Re: [Asterisk-Users] Asterisk 1.0.10
Can anyone point me to the changelog for 1.0.10? Craig - Original Message - From: "Pedro" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, November 22, 2005 10:04 PM Subject: [Asterisk-Users] Asterisk 1.0.10 I noticed that asterisk.org <http://asterisk.org> now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.xversions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.10
I noticed that asterisk.org now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.x versions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10 version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
great! tnx matt.On 11/16/05, Matt Riddell <[EMAIL PROTECTED]> wrote: Mark Quitoriano wrote:> you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9?Yes, you can read the upgrade.txt file inside the RC2 distribution forinformation on the required changes. --Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Mark Quitoriano, CCNAFan the flame...http://www.spreadfirefox.com/?q=user/register&r=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
Mark Quitoriano wrote: > you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9? Yes, you can read the upgrade.txt file inside the RC2 distribution for information on the required changes. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9?On 11/14/05, Matt Riddell <[EMAIL PROTECTED] > wrote:Mark Quitoriano wrote:> but that's already 1.2? is it advisable to upgrade my current version > 1.0.9 to 1.2 already? any big changes to be done to my current setup to> upgrade it to 1.2?If in three or four days you go to upgrade your version of 1.0.9, it will beupgraded to 1.2.So, why not do it now, that way you won't end up creating a dialplan only for it to not work in a couple of days. -- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
Mark Quitoriano wrote: > but that's already 1.2? is it advisable to upgrade my current version > 1.0.9 to 1.2 already? any big changes to be done to my current setup to > upgrade it to 1.2? If in three or four days you go to upgrade your version of 1.0.9, it will be upgraded to 1.2. So, why not do it now, that way you won't end up creating a dialplan only for it to not work in a couple of days. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
but that's already 1.2? is it advisable to upgrade my current version 1.0.9 to 1.2 already? any big changes to be done to my current setup to upgrade it to 1.2?On 11/14/05, Matt Riddell <[EMAIL PROTECTED]> wrote: Mark Quitoriano wrote:> hey guys,>> if i get the asterisk from CVS like "cvs checkout -r v1-0 zaptel libpri> asterisk asterisk-addons asterisk-sounds" do i get a stable one?Yes, although version 1.2 in in release candidate stage and should be releasedlater this week.This means that if you were to do that, you would end up with a copy whichwill not be current by the end of the week.I would recommend grabbing the release candidate from the http://www.asterisk.org site.Hope this makes sense!:)--Cheers,Matt Riddell___ http://www.sineapps.com/news.php (Daily Asterisk News - html)http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame...http://www.spreadfirefox.com/?q=user/register&r=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
Mark Quitoriano wrote: > hey guys, > > if i get the asterisk from CVS like "cvs checkout -r v1-0 zaptel libpri > asterisk asterisk-addons asterisk-sounds" do i get a stable one? Yes, although version 1.2 in in release candidate stage and should be released later this week. This means that if you were to do that, you would end up with a copy which will not be current by the end of the week. I would recommend grabbing the release candidate from the http://www.asterisk.org site. Hope this makes sense! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
hey guys, if i get the asterisk from CVS like "cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds" do i get a stable one? On 11/11/05, Mark Quitoriano <[EMAIL PROTECTED]> wrote: Great! tnx matt!On 11/11/05, Matt Florell < [EMAIL PROTECTED]> wrote: It's CVS v1-0. Digium has said that they will do a release of 1.0.10at the same time they release 1.2.I highly recommend upgrading to this if you are still on the 1.0 tree.It has a lot of bug fixes, and the new v2 firmware telco cards from Digium run much better on it than they do on 1.0.9.If you want it now, just checkout from CVS like this:cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-soundsMATT--- On 11/10/05, Mark Quitoriano <[EMAIL PROTECTED]> wrote:> in the Changelog on> http://ftp.digium.com/pub/asterisk/ChangeLog there's a> asterisk 1.0.10 which i can't find anywhere, any hints?>>> --snip from ChangeLog--> Asterisk 1.0.10>> -- chan_local> -- In releases 1.0.8 and 1.0.9 , the Local channels that are created would> not be masqueraded into the new channel type. This has now been fixed.> -- chan_sip>> -- The 'insecure' options have been changed to support matching peersby IP > only, not requiring authentication on incoming invites, or both. Before,> to not require authentication on incoming invites also required matching>> peers based on IP only.> -- chan_zap > -- Before, call waiting could occur during the initial ringing on the line.> This has now been fixed.> -- app_disa> -- We will now not set the accountcode if one is not supplied.> > -- app_meetme> -- If the first caller into a conference hangs up while being prompted for> the conference pin number, the conference will no longer be held open.> -- app_userevent> -- Events created with this application were indicated as a "call" event >> instead of a "user" event. This made the "user" event permissions> not work correctly.> -- app_voicemail> -- When using the externpass option for voicemail, the password will be >> immediately updated in memory as well, instead of having to wait for> the next time the configuration is reloaded.> -- app_zapras> -- We now ensure buffer policy is restored after RAS is done with a > channel.>> This could cause audio problems on the channel after zapras is done> with it.> -- res_agi> -- We now unmask the SIGHUP signal before executing an AGI script. This > fixes problems where some AGI scripts would continue running long after>> the call is over.> -- extensions> -- A potential crash has been fixed when calling LEN() to get the length of > a string that was 80 characters or larger.> -- logger> -- The Asterisk logger will automatically detect when a log file needs to>> be rotated. However, this feature could put Asterisk in a nasty loop > that would result in a crash.> -- general> -- Added man pages for astgenkey, autosupport, and safe_asterisk> --end of snip-->> --> Regards,> Mark Quitoriano, CCNA > http://www.atamanetworks.com>> Fan the flame...> http://www.spreadfirefox.com/?q=user/register&r=19441 > ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users> >___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Mark Quitoriano, CCNA http://www.atamanetworks.comFan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441 -- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
Great! tnx matt!On 11/11/05, Matt Florell <[EMAIL PROTECTED]> wrote: It's CVS v1-0. Digium has said that they will do a release of 1.0.10at the same time they release 1.2.I highly recommend upgrading to this if you are still on the 1.0 tree.It has a lot of bug fixes, and the new v2 firmware telco cards from Digium run much better on it than they do on 1.0.9.If you want it now, just checkout from CVS like this:cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-soundsMATT--- On 11/10/05, Mark Quitoriano <[EMAIL PROTECTED]> wrote:> in the Changelog on> http://ftp.digium.com/pub/asterisk/ChangeLog there's a> asterisk 1.0.10 which i can't find anywhere, any hints?>>> --snip from ChangeLog--> Asterisk 1.0.10>> -- chan_local> -- In releases 1.0.8 and 1.0.9 , the Local channels that are created would> not be masqueraded into the new channel type. This has now been fixed.> -- chan_sip>> -- The 'insecure' options have been changed to support matching peersby IP > only, not requiring authentication on incoming invites, or both. Before,> to not require authentication on incoming invites also required matching>> peers based on IP only.> -- chan_zap > -- Before, call waiting could occur during the initial ringing on the line.> This has now been fixed.> -- app_disa> -- We will now not set the accountcode if one is not supplied.> > -- app_meetme> -- If the first caller into a conference hangs up while being prompted for> the conference pin number, the conference will no longer be held open.> -- app_userevent> -- Events created with this application were indicated as a "call" event >> instead of a "user" event. This made the "user" event permissions> not work correctly.> -- app_voicemail> -- When using the externpass option for voicemail, the password will be >> immediately updated in memory as well, instead of having to wait for> the next time the configuration is reloaded.> -- app_zapras> -- We now ensure buffer policy is restored after RAS is done with a > channel.>> This could cause audio problems on the channel after zapras is done> with it.> -- res_agi> -- We now unmask the SIGHUP signal before executing an AGI script. This > fixes problems where some AGI scripts would continue running long after>> the call is over.> -- extensions> -- A potential crash has been fixed when calling LEN() to get the length of > a string that was 80 characters or larger.> -- logger> -- The Asterisk logger will automatically detect when a log file needs to>> be rotated. However, this feature could put Asterisk in a nasty loop > that would result in a crash.> -- general> -- Added man pages for astgenkey, autosupport, and safe_asterisk> --end of snip-->> --> Regards,> Mark Quitoriano, CCNA > http://www.atamanetworks.com>> Fan the flame...> http://www.spreadfirefox.com/?q=user/register&r=19441 > ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users>>___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.10?
It's CVS v1-0. Digium has said that they will do a release of 1.0.10 at the same time they release 1.2. I highly recommend upgrading to this if you are still on the 1.0 tree. It has a lot of bug fixes, and the new v2 firmware telco cards from Digium run much better on it than they do on 1.0.9. If you want it now, just checkout from CVS like this: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds MATT--- On 11/10/05, Mark Quitoriano <[EMAIL PROTECTED]> wrote: > in the Changelog on > http://ftp.digium.com/pub/asterisk/ChangeLog there's a > asterisk 1.0.10 which i can't find anywhere, any hints? > > > --snip from ChangeLog-- > Asterisk 1.0.10 > > -- chan_local > -- In releases 1.0.8 and 1.0.9, the Local channels that are created would > not be masqueraded into the new channel type. This has now been fixed. > -- chan_sip > > -- The 'insecure' options have been changed to support matching peersby IP > only, not requiring authentication on incoming invites, or both. Before, > to not require authentication on incoming invites also required matching > > peers based on IP only. > -- chan_zap > -- Before, call waiting could occur during the initial ringing on the line. > This has now been fixed. > -- app_disa > -- We will now not set the accountcode if one is not supplied. > > -- app_meetme > -- If the first caller into a conference hangs up while being prompted for > the conference pin number, the conference will no longer be held open. > -- app_userevent > -- Events created with this application were indicated as a "call" event > > instead of a "user" event. This made the "user" event permissions > not work correctly. > -- app_voicemail > -- When using the externpass option for voicemail, the password will be > > immediately updated in memory as well, instead of having to wait for > the next time the configuration is reloaded. > -- app_zapras > -- We now ensure buffer policy is restored after RAS is done with a > channel. > > This could cause audio problems on the channel after zapras is done > with it. > -- res_agi > -- We now unmask the SIGHUP signal before executing an AGI script. This > fixes problems where some AGI scripts would continue running long after > > the call is over. > -- extensions > -- A potential crash has been fixed when calling LEN() to get the length of > a string that was 80 characters or larger. > -- logger > -- The Asterisk logger will automatically detect when a log file needs to > > be rotated. However, this feature could put Asterisk in a nasty loop > that would result in a crash. > -- general > -- Added man pages for astgenkey, autosupport, and safe_asterisk > --end of snip-- > > -- > Regards, > Mark Quitoriano, CCNA > http://www.atamanetworks.com > > Fan the flame... > http://www.spreadfirefox.com/?q=user/register&r=19441 > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk 1.0.10?
in the Changelog on http://ftp.digium.com/pub/asterisk/ChangeLog there's a asterisk 1.0.10 which i can't find anywhere, any hints? --snip from ChangeLog-- Asterisk 1.0.10 -- chan_local-- In releases 1.0.8 and 1.0.9, the Local channels that are created would not be masqueraded into the new channel type. This has now been fixed. -- chan_sip -- The 'insecure' options have been changed to support matching peersby IP only, not requiring authentication on incoming invites, or both. Before, to not require authentication on incoming invites also required matching peers based on IP only. -- chan_zap-- Before, call waiting could occur during the initial ringing on the line. This has now been fixed. -- app_disa-- We will now not set the accountcode if one is not supplied. -- app_meetme-- If the first caller into a conference hangs up while being prompted for the conference pin number, the conference will no longer be held open. -- app_userevent-- Events created with this application were indicated as a "call" event instead of a "user" event. This made the "user" event permissions not work correctly. -- app_voicemail-- When using the externpass option for voicemail, the password will be immediately updated in memory as well, instead of having to wait for the next time the configuration is reloaded. -- app_zapras-- We now ensure buffer policy is restored after RAS is done with a channel. This could cause audio problems on the channel after zapras is done with it. -- res_agi-- We now unmask the SIGHUP signal before executing an AGI script. This fixes problems where some AGI scripts would continue running long after the call is over. -- extensions-- A potential crash has been fixed when calling LEN() to get the length of a string that was 80 characters or larger. -- logger-- The Asterisk logger will automatically detect when a log file needs to be rotated. However, this feature could put Asterisk in a nasty loop that would result in a crash. -- general-- Added man pages for astgenkey, autosupport, and safe_asterisk --end of snip Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
