #15940)
Reported by: dimas
Patches:
v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/408/
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r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines
Merged revisions 224330 via svnmerge from
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r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines
Fix stale caller id data from being reported in AMI NewChannel event
The problem here is that chan_dahdi is designed in such a way to set
certain values in the dahdi_pvt only once. One of those such values
is the configured caller id data in chan_dahdi.conf. For PRI, the
configured caller id data could be overwritten during a call. Instead
of saving the data and restoring, it was decided that for all non-analog
channels it was simply best to not set the configured caller id in the
first place and also clear it at the end of the call.
(closes issue #15883)
Reported by: jsmith
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r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines
Merged revisions 224260 via svnmerge from
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r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines
Never released PRI channels when using Busy() or Congestion() dialplan apps.
When the Busy() or Congestion() application is used towards ISDN (an ISDN
progress is sent), the responding ISDN Disconnect or Release may contain
the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c
these causes will only set the needbusy or needcongestion flags and not
activate the softhangup procedure. Unfortunately only the latter can
interrupt the endless wait loop of Busy()/Congestion().
Result: PRI channels staying in state busy for the rest of asterisk life
or until the other end times out and forces the call to clear.
(in issue 0014292)
Reported by: tomaso
Patches:
disc_rel_userbusy.patch uploaded by tomaso (license 564)
(This patch is unrelated to the issue.)
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r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines
Remove automatic switching from T.38 to voice mode in chan_sip.
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and changes app_fax to
explicitly switch off T.38 mode when the FAX transmission process ends.
(closes issue #16025)
Reported by: jamicque
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This commit is the simplest way to solve a problem that has already been solved
in trunk with the "COLP/CONP and Redirecting party information into Asterisk"
commit. In trunk the redirection reason is translated into a generic redirect
reason. I would have had to do the same fix except chan_sip never reads
PRIREDIRECTREASON. So both chan_dahdi and chan_h323 have been modified to
interpret the one different redirect reason of "no-answer" properly and set the
ISDN reason code 2 of "no reply".
(closes issue #15033)
Reported by: steinwej
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r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
p->peerauth is always empty in transmit_register()
When using callbackextension or specifing the peer name
in a registration string, the peer's specific auth settings
set by the "auth=" strings within the peer definition are not
used by the registration. Thanks to ebroad for reporting the
issue and providing the patch.
(closes issue #15955)
Reported by: ebroad
Patches:
regauthfix.patch uploaded by ebroad (license 878)
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r222799 | rmudgett | 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) | 19 lines
Merged revisions 222797 via svnmerge from
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r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines
Fix memory leak if chan_misdn config parameter is repeated.
Memory leak when the same config option is set more than once in an
misdn.conf section. Why must this be considered? Templates! Defining a
template with default port options and later adding to or overriding some
of them.
Patches:
memleak-misdn.patch
JIRA ABE-1998
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r222692 | rmudgett | 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) | 21 lines
Merged revisions 222691 via svnmerge from
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r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines
chan_misdn.c:process_ast_dsp() memory leak
misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not
occur.
The translated frame "f2" when passing through ast_dsp_process() is not
freed whenever it is not used further in process_ast_dsp(). Then in the
end it is never ever freed.
Patches:
translate.patch
JIRA ABE-1993
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The variable index used in this scenario for accessing the dahdi_pvts was
wrong and was most likely copied from the several other places it is used
correctly.
(closes issue #15998)
Reported by: tsearle
Patches:
dahdi_reset_crash.patch uploaded by tsearle (license 373)
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r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) | 9 lines
Fix 222298 (crash during destruction of second channel when variable set with
setvar).
I mistakenly reasoned that setvar would be used on all channels. Since it can
be set per channel, give each dahdi channel a copy of the variable.
(related to #15899)
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r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) | 9 lines
Fix crash during destruction of second channel when variable set with setvar.
The setvar line in chan_dahdi.conf is shared among all the channels, so make
sure to only free the resources only when the last channel is destroyed.
(closes issue #15899)
Reported by: tzafrir
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r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines
Recorded merge of revisions 222152 via svnmerge from
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
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r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines
Allow non-compliant T.38 endpoints to be supportable via configuration option.
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
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r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines
Merged revisions 221769 via svnmerge from
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r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines
Occasionally losing use of B channels in chan_misdn.
I have not been able to reproduce the problem of losing channels.
However, I have seen in the code a reentrancy problem that might give
these symptoms.
The reentrancy patch does several things:
1) Guards B channel and B channel structure allocation.
2) Makes the B channel structure find routines more precise in locating records.
3) Never leave a B channel allocated if we received cause 44.
The last item may cause temporary outgoing call problems, but they should
clear when the line becomes idle.
(closes issue #15490)
Reported by: slutec18
Patches:
issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: rmudgett, slutec18
(closes issue #15458)
Reported by: FabienToune
Patches:
issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: FabienToune, rmudgett, slutec18
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r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines
Merged revisions 221360 via svnmerge from
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r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
Fix SRV lookup and Request-URI generation in chan_sip.
This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur.
(closes issue #14418)
Reported by: klaus3000
Tested by: klaus3000, mnicholson
Review: https://reviewboard.asterisk.org/r/369/
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r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
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r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
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r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines
Merged revisions 220873 via svnmerge from
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r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines
Reduce CPU usage related to building a peer merely for devicestates.
This fixes a 100% CPU problem in the SIP driver, found by profiling
the driver while the problem was occurring.
(closes issue #14309)
Reported by: pkempgen
Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban
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r220718 | jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
Fix building of registration entry in build_peer when using callbackextension
Check for remotesecret option was unintentionally always true, which therefore
caused the secret option to never be used. Thanks to dvossel for pointing out
the exact fix.
(closes issue #15943)
Reported by: tpsast
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r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines
Merged revisions 219519 via svnmerge from
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r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
iax2 frame double free
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
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r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines
Merged revisions 219450 via svnmerge from
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r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
(closes issue #15262)
Reported by: maniax
Patches:
asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
invite_new_branch_trunk.diff uploaded by dvossel (license 671)
Tested by: maniax, dvossel
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r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines
Merged revisions 219303 via svnmerge from
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r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines
INVITE w/Replaces deadlock fix
This patch cleans up the locking logic in chan_sip.c's
handle_invite_replaces() function as well as making use
of ast_do_masquerade() rather than forcing the masquerade
on an ast_read(). The code had several redundant unlocks
that would result in 'freed more times than we've locked!'
errors. I cleaned these up as well as moving all the unlock
logic to the end of the function. This patch should also
resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls.
(closes issue #15151)
Reported by: irroot
Patches:
invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/
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r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines
On TCP and TLS connections do not attempt to stop retransmission of the packet internally.
This was preventing responses from being properly processed because the packet was not being found
causing handle_response to return prematurely.
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r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) | 18 lines
Merged revisions 218401 via svnmerge from
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r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines
Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor.
After talking to rmudgett about some of his recent iflist locking changes, it
was determined that the only place that would destroy a channel without being
explicitly to do so was in handle_init_event. The loop to walk the interface
list has been modified to wait to destroy the channel until the dahdi_pvt of
the channel to be destroyed is no longer needed.
(closes issue #15378)
Reported by: samy
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r217807 | dvossel | 2009-09-10 16:07:47 -0500 (Thu, 10 Sep 2009) | 28 lines
Merged revisions 217806 via svnmerge from
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r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines
IAX2 encryption regression
The IAX2 Call Token security patch inadvertently broke the use of
encryption due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must first
be decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this, decryption
of full frames is once again done before looking into the frame. This
involves searching for an existing callno, checking the pvt to see if
encryption is turned on, and decrypting the packet before the internal
fields of the full frame are accessed.
(closes issue #15834)
Reported by: karesmakro
Patches:
iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, karesmakro
Review: https://reviewboard.asterisk.org/r/355/
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r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
caller id number empty
parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.
(closes issue #15839)
Reported by: ebroad
Patches:
blank_cidv2.patch uploaded by ebroad (license 878)
parse_uri_fix.diff uploaded by dvossel (license 671)
Tested by: ebroad, dvossel
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r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines
Merged revisions 216430 via svnmerge from
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r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
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r216368 | russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines
Do not treat every SIP peer as if they were configured with insecure=port.
There was a problem in the function responsible for doing peer matching by
IP address and port number such that during the second pass for checking for
a peer configured with insecure=port, it would end up treating every peer as
if it had been configured that way. These changes fix the logic in the peer
IP and port comparison callback to handle insecure=port checking properly.
This problem was introduced when SIP peers were converted to astobj2. Many
thanks to dvossel for noticing this while working on another peer matching
issue.
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r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) | 25 lines
Merged revisions 215682 via svnmerge from
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r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines
Re-send non-100 provisional responses to prevent cancellation
From section 13.3.1.1 of RFC 3261:
If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction. A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction. To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.
(closes issue #11157)
Reported by: rjain
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/315/
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r215681 | dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines
port string to int conversion using sscanf
There are several instances where a port is parsed
from a uri or some other source and converted to
an int value using atoi(), if for some reason the
port string is empty, then a standard port is used.
This logic is used over and over, so I created a function
to handle it in a safer way using sscanf().
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r215665 | mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
add Parkinglot info to sip show peer <foo> and skinny show line <foo>
If we had this from the start, debugging the 'parking not using configured parkinglot'
bug would have been easier.
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r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) | 3 lines
like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel.
This makes callparking honor the configured parkinglot for skinny lines as well.
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r215466 | dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines
SIP support for keep-alive event
keep-alive events are used by Sipura/Linksys for NAT keepalive.
There currently don't appear to be any problems with NAT, but
everytime a keep-alive event is received, Asterisk responds with a
"489 Bad event". This error may indicate to a user that NAT
problems exist just because this even is not supported. Now,
rather than respond with an error, the packet is consumed and
a "200 ok" is sent just to indicate we received the packet.
(issue #15084)
Patches:
chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
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r215462 | mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 lines
Honor configured parkinglot when parking and retrieving parked calls
Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer
into the newly created channel.
(closes issue #15538)
Reported by: gracedman
Patches:
2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7)
With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call)
Tested by: gracedman, mvanbaak
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r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines
Better parsing for the "register" line
Allows characters that are otherwise used as delimiters to be used within
certain fields (like the secret).
(closes issue #15008, closes issue #15672)
Reported by: tilghman
Patches:
20090818__issue15008.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
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r212506 | jpeeler | 2009-08-17 11:50:45 -0500 (Mon, 17 Aug 2009) | 19 lines
Merged revisions 212498 via svnmerge from
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r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) | 12 lines
Fix segfault when reloading chan_misdn.
If more ports were specified than configured in misdn.conf a reload would crash
asterisk. The problem was the unconfigured port was using data from the
previously configured port. When the data for an unconfigured port was freed a
crash would result from the double free.
(closes issue #12113)
Reported by: agupta
Patches:
bug12113.patch uploaded by jpeeler (license 325)
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r211876 | mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11 lines
Make asterisk handle 423 Interval Too Short messages better.
This change uses separate values for the acceptable minimum expiry provided by the 423 error and the expiry value stored in the configuration file. Previously, the value pulled from the configuration file would be overwritten.
(closes issue #14366)
Reported by: Nick_Lewis
Patches:
sip-expiry-fix1.diff uploaded by mnicholson (license 96)
chan_sip.c-reqexpiry.patch uploaded by Nick (license 657)
Tested by: mnicholson
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r210817 | file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
Accept additional T.38 reinvites after an initial one has been handled.
Discussion of this subject has yielded that it is not actually acceptable to change
T.38 parameters after the initial reinvite but declining is harsh and can cause the
fax to fail when it may be possible to allow it to continue. This patch changes things
so that additional T.38 reinvites are accepted but parameter changes ignored. This gives
the fax a fighting chance.
(closes issue #15610)
Reported by: huangtx2009
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r210640 | rmudgett | 2009-08-05 14:40:03 -0500 (Wed, 05 Aug 2009) | 21 lines
Merged revisions 210575 via svnmerge from
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r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines
Dialplan starts execution before the channel setup is complete.
* Issue 15655: For the case where dialing is complete for an incoming
call, dahdi_new() was asked to start the PBX and then the code set more
channel variables. If the dialplan hungup before these channel variables
got set, asterisk would likely crash.
* Fixed potential for overlap incoming call to erroneously set channel
variables as global dialplan variables if the ast_channel structure failed
to get allocated.
* Added missing set of CALLINGSUBADDR in the dialing is complete case.
(closes issue #15655)
Reported by: alecdavis
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r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines
Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
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r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul 2009) | 13 lines
Merged revisions 209759 via svnmerge from
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r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines
Minor changes inspired by testing with latest GCC.
The latest GCC (what will become 4.5.x) has a few new warnings, that in these
cases found some either downright buggy code, or at least seriously poorly
designed code that could be improved.
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r209761 | kpfleming | 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line
Revert accidental Makefile change.
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r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines
Merged revisions 208746 via svnmerge from
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r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines
Fix compiling under dev-mode with gcc 4.4.0.
Mostly trivial changes, but I did not know of any other way to fix the
"dereferencing type-punned pointer will break strict-aliasing rules" error
without creating a tmp variable in chan_skinny.
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r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines
Merged revisions 208587 via svnmerge from
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r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines
Only send a BYE when hanging up a channel that is up.
For cases where Asterisk sends an INVITE and receives a non 2XX final
response, Asterisk would follow the INVITE transaction by immediately
sending a BYE, which was unnecessary.
(closes issue #14575)
Reported by: chris-mac
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r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines
Resolve a T.38 negotiation issue left over from the udptl-updates merge.
The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.
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r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
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r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) | 24 lines
Merged revisions 208386 via svnmerge from
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r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines
Fix a problem where a 491 response could be sent out of dialog.
This generalizes the fix for issue 13849. The initial fix corrected the
problem that Asterisk would reply with a 491 if a reinvite were received
from an endpoint and we had not yet received an ACK from that endpoint
for the initial INVITE it had sent us. This expansion also allows Asterisk
to appropriately handle an INVITE with authorization credentials if Asterisk
had not received an ACK from the previous transaction in which Asterisk had
responded to an unauthorized INVITE with a 407.
(closes issue #14239)
Reported by: klaus3000
Patches:
14239.patch uploaded by mmichelson (license 60)
Tested by: klaus3000
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r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines
Merged revisions 207647 via svnmerge from
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r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
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This patch adds a new dahdi_wait function to specifically wait for the wink
event. If the wink is not eventually received the channel is hung up.
(closes issue #14434)
Reported by: araasch
Patches:
emwinkmod uploaded by araasch (license 693)
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r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines
Merged revisions 207423 via svnmerge from
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r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines
Answer video SDP offers properly when videosupport is not enabled.
Copied from Review board:
In issue 12434, the reporter describes a situation in which audio and video
is offered on the call, but because videosupport is disabled in sip.conf,
Asterisk gives no response at all to the video offer. According to RFC 3264,
all media offers should have a corresponding answer. For offers we do not
intend to actually reply to with meaningful values, we should still reply
with the port for the media stream set to 0.
In this patch, we take note of what types of media have been offered and
save the information on the sip_pvt. The SDP in the response will take into
account whether media was offered. If we are not otherwise going to answer
a media offer, we will insert an appropriate m= line with the port set to 0.
It is important to note that this patch is pretty much a bandage being
applied to a broken bone. The patch *only* helps for situations where video
is offered but videosupport is disabled and when udptl_pt is disabled but
T.38 is offered. Asterisk is not guaranteed to respond to every media offer.
Notable cases are when multiple streams of the same type are offered.
The 2 media stream limit is still present with this patch, too.
In trunk and the 1.6.X branches, things will be a bit different since Asterisk
also supports text in SDPs as well.
(closes issue #12434)
Reported by: mnnojd
Review: https://reviewboard.asterisk.org/r/311
Review: https://reviewboard.asterisk.org/r/313
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to make merging easier. These changes are already on trunk.
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r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines
channels/chan_misdn.c
channels/misdn/isdn_lib.c
* Miscellaneous other fixes from trunk to make merging easier later.
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r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines
* Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
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r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
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r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
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r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
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r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
Merged revision 157977 from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
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Fixes JIRA ABE-1726
The dial extension could be empty if you are using MISDN_KEYPAD
to control ISDN provider features.
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r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines
Merged revisions 206938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines
SIP incorrect From: header information when callpres is prohib
Some ITSP make use of the "Anonymous" display name to detect a
requirement to withhold caller id across the PSTN. This does
not work if the display name is "Unknown".
(closes issue #14465)
Reported by: Nick_Lewis
Patches:
chan_sip.c-callerpres.patch uploaded by Nick (license 657)
chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
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r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
Session timer were not activated if Supported header field in INVITE had both "timer" and other options.
(closes issue #15403)
Reported by: makoto
Patches:
sip-session-timer.patch uploaded by makoto (license
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r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines
Merged revisions 206706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines
Merged revision 206700 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
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Fixed chan_misdn crash because mISDNuser library is not thread safe.
With Asterisk the mISDNuser library is driven by two threads concurrently:
1. channels/misdn/isdn_lib.c::manager_event_handler()
2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher()
Calls into the library are done concurrently and recursively from
isdn_lib.c.
Both threads can fiddle with the master/child layer3_proc_t lists. One
thread may traverse the list when the other interrupts it and then removes
the list element which the first thread was currently handling. This is
exactly what caused the crash. About 60 calls were needed to a Gigaset
CX475 before it occurred once.
This patch adds locking when calling into the mISDNuser library.
This also fixes some cb_log calls with wrong port parameter.
JIRA ABE-1913
Patches: misdn-locking.patch (Modified with mostly cosmetic changes)
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r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
callerid(num) is wrong when username is missing
A domain only sip uri <sip:123.123.123.123> would return
123.123.123.123 as callid num. Now, if the username is
missing from a uri, the callerid num field is left empty.
(closes issue #15476)
Reported by: viraptor
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r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines
Merged revisions 206487 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines
Fixes several call transfer issues with chan_misdn.
* issue #14355 - Crash if attempt to transfer a call to an application.
Masquerade the other pair of the four asterisk channels involved in the
two calls. The held call already must be a bridged call (not an
applicaton) or it would have been rejected.
* issue #14692 - Held calls are not automatically cleared after transfer.
Allow the core to initate disconnect of held calls to the ISDN port. This
also fixes a similar case where the party on hold hangs up before being
transferred or taken off hold.
* JIRA ABE-1903 - Orphaned held calls left in music-on-hold.
Do not simply block passing the hangup event on held calls to asterisk
core.
* Fixed to allow held calls to be transferred to ringing calls.
Previously, held calls could only be transferred to connected calls.
* Eliminated unused call states to simplify hangup code.
* Eliminated most uses of "holded" because it is not a word.
(closes issue #14355)
(closes issue #14692)
Reported by: sodom
Patches:
misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
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r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines
Merged revisions 206385 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines
Merged revisions 206384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines
Ensure apathetic replies are sent out on the proper socket.
chan_iax2 supports multiple address bindings. The send_apathetic_reply()
function did not attempt to send its response on the same socket that the
incoming message came in on.
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r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
SIP register not using peer's outbound proxy
If callbackextension is defined for a peer it successfully causes
a registration to occur, but the registration ignores the
outboundproxy settings for the peer. This patch allows the
peer to be passed to obproxy_get() in transmit_register().
(closes issue #14344)
Reported by: Nick_Lewis
Patches:
callbackextension_peer_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/294/
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r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines
Merged revisions 205804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines
SIP registration auth loop caused by stale nonce
If an endpoint sends two registration requests in a very short
period of time with the same nonce, both receive 401 responses
from Asterisk, each with a different nonce (the second 401
containing the current nonce and the first one being stale).
If the endpoint responds to the first 401, it does not match
the current nonce so Asterisk sends a third 401 with a newly
generated nonce (which updates the current nonce)... Now if
the endpoint responds to the second 401, it does not match the
current nonce either and Asterisk sends a fourth 401 with a
newly generated nonce... This loop goes on and on.
There appears to be a simple fix for this. If the nonce from
the request does not match our nonce, but is a good response
to a previous nonce, instead of sending a 401 with a newly
generated nonce, use the current one instead. This breaks
the loop as the nonce is not updated until a response is
received. Additional logic has been added to make sure no
nonce can be responded to twice though.
(closes issue #15102)
Reported by: Jamuel
Patches:
patch-bug_0015102 uploaded by Jamuel (license 809)
nonce_sip.diff uploaded by dvossel (license 671)
Tested by: Jamuel
Review: https://reviewboard.asterisk.org/r/289/
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r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines
Merged revisions 205775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines
Ensure that outbound NOTIFY requests are properly routed through stateful proxies.
With this change, we make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will have the proper
Route headers in them.
(closes issue #14725)
Reported by: ibc
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r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines
No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
(i.e. When libpri generates the event PRI_EVENT_ANSWER.)
(closes issue #15420)
Reported by: scottbmilne
Patches:
bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
Tested by: scottbmilne, alecdavis
(closes issue #15416)
Reported by: avinoash
(closes issue #15389)
Reported by: alecdavis
This patch should also fix the following issue:
(issue #15205)
Reported by: vinsik
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r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
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r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines
Merged revisions 204834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines
Removed confusing warning message "Got Busy in Connected State"
If an incoming mISDN call is answered with the Answer application and a
subsequent Dial gets a busy endpoint then it is valid for that already
connected channel to get the busy indication. Asterisk will play the busy
tones until the dialplan plays something else or hangs up the call.
(closes issue #11974)
Reported by: fvdb
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r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines
Merged revisions 204300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines
Add error message so that it is clear why a SIP peer was not processed when
a DNS lookup fails on a host or outboundproxy.
(closes issue #13432)
Reported by: p_lindheimer
Patches:
outboundproxy.patch uploaded by p (license 558)
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r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines
Merged revisions 204243,204246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines
Fix a problem where chan_sip would ignore "old" but valid responses.
chan_sip has had a problem for quite a long time that would manifest when
Asterisk would send multiple SIP responses on the same dialog before receiving
a response. The problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two requests out,
and a response arrived for the first request sent, then Asterisk would ignore
the response. The result was that Asterisk would continue retransmitting the
requests and ignoring the responses until the maximum number of retransmissions
had been reached.
The fix here is to rearrange the code a bit so that instead of simply comparing
the sequence number of the response to our latest outgoing sequence number, we
walk our list of outstanding packets and determine if there is a match. If there is,
we continue. If not, then we ignore the response.
In doing this, I found a few completely useless variables that I have now removed.
(closes issue #11231)
Reported by: flefoll
Review: https://reviewboard.asterisk.org/r/298
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r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines
Fix build oops.
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r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines
Merged revisions 203908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines
The ISDN CPE side should not exclusively pick B channels normally.
Before this patch, Asterisk unconditionally picked B channels exclusively
on the CPE side and normally allowed alternative B channels on the network
side. Now Asterisk does the opposite.
Reasons for the CPE side to normally not pick B channels exclusively:
* For CPE point-to-multipoint mode (i.e. phone side), the CPE side does
not have enough information to exclusively pick B channels. (There may be
other devices on the line.)
* Q.931 gives preference to the network side picking B channels.
* Some telcos require the CPE side to not pick B channels exclusively.
(closes issue #14383)
Reported by: mbrancaleoni
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r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines
Check if polarityonanswerdelay has elapsed before setting a channel as answered
after a polarity reversal.
Previously on a polarity switch event chan_dahdi would set the channel
immediately as answered. This would cause problems if a polarity reversal
occurred when the line was picked up as the dial would not have yet occurred.
Now if the polarity reversal occurs before delay has elapsed after coming off
hook or an answer, it is ignored. Also, some refactoring was done in
_handle_event.
(closes issue #13917)
Reported by: alecdavis
Patches:
chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | 10 lines
Unmute when we get a dtmfup (we muted on dtmfdown) event.
This would occasionally cause one-way audio when using hardware DTMF detection.
(closes issue #14761)
Reported by: tzafrir
Patches:
v1-14761.patch uploaded by dimas (license 88)
Tested by: tzafrir, dimas
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r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines
Merged revisions 203115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
Resolve a crash related to a T.38 reinvite race condition.
This change resolves a crash observed locally during some T.38 testing.
A call was set up using a call file, and when the T.38 reinvite came in,
the channel state was still AST_STATE_DOWN. The reason is explained by
a comment in the code that previously lived in the handling of
AST_STATE_RINGING. This change modifies the logic to handle the same
race condition for any channel state that is not UP.
(closes ABE-1895)
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r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines
Merged revisions 203036 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines
Improved chan_dahdi.conf pritimer error checking.
Valid format is: pritimer=timer_name,timer_value
* Fixed segfault if the ',' is missing.
* Completely check the range returned by pri_timer2idx() to prevent
possible access outside array bounds.
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r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines
Merged revisions 202341-202342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines
Fix a situation in which Asterisk would not stop retransmitting 487s.
If a CANCEL were received by Asterisk, we would send a 487 in response
to the original INVITE and a 200 OK for the CANCEL. If there were a network
hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating
with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used
to be to try sending another 487 to the canceled INVITE and another 200 OK to the
CANCEL.
The problem here is that the originally-sent 487 was sent "reliably" meaning that
it will be retransmitted until it is received properly. So when we receive the second
CANCEL it is likely that the first batch of 487s we sent is still going strong and
reaches the UA. The result was that the second set of 487s would be retransmitted
constantly until the maximum number of retries had been reached.
The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel
the retransmission of the first set of 487s and start a second set. This causes the
dialog to be terminated reasonably.
(closes issue #14584)
Reported by: klaus3000
Patches:
14584_v2.patch uploaded by mmichelson (license 60)
Tested by: klaus3000
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r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines
Remove an extra debug line left from previous commit.
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r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines
Merged revisions 202336 via svnmerge from
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r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines
Fix a possible infinite loop in SDP parsing during glare situation.
There was a while loop in get_ip_and_port_from_sdp which was controlled
by a call to get_sdp_iterate. The loop would exit either if what we were
searching for was found or if the return was NULL. The problem is that
get_sdp_iterate never returns NULL. This means that if what we were searching
for was not present, the loop would run infinitely. This modification of the
loop fixes the problem.
(closes issue #15213)
Reported by: schmidts
(closes issue #15349)
Reported by: samy
(closes issue #14464)
Reported by: pj
(closes issue #15345)
Reported by: aragon
Patches:
sip_inf_loop.patch uploaded by mmichelson (license 60)
Tested by: aragon
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r201570 | dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines
parsing extension correctly from sip register lines
If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'.
(closes issue #15111)
Reported by: ffs
Patches:
chan_sip.c_register-parser.patch uploaded by ffs (license 730)
Tested by: ffs, dvossel
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r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines
Fix problem with no audio due to ignoring the SDP.
A recent change to our SDP version comparison made audio not function
on some calls. This was because of a test wherein we were trying to
see if an unsigned value was less than 0. This is a dumb comparison
and arguably the compiler should have warned about it. Alas, though,
it slipped past. Now it's fixed by changing the variable to be a
signed type.
Found by several developers. Tested by mnicholson and dbrooks.
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r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines
SIP registry ref count error
During a sip reload, the list of sip_registry objects are
supposed to be traversed, unlinked, and destroyed, but
destruction never takes place due to a ref counting error.
This causes a memory leak when registry items are removed
from sip.conf and reloaded. While the registries are removed
from the global list, they are not removed from the scheduler.
Because of this, SIP register attempts continue to be sent
out for the item even though it may no longer be in the .conf.
(closes issue #15295)
Reported by: amorsen
Review: https://reviewboard.asterisk.org/r/282/
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r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
SIP transport type issues
What this patch addresses:
1. ast_sip_ouraddrfor() by default binds to the UDP address/port
reguardless if the sip->pvt is of type UDP or not. Now when no
remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
transport type, attempting to set the address and port to the
correct TCP/TLS bindings if necessary.
2. It is not necessary to send the port number in the Contact
header unless the port is non-standard for the transport type.
This patch fixes this and removes the todo note.
3. In sip_alloc(), the default dialog built always uses transport
type UDP. Now sip_alloc() looks at the sip_request (if present)
and determines what transport type to use by default.
4. When changing the transport type of a sip_socket, the file
descriptor must be set to -1 and in some cases the tcptls_session's
ref count must be decremented and set to NULL. I've encountered
several issues associated with this process and have created a function,
set_socket_transport(), to handle the setting of the socket type.
(closes issue #13865)
Reported by: st
Patches:
dont_add_port_if_tls.patch uploaded by Kristijan (license 753)
13865.patch uploaded by mmichelson (license 60)
tls_port_v5.patch uploaded by vrban (license 756)
transport_issues.diff uploaded by dvossel (license 671)
Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel
Review: https://reviewboard.asterisk.org/r/278/
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r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11 lines
Accept T.38 re-INVITE responses with invalid SDP versions.
This commit changes the 'incoming SDP version' check logic a bit more; when
'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to
switch to T.38, we'll always accept the peer's SDP response, even if they
don't properly increment the SDP version number as they should. If this situation
occurs, a warning message will be generated suggesting that the peer's
configuration be changed to include the 'ignoresdpversion' configuration option
(although ideally they'd fix their SIP implementation to be RFC compliant).
AST-221
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r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun 2009) | 3 lines
added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways
also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample
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r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
CLI NOTIFY sending wrong transport type.
SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer.
(closes issue #15283)
Reported by: jthurman
Patches:
sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
Tested by: jthurman, dvossel
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r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines
Correct "dahdi show channels" output when specifying a group.
Since a DAHDI channel may belong to multiple groups, we need to use
a bitwise and instead of equivalence to determine whether to display
the channel information.
(closes issue #15248)
Reported by: gentian
Patches:
15248.patch uploaded by mmichelson (license 60)
Tested by: gentian
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r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines
fixes issue with channels not going down after transfer
Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop.
(closes issue #15216)
Reported by: oxymoron
Tested by: dvossel
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r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines
Merged revisions 197562 via svnmerge from
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r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines
Use the address we already know when reloading a peer with nat=yes.
If we already have an address for a peer, and we are reloading the sip
configuration, try to use that address to contact the peer, instead of
getting it from the Contact.
(closes issue #15194)
Reported by: ibc
Patches:
sip.patch uploaded by eliel (license 64)
Tested by: manwe
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r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
Recorded merge of revisions 197588 via svnmerge from
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r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
Allow for media to arrive from an alternate source when responding to a reinvite with 491.
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
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r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines
Merged revisions 197466 via svnmerge from
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r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines
Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting.
The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
(or it passes through unauthenticated) the proper nat flag is set.
(closes issue #13823)
Reported by: dimas
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r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines
Display an error message when chan_alsa fails to load due to a missing
or inaccessible configuration file.
Before this change, when chan_alsa failed to load due to a missing or
inaccessible configuration file, no message would be displayed. With this
change, when chan_alsa fails to load due to a missing or inaccessible
configuration file, a message will be displayed.
(closes issue #14760)
Reported by: Nick_Lewis
Patches:
chan_alsa.c-confload.patch uploaded by Nick (license 657)
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r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
SIP set outbound transport type from Registration
In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset.
(closes issue #12282)
Reported by: rjain
Patches:
reg_patch_1.diff uploaded by dvossel (license 671)
Tested by: dvossel
(closes issue #14727)
Reported by: pj
Patches:
reg_patch_3.diff uploaded by dvossel (license 671)
Tested by: pj, dvossel
Review: https://reviewboard.asterisk.org/r/249/
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r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines
Merged revisions 195991 via svnmerge from
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r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines
Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer.
There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement.
(closes issue #15032)
Reported by: guillecabeza
Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380)
Tested by: guillecabeza
(closes issue #14216)
Reported by: Andrey Sofronov
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r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines
Merged revisions 194557,194685 via svnmerge from
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r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines
IAX2 "Ghost" Channels
There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away.
(closes issue #14207)
Reported by: clive18
Review: https://reviewboard.asterisk.org/r/246/
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r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
Update to previous IAX2 "Ghost" Channels patch.
Fixed some comments made on reviewboard for the previous patch.
(issue #14207)
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r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines
Merged revisions 194484 via svnmerge from
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r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines
Fix a race condition where a reinvite could trigger a 482 response.
The loop detection/spiral detection code in chan_sip used the owner
channel's state as a criterion for determining if the incoming INVITE
is a looped request. The problem with this is that the INVITE-handling
code happens in a different thread than the thread that marks the owner
channel as being up. As a result, if a reinvite were to come in very quickly,
say from another Asterisk on the same LAN, it was possible for the reinvite
to arrive before the owner channel had been set to the up state.
This patch corrects the problem by using the invitestate of the sip_pvt
instead, since that can be guaranteed to be set correctly by the time
the reinvite arrives. Since there is a switch statement further in the
INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate
of the sip_pvt in case we should actually be treating the channel as if it were
up already.
(closes issue #12215)
Reported by: jpyle
Patches:
12215_confirmed.patch uploaded by mmichelson (license 60)
Tested by: lmadsen
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r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines
Update spiral support in trunk and 1.6.X to match what is in 1.4.
In 1.4, a SIP spiral is treated the same way as a call forward. This
works much better than what is currently in trunk and 1.6.X. The code
in trunk and 1.6.X did not create a new call to the recipient of the spiral,
instead trying to continue the same call. In addition to just being plain
wrong, this also had the side effect of only being able to spiral calls
to other SIP channels.
With this in place, as long as call forwards are honored, SIP spirals
will work properly. This means that it will work for outbound calls
made by the Queue, Dial, and Page applications. For originated calls and
spool calls, however, the spiral will not work properly until a generic
call forward mechanism is introduced into Asterisk.
(relates to issue #13630)
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r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines
Merged revisions 193613 via svnmerge from
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r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines
Sent wrong message to clear a call we started if the other end has not responed yet.
In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet),
it is not allowed to clear the call with RELEASE_COMPLETE. It must be
cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer
to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b)
Patches:
chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862
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r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines
TCP not matching valid peer.
find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it.
Review: http://reviewboard.digium.com/r/236/
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r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines
Merged revisions 193050 via svnmerge from
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r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines
Give a more helpful message when an incoming call's dialed extension does not match.
Added the dialed extension and context to the chan_misdn messages warning
that the dialed number cannot be matched in the dialplan.
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r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines
Merged revisions 192932 via svnmerge from
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r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines
Eliminate repetition of fullcontact during reconstruction.
If the fullcontact field appears in both the sippeers and the
sipregs table, then during reconstruction of the field, it will
otherwise be doubled.
(closes issue #14754)
Reported by: Alexei Gradinari
Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
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r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) | 10 lines
Fix a bug where a timer would be created but not acknowledged.
This scenario crept up if chan_iax2 was loaded with no configuration file present.
It would create a timer and tell it to go at an interval but the thread that normally
acknowledges it would not be created because no configuration file was present. The timer
will now be closed if no configuration file is present.
(closes issue #15014)
Reported by: madkins
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r192387 | file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines
Fix a bug with setting t38pt_udptl at the user or peer level.
If an incoming call authenticated as a user or peer and t38pt_udptl was
not set to yes in general then no UDPTL session would be present and any
T38 related things would fail. This commit changes it so that if after
authenticating T38 is enabled but no UDPTL session is present one will be
created.
(issue AST-215)
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r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, 23 Apr 2009) | 13 lines
Merged revisions 190286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines
Fix a bug in chan_local glare hangup detection.
If both sides of a Local channel were hung up at around the same time it was
possible for one thread to destroy the local private structure and have the other thread
immediately try to remove the already freed structure from the local channel list.
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r189993 | jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines
Make chan_h323 respect packetization settings and fix small reload issue.
Previously, packetization settings were ignored and now they are not. A new
config option 'autoframing' has been added to mirror the way chan_sip handles
it. Turning on the autoframing option (available both as a global option or per
peer) overrides the local settings with the remote packetization settings.
Testing was performed with varying packetization levels with the following
codecs: ulaw, alaw, gsm, and g729.
Also, an unrelated config reload issue has been fixed in the case of the config
file not changing.
(closes issue #12415)
Reported by: pj
Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7),
modified by me
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r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) | 7 lines
Do not continue to receive DTMF, when the channel is hungup and about to be destroyed.
(closes issue #14858)
Reported by: barryf
Patches:
20090421__bug14858.diff.txt uploaded by tilghman (license 14)
Tested by: barryf
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r189771 | dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
Fixes segfault when switching UDP to TCP in sip.conf after reload.
If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found.
(closes issue #14727)
Reported by: pj
Tested by: dvossel
Review: http://reviewboard.digium.com/r/229/
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r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
Fix a bug with non-UDP connections that caused dialogs to not get freed.
This issue crept up because of a reference count issue on non-UDP based dialogs.
The dialog reference count was increased when transmitting a packet reliably but never
decreased. This caused the dialog structure to hang around despite being unlinked from
the dialogs container.
(closes issue #14919)
Reported by: vrban
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r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines
Merged revisions 188646 via svnmerge from
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r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines
National prefix inserted even when caller ID not available
When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
(closes issue #13207)
Reported by: shawkris
Patches:
national_prefix.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/220/
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r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines
Merged revisions 189203 via svnmerge from
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r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines
Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app
An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening.
(closes issue #14091)
Reported by: evandro
Patches:
autologoff.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/225/
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r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines
Prevent a crash when SIP blonde transferring an unbridged call.
If one attempts to use the attended transfer button on a SIP phone
to transfer an unbridged call (such as a call to an IVR) but hangs
up while the target of the transfer is still ringing, we need to not
crash.
The problem was that ast_hangup was called from outside the channel
thread.
AST-211
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r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
Merged revisions 188946 via svnmerge from
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r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
Fix a bug where a value used to create the channel name was bogus.
This commit fixes the scenario where an incoming call is authenticated
using a peer entry. Previously the channel name was created using either
the username setting from the sip.conf entry or the IP address that the
call came from. Now the channel name will be created using the peer name
itself. This commit will not change the way the channel name is generated
for users or friends.
(closes issue #14256)
Reported by: Nick_Lewis
Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file
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r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines
Fix a bug with the change I made yesterday to outbound proxy support.
Per discussion with oej on IRC we need the actual IP address, not the
outbound proxy IP address, in the sa field. Upon further inspection
this should make the behaviour of all other uses of the outbound proxy
in the code.
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r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1.
Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will
be sending to. This has to be done because the logic that determines what local IP address to use
in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address
we are sending to.
(closes issue #12006)
Reported by: mnicholson
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r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
Fix module embedding for chan_h323.
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
Patches:
issue_11966.patch uploaded by kpfleming (license 421)
Tested by: jpeeler
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r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines
Merged revisions 185845 via svnmerge from
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r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines
Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491
Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well.
(closes issue #12013)
Reported by: alx
Review: http://reviewboard.digium.com/r/213/
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r186928 | russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines
Update some comments and resolve potential memory corruption in chan_sip.
While browsing chan_sip the other day, I noticed this dangerous code in
dialog_needdestroy(). This function is an ao2_callback. It is absolutely
_not_ okay to unlock the container from within this function. It's also not
clear why it was useful. Given that it could cause memory corruption, I have
removed it.
There was also a TODO comment left describing a potential implementation of
an improvement to the needdestroy handling. I'm not convinced that what was
described is the best choice here, so I have briefly described the way that
this function is used today that could be improved.
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r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines
Merged revisions 186458 via svnmerge from
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r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines
Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested
Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later).
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r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines
Merged revisions 185952 via svnmerge from
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r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines
the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior.
this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized.
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r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines
Merged revisions 185362 via svnmerge from
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r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines
Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces
To drill into the xmpp to find the capabilities between channels, chan_gtalk
calls iks_child() and iks_next(). iks_child() and iks_next() are functions in
the iksemel xml parsing library that traverse xml nodes. The bug here is that
both iks_child() and iks_next() will return the next iks_struct node
*regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG,
which in most cases, it is, but in this case (a call being made from the
Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data
(they are extraneous whitespaces), and chan_gtalk doesn't handle that case,
so capabilities don't match, and a call cannot be made.
iks_first_tag() and iks_next_tag(), on the other hand, will not return the
very next iks_struct, but will check to see if the next iks_struct is of
type IKS_TAG. If it isn't, it will be skipped, and the next struct of type
IKS_TAG it finds will be returned. This assures that chan_gtalk will find
the iks_struct it is looking for.
This fix simply changes all calls to iks_child() and iks_next() to become
calls to iks_first_tag() and iks_next_tag(), which resolves the capability
matching.
The following is a payload listing from Empathy, which, due to the extraneous
whitespace, will not be parsed correctly by iksemel:
<iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/>
<payload-type clockrate='8000' name='PCMA' id='8'/>
<payload-type clockrate='8000' name='PCMU' id='0'/>
<payload-type clockrate='90000' name='MPA' id='97'/>
<payload-type clockrate='16000' name='SIREN' id='98'/>
<payload-type clockrate='8000' name='telephone-event' id='99'/>
</description>
</session>
</iq>
Review: http://reviewboard.digium.com/r/181/
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r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines
Merged revisions 185120 via svnmerge from
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r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines
Make chan_misdn BRI TE side normally defer channel selection to the NT side.
Channel allocation collisions are not handled by chan_misdn very well.
This patch simply avoids the problem for BRI only.
For PRI, allocation collisions are still possible but less likely since
there are simply more channels available and each end could use a different
allocation strategy.
misdn.conf options available:
te_choose_channel - Use to force the TE side to allocate channels.
method - Specify the channel allocation strategy.
(closes issue #13488)
Reported by: Christian_Pinedo
Patches:
isdn_lib.patch.txt uploaded by crich
Tested by: crich, siepkes, festr
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r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines
Merged revisions 184947 via svnmerge from
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r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines
Improve our handling of T38 in the initial INVITE from a device.
We now answer with matching media streams to what is requested. If an INVITE
is received with both a T38 and RTP media stream this means we answer with both.
For any outgoing calls created as a result of this inbound one no T38 is requested
in the initial INVITE. Instead if we start receiving udptl packets we trigger a
reinvite on the outbound side.
(closes issue #12437)
Reported by: marsosa
Tested by: pinga-fogo, okrief, file, afu
Review: http://reviewboard.digium.com/r/208/
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r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
Simplify chan_h323 build to not require a second run of "make".
(closes issue #14715)
Reported by: jthurman
Patches:
h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614)
Tested by: tzafrir, russell
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r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines
Improve timing interface to remember which provider provided a timer
The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
(closes issue #14697)
Reported by: moy
Review: http://reviewboard.digium.com/r/211/
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r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines
Merged revisions 184565 via svnmerge from
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r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines
Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls.
If calls were placed using an IP address or hostname the global nat setting was copied over
but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP
actions.
(closes issue #14546)
Reported by: acunningham
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r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines
Improve performance of the ast_event cache functionality.
This code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.
1) Asterisk 1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost.
One relatively minor change in this patch is that the extra processing
required for distributed device state is now completely bypassed if
it's not needed.
2) One of the things that I noticed when profiling this code was that a
_lot_ of time was spent doing string comparisons. I changed the way
strings are represented in an event to include a hash value at the front.
So, before doing a string comparison, we do an integer comparison on the
hash.
3) Finally, the code that handles the event cache has been re-written.
I tried to do this in a such a way that it had minimal impact on the API.
I did have to change one API call, though - ast_event_queue_and_cache().
However, the way it works now is nicer, IMO. Each type of event that
can be cached (MWI, device state) has its own hash table and rules for
hashing and comparing objects. This by far made the biggest impact on
performance.
For additional details regarding this code and how it was tested, please see the
review request.
(closes issue #14738)
Reported by: russell
Review: http://reviewboard.digium.com/r/205/
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r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines
Fix issue with a T38 reinvite being sent even if not configured to do so.
If we receive a T38 request negotiate control frame we should only attempt to do so
if the option is enabled on the dialog.
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r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines
Exclude slin16, siren7, and siren14 from bandwidth=low and =medium
The default codec configuration for chan_iax2 is bandwidth=low. I noticed
slin16 being negotiated as the codec in some test calls, but that no longer
happens after this change.
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r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines
Merged revisions 183115 via svnmerge from
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r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines
Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."
A user was having an issue where if an outgoing SIP call was canceled, the SIP device
would remain in use if we had not received any response to the initial INVITE we sent out.
The SIP device would remain in use until the autocongestion timer was exhausted.
I tracked down the cause of this to be the section of code I am removing here. I asked several
people what the purpose of this code was meant to be, but no one could give me any sort of
answer as to why this was here. The person who was having this issue has been using this patch
for several months and it has stopped the problems they have had.
AST-196
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r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
Improve our triggering of a T38 switchover internally when triggered by a received reinvite.
Previously we reached across the channel bridge to get the other party's SIP dialog
structure in order to trigger an outgoing reinvite. This is extremely dangerous to do
and only works if bridged to another SIP channel. This patch changes this to use the
T38 control frame method of requesting a switchover. This change also causes the SIP
channel driver to propogate back whether the switchover worked or not instead of blindly
accepting the incoming T38 reinvite.
Review: http://reviewboard.digium.com/r/200/
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r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines
Merged revisions 182810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
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r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
Allow H.323 Plus library to be used in addition to the OpenH323 library
Chan_h323 can now be compiled against both the previously supported versions of
OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure
script has been modified to look in the default install location of h323 to
hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR.
Also, the CLI command "h323 show version" has been added which indicates which
version of h323 is in use.
(closes issue #11261)
Reported by: vhatz
Patches:
asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614)
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r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) | 8 lines
OPENR2 uses an incorrect string value if the extension delimiter is not present.
* Fixed OPENR2 using an incorrect string value if the extension
delimiter is not present in the Dial() function. This was fixed for
SS7 and PRI in trunk -r172400.
* Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others.
* Removed trailing whitespace that appeared with OPENR2.
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This commit introduces official support for R2 signaling in chan_dahdi. The
modifications to chan_dahdi, and the supporting library, LibOpenR2, were both
written by Moises Silva.
Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6
in Brazil, México and Argentina. An unknown number of users (but at least 1)
are using it in each of the following countries: Colombia, Nepal, Thailand,
Venezuela, Perú, and probably others.
To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/.
Information about configuration can be found in configs/chan_dahdi.conf.sample.
The code committed is the most up to date version, which was being maintained
in svn/asterisk/team/moy/mfcr2/.
I would also like to include a Thank You to the many others that tested this
code beyond those listed in this commit message. These are the names that I
could find in the mantis issue.
(closes issue #12509)
Reported by: moy
Patches:
chan_zap-mfr2.patch uploaded by moy (license 222)
Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen
Review: http://reviewboard.digium.com/r/40/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
Randomize IAX2 encryption padding
The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
Review: http://reviewboard.digium.com/r/193/
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The code responsible for sending the T38 reinvite did not check if an INVITE was
already being handled. This caused things to get confused and the call to fail.
The code now defers sending the T38 reinvite until the current INVITE is done being
handled.
(issue AST-191)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines
Properly send a 487 on an INVITE we have not responded to if we receive a BYE.
If we receive an INVITE from an endpoint and then later receive a BYE from that
same endpoint before we have sent a final response for the INVITE, then we need
to respond to the INVITE with a 487.
There was logic in the code prior to this commit which seemed to exist solely to
handle this situation, but there was one condition in an if statement which
was incorrect. The only way we would send a 487 was if the sip_pvt had no owner
channel. This made no sense since we created the owner channel when we received
the INVITE, meaning that the majority of the time we would never send the 487.
The 487 being sent should not rely on whether we have created a channel. Its
delivery should be dependent on the current state of the initial INVITE transaction.
With this commit, that logic is now correctly in place.
(closes issue #14149)
Reported by: legranjl
Patches:
14149.patch uploaded by mmichelson (license 60)
Tested by: legranjl
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r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
(closes issue #14607)
Reported by: stevenla
Tested by: dvossel
Review: http://reviewboard.digium.com/r/192/
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r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines
Fix issue where an attended transfer could not be completed under a rare scenario.
When completing an attended transfer chan_sip does a check to make sure the extension
in the URI portion of the Refer-To header is a local valid extension. We don't actually
need to check this since we know for sure the other channel is already up and talking to
the extension. Some devices do not put the extension in the Refer-To header either, which
can cause the extension check to fail. We now no longer do this check if it is an attended
transfer.
(closes issue #14628)
Reported by: sverre
Patches:
14628.diff uploaded by file (license 11)
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r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines
Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
When dtmfmode was set to auto the inband DTMF detector was not setup
on outgoing SIP calls. This caused inband DTMF detection to fail.
The inband DTMF detector is now setup for both dtmfmode inband and auto.
(closes issue #13713)
Reported by: makoto
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue.
A few other issues were addressed:
- There were a few instances of functions improperly passing ast_free instead
of ast_free_ptr.
- Some clean up was done to avoid the debug macros intentionally being redefined.
(copied below from Kevin's commit, appreciate the help)
- disable astmm.h from doing anything when STANDALONE is defined, which is used
by the tools in the utils/ directory that use parts of Asterisk header files in
hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined.
(closes issue #13593)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit brings in the bridging core, bridging technologies,
and the ConfBridge application.
For usage information on the ConfBridge application please see
the output of "core show application ConfBridge" from the CLI.
For API documentation please see the doxygen page describing the
architecture and the documentation for each API call.
Review: http://reviewboard.digium.com/r/93/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, chan_sip had both sip_peer and sip_user objects in memory. A
patch went in to remove sip_user to simplify the code, since everything
could be done with just sip_peer. This patch resolves some regressions
found that were introduced by those changes.
This code comes from svn/asterisk/team/group/sip-object-matching/.
Here is a list of the changes that have been made:
1) When doing a match by name with the find_peer() function, make it much
easier to specify which objects should be matched by having a parameter
that specifies exactly which object types should be considered. Also,
update find_by_name() to handle this parameter. Finally, update all
code to use the new option values.
2) When looking up an object for an outbound request by name, consider
peers only. (create_addr())
3) Only match peers on an incoming registration request.
4) When doing authentication (except for SUBSCRIBE), look up users
by name, instead of all objects by name.
5) When doing authentication (except for SUBSCRIBE), after looking for
a user by name, look for a peer by IP address, instead of all objects
by IP address.
6) When handling the SIP qualify CLI command or manager action, look for
a peer by name, instead of any object by name.
7) When handling the SIP unregister CLI command, look for a peer by name,
instead of any object by name.
9) In sip_do_debug_peer(), search for a peer by name, instead of any object
by name.
9) When handling the SIPPEER() dialplan function, search for a peer by name,
instead of any object by name.
10) In the following session timer related functions, st_get_se(),
st_get_refresher(), and st_get_mode(), when looking for an object for a
given sip_pvt using pvt->peername, look for a peer by name, instead of any
object by name.
11) Fix build_peer() to properly handle the case where separate type=peer and
type=user entries were specified in sip.conf.
(closes issue #14505)
Reported by: lmadsen
Review: http://reviewboard.digium.com/r/172/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the subscription context for a call pickup subscription differs
from the context of the call pickup target, there's not an easy way
to divine what context should be used for the pickup. The way to work
around this is to use PICKUPMARK as the context for the pickup.
This has been documented in the sip.conf.sample file
(ABE-1708)
closes issue #14567
submitted by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit
was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called,
this inevitably resulted in the reading and writing of freed memory.
XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet
at all. The proper action to take is to remove the scheduler entry we just created,
free the packet's data as well as the packet itself, and unlink it from the list of
packets on the sip_pvt structure.
(closes issue #14455)
Reported by: Nick_Lewis
Patches:
14455.patch uploaded by mmichelson (license 60)
Tested by: Nick_Lewis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the way
we configure devices and lines.
(closes issue #10297)
Reported by: DEA
Patches:
skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
With mods by me based on feedback from wedhorn and Russell and seanbright
Tested by: DEA, mvanbaak, pj
Review: http://reviewboard.digium.com/r/130/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
(closes issue #14479)
Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.
(closes issue #14427)
Reported by: snuffy
Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
Review: http://reviewboard.digium.com/r/173/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
Skip check for extension when subscribing for MWI.
Since the remote side is not actually subscribing to a specific extension when
subscribing for MWI just skip the check to see if the extension exists. They can't use it
to specify the mailbox either since we require configuration of that in sip.conf
(closes issue #14531)
Reported by: festr
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Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled.
(closes issue #14523)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Several changes in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option change, install
location change, and a boolean type change which is handled by ast_ptlib.h.
Also, the sed check has been modified to properly work with autoconf >= 2.62.
(closes issue #14224)
Reported by: bergolth
Patches:
asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous implementation of T38 faxdetect resulted in both sides of the
call jumping to a fax extension when both sides had 't38pt_udptl=yes' and
'faxdetect=yes' in sip.conf and a 'fax' extension in the current context.
This revision will jump to a 'fax' extension on incoming calls only.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is required to create a UDPTL structure in create_addr_from_peer() to handle the
scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but
is defined the peer's context. I tested this patch by enabling t38pt_udptl in the
[general] section on one system and only enabling t38pt_udptl in a peer's context on
the system sending a fax. Without the patch, the sending system will fail to initiate
T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure".
When this patch is applied the sending side will successfully initiate T38 negotiation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
masq, so that the group count lingers past the bridge.
(closes issue #14275)
Reported by: kowalma
Patches:
20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines
After a 'sip reload', qualifies for realtime peers weren't immediately
restarted, instead waiting until the next registration. We're now
caching the qualify across a reload/restart and starting the qualify
immediately upon loading the peer.
(closes issue #14196)
Reported by: pdf
Patches:
20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
Tested by: pdf
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r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines
Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging
This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that.
issue #13749
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r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
Fixed iax2 key rotation backwards compatibility
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
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r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines
Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.
This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
pool was used for the value while the old was left untouched/unused. If the current pool was full a new
pool was created. This would cause memory usage to increase steadily.
(issue #AA50-2332)
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1) It had numerous coding guidelines violations with regards to formatting.
2) It allocated memory using ast_calloc() that was never freed.
3) It didn't check for failure from the allocation.
4) It used sprintf() and strcat() to build the result, doing zero checking to
prevent writing past the end of the provided buffer.
The function also lacks API documentation, but that has not been addressed in
this commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
Review: http://reviewboard.digium.com/r/159/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the 'faxdetect' configuration option is used, one may also want to use
the 'faxbuffers' configuration option in chan_dahdi.conf. This option will
dynamically use the configured 'faxbuffers' buffer policy on a channel for
the life of the call following the detection of fax tones. The faxbuffers
buffer policy will be reverted during call teardown.
An example use of 'faxbuffers' is below. This example would switch to using
6 buffers with a full buffer policy.
faxbuffers=>6,full
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
David Vossel
Jeff Peeler
Richard Mudgett
Kevin P. Fleming
Tilghman Lesher
Matthew Nicholson
Terry Wilson
Russell Bryant
Joshua Colp
Mark Michelson
Tzafrir Cohen
Moises Silva
Olle Johansson
Michiel van Baak
Sean Bright
David Brooks
Jason Parker
Matthew Fredrickson
Eliel C. Sardanons
Leif Madsen
Dwayne M. Hubbard