ASTERWEB Blog

6Set/110

Rilasciato Asterisk 1.8.6.0

Il giorno 31 agosto, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.6

Dal post originale:
The release of Asterisk 1.8.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Fix an issue with Music on Hold classes losing files in playlist when realtime
is used.
(Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor
Goncharovsky)
Resolve a potential crash in chan_sip when utilizing auth= and performing a
'sip reload' from the console.
(Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett)
Address some improper sql statements in res_odbc that would cause an update
to fail on realtime peers due to trying to set as "(NULL)" rather than an
actual NULL.
(Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by Tilghman
Lesher)
Resolve issue where 403 Forbidden would always be sent maximum number of times
regardless to receipt of ACK.
(Patched by Richard Mudgett)
Resolve issue where if a call to MeetMe includes both the dynamic(D) and
always request PIN(P) options, MeetMe will ask for the PIN two times: once for
creating the conference and once for entering the conference.
(Patched by Kinsey Moore)
Fix New Zealand indications profile based on
http://www.telepermit.co.nz/TNA102.pdf
(Closes issue ASTERISK-16263. Reported, Patched by richardf)
Segfault in shell_helper in func_shell.c
(Closes issue ASTERISK-18109. Reported by Michael Myles, patched by Richard
Mudgett)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0

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10Ago/110

Rilasciato Asterisk 1.6.20.0

Il giorno 08 agosto, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.6.20.0

Dal post originale:
The Asterisk Development Team announces the release of Asterisk 1.6.2.20. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.20 resolves a regression that was introduced just
prior to the release of Asterisk 1.6.2.19.

Fix reload crash caused by destroying default parking lot.
(Closes issue ASTERISK-18103. Reported by 808blogger. Patched by jrose.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

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2Ago/110

Rilasciato Asterisk 10.0.0 Beta 1

Il giorno 22 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 10.0.0 Beta 1

Dal post originale:
The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 10.0.0-beta1. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at
http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a...

All interested users of Asterisk are encouraged to participate in the
Asterisk 10 testing process. Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira. It is also very useful to see
successful test reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.
Additionally users can make use of the RPM and DEB packages now being built for
all Asterisk releases. More information available at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of included features includes:

T.38 gateway functionality has been added to res_fax.
Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
Support for defining hints has been added to pbx_lua.
Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

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13Lug/110

Rilasciato Asterisk 1.8.5

Il giorno 11 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.5

Dal post originale:
The release of Asterisk 1.8.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
cmaj)
Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
rossbeer, kowalma, Freddi_Fonet)
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
Fix a nasty chanspy bug which was causing a channel leak every time a spied on
channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant)
Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
the call that the dialplan started an AGI script for is hungup while the AGI
script is in the middle of a command then the AGI script is not notified of
the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
Resolve issue where leaving a voicemail, the MWI message is never sent. The
same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
Mudgett)
Resolve issue where wait for leader with Music On Hold allows crosstalk
between participants. Parenthesis in the wrong position. Regression from issue
#14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

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7Lug/110

Rilasciato libpri 1.4.12

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Il giorno 06 luglio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Libpri 1.4.12

Dal post originale:
The Asterisk Development Team announces the release of libpri version
1.4.12. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/

The following are some of the issues resolved in this release:

Add call transfer exchange of subaddresses support and fix PTMP call
transfer signaling.
Invalid PTMP redirecting signaling as TE towards NT.
Add Q931_IE_TIME_DATE to CONNECT message when in network mode.
(issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett)
Swap of master/slave in pri_enslave() incorrect.
(issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie)
Fix I-frame retransmission quirks.
Crash if NFAS swaps D channels on a call with an active timer.
DMS-100 not receiving caller name anymore.
(issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett)
B channel lost by incoming call in BRI NT PTMP mode.
Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls.

This release contains several new features, among them:

ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
ETSI Advice Of Charge (AOC) support
ETSI Explicit Call Transfer (ECT) support
ETSI Call Waiting support for ISDN phones
ETSI Malicious Call ID support
Add Display IE text handling options.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1....

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30Giu/110

Rilasciato Asterisk 1.8.5-rc1

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Il giorno 29 giugno, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.5-rc1

Dal post originale:
The Asterisk Development Team announces the first release candidate of
Asterisk 1.8.5. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.5-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
cmaj)
Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
rossbeer, kowalma, Freddi_Fonet)
Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
Fix a nasty chanspy bug which was causing a channel leak every time a spied on
channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
This patch fixes a bug with MeetMe behavior where the 'P' option for always
prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant)
Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
the call that the dialplan started an AGI script for is hungup while the AGI
script is in the middle of a command then the AGI script is not notified of
the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
Resolve issue where leaving a voicemail, the MWI message is never sent. The
same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
Mudgett)
Resolve issue where wait for leader with Music On Hold allows crosstalk
between participants. Parenthesis in the wrong position. Regression from issue
#14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
Fix timerfd locking issue.
(Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1

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30Giu/110

Rilasciato Asterisk 1.6.2.19 (Final Maintenance Release)

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Il giorno 29 giugno, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.6.2.19 (Final Maintenance Release)

Dal post originale:
The Asterisk Development Team has announced the final maintenance release of
Asterisk, version 1.6.2.19. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Please note that Asterisk 1.6.2.19 is the final maintenance release from the
1.6.2 branch. Support for security related issues will continue until April 21,
2012. For more information about support of the various Asterisk branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.6.2.19 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection
every time someone connects via AMI. It should only be sent to
the user who just connected.
(Closes issue #18168. Reported, patched by FeyFre)
Fix thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma,
Freddi_Fonet. Patched by dvossel)
Don't delay DTMF in core bridge while listening for DTMF features.
(Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by
globalnetinc, jde. Patched by oej, twilson)
Fix chan_local crashs in local_fixup()
Thanks OEJ for tracking down the issue and submitting the patch.
(Closes issue #19053. Reported, patched by oej)
Don't offer video to directmedia callee unless caller offered it as well
(Closes issue #19195. Reported, patched by one47)

Additionally security announcements AST-2011-008, AST-2011-010, and
AST-2011-011 have been resolved in this release.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.19

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30Giu/110

Rilasciato Asterisk 1.4.42 (Final Maintenance Release)

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Il giorno 29 giugno, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.4.42 (Final Maintenance Release)

Dal post originale:
he Asterisk Development Team has announced the final maintenance release of
Asterisk, version 1.4.42. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Please note that Asterisk 1.4.42 is the final maintenance release from the
1.4 branch. Support for security related issues will continue until April 21,
2012. For more information about support of the various Asterisk branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.4.42 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Resolve regression with ring groups in the Dial() application
(Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero)
Resolve deadlock when using tab completion on the 'meetme kick' CLI command
when an invalid (non-existent) conference room is specified.
(Closes issue ASTERISK-17771. Reported, patched by zvision)
Resolve issue where voice frames could be dropped when checking for T.38
during early media.
(Closes issue ASTERISK-17705. Reported, patched by oej)
Resolve issue where DYNAMIC_FEATURES would not activate after a recent
DTMF fix.
(Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)

Additionally security announcements AST-2011-010, and AST-2011-011 have been
resolved in this release.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.42

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30Giu/110

Rilasciati Asterisk 1.4.41.2, 1.6.2.18.2 e 1.8.4.4 (Security Release)

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Il giorno 28 giugno, il Team di Sviluppo di Asterisk ha annunciato il rilascio delle versioni Asterisk 1.4.41.2, 1.6.2.18.2 e 1.8.4.4 (Security Release)

Dal post originale:
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the
following issue:

AST-2011-011: Asterisk may respond differently to SIP requests from an
invalid SIP user than it does to a user configured on the system, even when the
alwaysauthreject option is set in the configuration. This can leak information
about what SIP users are valid on the Asterisk system.

For more information about the details of this vulnerability, please read
the security advisory AST-2011-011, which was released at the same time as this
announcement.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

Security advisory AST-2011-011 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-011.pdf

27Giu/110

Rilasciato Asterisk 1.6.2.19-rc1

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Il giorno 24 giugno, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.6.2.19-rc1

Dal post originale:
Please note that Asterisk 1.6.2.19 will be the final maintenance release from the
1.6.2 branch. Support for security related issues will continue for one
additional year. For more information about support of the various Asterisk
branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.6.2.19-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection
every time someone connects via AMI. It should only be sent to
the user who just connected.
(Closes issue #18168. Reported, patched by FeyFre)
Fix thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma,
Freddi_Fonet. Patched by dvossel)
Don't delay DTMF in core bridge while listening for DTMF features.
(Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by
globalnetinc, jde. Patched by oej, twilson)
Fix chan_local crashs in local_fixup()
Thanks OEJ for tracking down the issue and submitting the patch.
(Closes issue #19053. Reported, patched by oej)
Don't offer video to directmedia callee unless caller offered it as well
(Closes issue #19195. Reported, patched by one47)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.19-rc1

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27Giu/110

Rilasciate nuove versioni Asterisk: 1.8.4.3, 1.6.2.18.1 e 1.4.41.1

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Il giorno 23 giugno, il Team di Sviluppo di Asterisk ha annunciato il rilascio delle versioni Asterisk 1.8.4.3, 1.6.2.18.1 e 1.4.41.1

Dal post originale:
he release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues
as outlined below:

AST-2011-008: If a remote user sends a SIP packet containing a null,
Asterisk assumes available data extends past the null to the
end of the packet when the buffer is actually truncated when
copied. This causes SIP header parsing to modify data past
the end of the buffer altering unrelated memory structures.
This vulnerability does not affect TCP/TLS connections.
-- Resolved in 1.6.2.18.1 and 1.8.4.3
AST-2011-009: A remote user sending a SIP packet containing a Contact header
with a missing left angle bracket (<) causes Asterisk to access a null pointer. -- Resolved in 1.8.4.3 AST-2011-010: A memory address was inadvertently transmitted over the network via IAX2 via an option control frame and the remote party would try to access it. -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3 The issues and resolutions are described in the AST-2011-008, AST-2011-009, and AST-2011-010 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-... http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-... http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-... Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available at: http://downloads.asterisk.org/pub/security/AST-2011-008.pdf http://downloads.asterisk.org/pub/security/AST-2011-009.pdf http://downloads.asterisk.org/pub/security/AST-2011-010.pdf

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4Giu/110

Rilasciato Asterisk 1.8.4.2 (security release)

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Il giorno 2 giugno, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.4.2

Dal post originale:
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which
can lead to a remotely exploitable crash:

Remote Crash Vulnerability in SIP channel driver (AST-2011-007)

The issue and resolution is described in the AST-2011-007 security
advisory.

For more information about the details of this vulnerability, please read the
security advisory AST-2011-007, which was released at the same time as this
announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

Security advisory AST-2011-007 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-007.pdf

25Mag/110

Rilasciato Asterisk 1.8.4.1

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Il giorno 24 maggio, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.4.1

Dal post originale:
The release of Asterisk 1.8.4.1 resolves several issues reported by the
community. Without your help this release would not have been possible.
Thank you!

Below is a list of issues resolved in this release:

Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)
(Closes issue #18951. Reported by jmls. Patched by wdoekes)
Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
This issue was found and reported by the Asterisk test suite.
(Closes issue #18951. Patched by mnicholson)
Resolve potential crash when using SIP TLS support.
(Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by
vois, Chainsaw)
Improve reliability when using SIP TLS.
(Closes issue #19182. Reported by st. Patched by mnicholson)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1

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6Mag/110

Rilasciato Asterisk 1.8.4-rc3

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Il giorno 26 aprile, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.8.4-rc3

Dal post originale:
The release of Asterisk 1.8.4-rc3 resolves a couple of issues since the last
release candidate, including two security related issues (AST-2011-005 and
AST-2011-006).

Use SSLv23_client_method instead of old SSLv2 only.
(Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
and chazzam.
Resolve crash in ast_mutex_init()
(Patched by twilson)
Includes changes per AST-2011-005 and AST-2011-006

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4-rc3

Information about the security releases are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

6Mag/110

Rilasciato Asterisk 1.6.2.18

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Il giorno 26 aprile, il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione Asterisk 1.6.2.18

Dal post originale:
The release of Asterisk 1.6.2.18 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

Only offer codecs both sides support for directmedia.
(Closes issue #17403. Reported, patched by one47)
Resolution of several DTMF based attended transfer issues.
(Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
shihchuan, grecco. Patched by rmudgett)
NOTE: Be sure to read the ChangeLog for more information about these changes.
Resolve deadlocks related to device states in chan_sip
(Closes issue #18310. Reported, patched by one47. Patched by jpeeler)
Fix channel redirect out of MeetMe() and other issues with channel softhangup
(Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.
Patched by russellb)
Fix voicemail sequencing for file based storage.
(Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by
jpeeler)
Guard against retransmitting BYEs indefinitely during attended transfers with
chan_sip.
(Review: https://reviewboard.asterisk.org/r/1077/)

In addition to the changes listed above, commits to resolve security issues
AST-2011-005 and AST-2011-006 have been merged into this release. More
information about AST-2011-005 and AST-2011-006 can be found at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18