https://origsvn.digium.com/svn/asterisk/trunk
........
r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) | 8 lines
Update confusing documentation for tlsbindaddr.
Update some confusing documentation for the tlsbindaddr
option in sip.conf.sample. Point at a link instead which
has better documentation.
(closes issue #17054)
Reported by: klaus3000
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@255054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r252762 | lmadsen | 2010-03-16 13:48:22 -0500 (Tue, 16 Mar 2010) | 15 lines
Merged revisions 252761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines
Additional extensions.ael global variable fixes.
Fixing up a couple more overlapping global variable namespaces shared with
extensions.conf.sample. Also noticed a few of the lines that were commented
out didn't have the closing semi-colon so I added that as well.
(issue #17035)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@252765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r252534 | lmadsen | 2010-03-15 15:52:32 -0500 (Mon, 15 Mar 2010) | 15 lines
Merged revisions 252533 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines
Update extensions.ael file to not overlap extensions.conf.
Updated the extensions.ael file so the global variables don't overlap
those that we have in extensions.conf (sample files). This way unexpected
things won't happed hopefully if both pbx_ael and res_config are loaded.
(closes issue #17035)
Reported by: pprindeville
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@252537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@252134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
fixes adaptive jitterbuffer configuration
When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default. This value is required
in order for the adaptive jitterbuffer to work correctly. To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@249907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #16491)
Reported by: jamicque
Patches:
20100114__issue16491.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
........
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
........
r213635 | dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines
fixes sip register parsing when user@domain is used
(issue #15008)
(issue #15672)
........
r215222 | tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines
Fix register such that lines with a transport string, but without an authuser, parse correctly.
(AST-228)
........
r215801 | tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines
Default the callback extension to "s". This is a regression.
(closes issue #15764)
Reported by: elguero
Change-type: bugfix
........
r235132 | dvossel | 2009-12-15 12:43:06 -0600 (Tue, 15 Dec 2009) | 14 lines
reverse minor sip registration regression
A registration regression caused by a code tweak in (issue #14331)
and a bug fix in (issue #15539) caused some sip registration
config entries to be constructed incorrectly. Origially
issue #14331 contained the code tweak as well as a bug fix, but since
the issue was reported as a tweak the bug fix portion was moved into
issue #15539. Both the tweak and the bug fix contained minor incorrect
logic that resulted in some SIP registrations to fail.
(issue #14331)
(issue #15539)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@242514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
Change fax detection in chan_sip so it behaves as one would expect.
Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@230882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines
Additional fixes to the extensions.conf.sample file.
Update the extensions.conf.sample [stdexten] context so that we use the
variable instead of requiring it to be passed explicitly. Also updated uses of
the [stdexten] context throughout.
(closes issue #15858)
Reported by: pprindeville
Patches:
stdexten-context-update.txt uploaded by lmadsen (license 10)
Tested by: pprindeville
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines
Merged revisions 226382 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines
Update documentation in sip.conf.sample.
Update the documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
is only used to stop Asterisk from generating a reINVITE, but does not stop
it from accepting them if necessary.
(closes issue #15644)
Reported by: lmadsen
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines
Merged revisions 225032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines
Remove ability to control T.38 FAX error correction from udptl.conf.
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@221598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines
Merged revisions 221360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@221486 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@221301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines
Merged revisions 216430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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 :-)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@216645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines
Merged revisions 209131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines
Allow for UDPTL to use only even-numbered ports if desired.
There are some VoIP providers out there that will not accept SDP
offers with odd numbered UDPTL ports. While it is my personal opinion
that these VoIP providers are misinterpreting RFC 2327, it really is
not a big deal to play along with their silly little games. Of course,
since restricting UDPTL ports to only even numbers reduces the range
of available ports by half, so the option to use only even port numbers
is off by default. A user can enable the behavior by setting
use_even_ports=yes in udptl.conf.
(closes issue #15182)
Reported by: CGMChris
Patches:
15182.patch uploaded by mmichelson (license 60)
Tested by: CGMChris
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
to make merging easier. These changes are already on trunk.
................
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.
........
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
........
r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
........
r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
........
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.
................
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
........
Fixes JIRA ABE-1726
The dial extension could be empty if you are using MISDN_KEYPAD
to control ISDN provider features.
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines
This patch adds a new 'ignoresdpversion' option to sip.conf. When this is
enabled (either globally or for a specific peer), chan_sip will treat any SDP
data it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session version
received is different from the current SDP session version. This option is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with Microsoft OCS
which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/
(closes issue #13958)
Reported by: toc
Tested by: toc
........
r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change does not affect any other 1.6 branches as they have
already been updated for other changes, which uses the word 'domain'
as I have here.
(closes issue #15204)
Reported by: okrief
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines
Merged revisions 193193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.)
The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 May 2009) | 8 lines
Ensure that by default only one console channel driver is loaded
This configuration file was changed to ensure that only one console channel driver
(chan_oss) is loaded by default, but the change would only work if chan_console
was not built. Now it will work as expected; if chan_alsa or chan_console are built
and installed, they will not be loaded unless explicity requested.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@191956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2. Fixed the issue caused by without certificate files.
3. Fixed the issue caused by number portability parameters in user part of RURI.
4. Updated for OSP Toolkit 3.5.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
