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...by Kashif Raza. Tested, Patched by seanbright.) * Change in 'sip show channels' display format allowing more digits for CID. (Closes issue #16459. Reported, Patched by Rzadzins. * Revise documentation on disposition values to the actual values used. (Closes issue #16289. Reported by wdoekes.) A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.29-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4...

...#16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces (for Solaris 10). (Patched by seanbright.) * Allow "REMAINDER" to function properly in expressions. (Closes issue #16427. Reported, Patched by wdoekes.) * Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (Reported, Tested by corruptor. Patched by tilghman.) * Fix channel name comparison for Bridge() application. (Closes issue #16528. Reported, Patched by telecos82.) A summary of changes in thi...

...rm after polarity reversal. (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. (Closes issue #16683. Reported by wdoekes) * Resolve a crash caused by a race condition in app_chanspy.c (Closes issue #16678. Reported, patched by tim_ringenbach. Tested by dvossel) * Fix deadlock in app_queue with use_weight during reload. (Closes issue #16677. Reported, patched by tim_ringenbach) * Stop playing the me...

...the realtime backend. This value is purely informational. It does not alter configuration at all. (Closes issue #16029. Reported, patched by Guggemand) * Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors. (Closes issue #17469. Reported, patched by wdoekes) * Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. (Closes issue #15871. Reported, patched by Ivan) * Correct h...

...n the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1-rc1 Thank you for your continued support of Asterisk!

...#17785. Reported, tested by ramonpeek. Patched by tilghman) * Set the caller id on CDRs when it is set on the parent channel. (Closes issue #17569. Reported, patched by tbelder) * Ensure user portion of SIP URI matches dialplan when using encoded characters (Closes issue #17892. Reported by wdoekes. Patched by jpeeler) * Fix a crash in res_jabber by ensuring that we don't alter memory after it's freed. (Closes issue #17387. Reported, tested by jmls. Patched by tilghman) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not onl...

...#17785. Reported, tested by ramonpeek. Patched by tilghman) * Set the caller id on CDRs when it is set on the parent channel. (Closes issue #17569. Reported, patched by tbelder) * Ensure user portion of SIP URI matches dialplan when using encoded characters (Closes issue #17892. Reported by wdoekes. Patched by jpeeler) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realt...

...as not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1 Thank you for your continued support of Asterisk!

...ves 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. Patc...

...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...

...by Kashif Raza. Tested, Patched by seanbright.) * Change in 'sip show channels' display format allowing more digits for CID. (Closes issue #16459. Reported, Patched by Rzadzins. * Revise documentation on disposition values to the actual values used. (Closes issue #16289. Reported by wdoekes.) A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.29-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4...

...#16509. Reported by Kashif Raza. Tested, Patched by seanbright.) * Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces (for Solaris 10). (Patched by seanbright.) * Allow "REMAINDER" to function properly in expressions. (Closes issue #16427. Reported, Patched by wdoekes.) * Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (Reported, Tested by corruptor. Patched by tilghman.) * Fix channel name comparison for Bridge() application. (Closes issue #16528. Reported, Patched by telecos82.) A summary of changes in thi...

...rm after polarity reversal. (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. (Closes issue #16683. Reported by wdoekes) * Resolve a crash caused by a race condition in app_chanspy.c (Closes issue #16678. Reported, patched by tim_ringenbach. Tested by dvossel) * Fix deadlock in app_queue with use_weight during reload. (Closes issue #16677. Reported, patched by tim_ringenbach) * Stop playing the me...

...the realtime backend. This value is purely informational. It does not alter configuration at all. (Closes issue #16029. Reported, patched by Guggemand) * Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors. (Closes issue #17469. Reported, patched by wdoekes) * Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. (Closes issue #15871. Reported, patched by Ivan) * Correct h...

...n the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1-rc1 Thank you for your continued support of Asterisk!

...#17785. Reported, tested by ramonpeek. Patched by tilghman) * Set the caller id on CDRs when it is set on the parent channel. (Closes issue #17569. Reported, patched by tbelder) * Ensure user portion of SIP URI matches dialplan when using encoded characters (Closes issue #17892. Reported by wdoekes. Patched by jpeeler) * Fix a crash in res_jabber by ensuring that we don't alter memory after it's freed. (Closes issue #17387. Reported, tested by jmls. Patched by tilghman) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not onl...

...#17785. Reported, tested by ramonpeek. Patched by tilghman) * Set the caller id on CDRs when it is set on the parent channel. (Closes issue #17569. Reported, patched by tbelder) * Ensure user portion of SIP URI matches dialplan when using encoded characters (Closes issue #17892. Reported by wdoekes. Patched by jpeeler) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realt...

...as not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1 Thank you for your continued support of Asterisk!

...ves 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. Patc...

...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...