[asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 63
Brian Bell
networkdesign at shaw.ca
Fri Jan 20 16:43:04 MST 2006
please refrain from sending me ANY emails in the future..thank you,
brian bellAt 03:40 PM 1/20/2006, you wrote:
>Send asterisk-dev mailing list submissions to
> asterisk-dev at lists.digium.com
>
>To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>or, via email, send a message with subject or body 'help' to
> asterisk-dev-request at lists.digium.com
>
>You can reach the person managing the list at
> asterisk-dev-owner at lists.digium.com
>
>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of asterisk-dev digest..."
>
>
>Today's Topics:
>
> 1. Re: Asterisk 1.2.2 Released! (Aryanto Rachmad)
> 2. Re: asterisk-dev Digest, Vol 18, Issue 62 (Brian Bell)
> 3. Re: ztdummy question (Kevin P. Fleming)
> 4. Re: Asterisk 1.2.2 Released! (Kevin P. Fleming)
> 5. RE: Re: Bugs that Need Your Input! (Dan Austin)
> 6. Re: Re: asterisk-dev Digest, Vol 18, Issue 62 (North Antara)
> 7. Asterisk Development and Release Cycle (Asterisk Development Team)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Fri, 20 Jan 2006 22:27:00 +0100
>From: "Aryanto Rachmad" <aryanto.rachmad at chello.at>
>Subject: Re: [Asterisk-Dev] Asterisk 1.2.2 Released!
>To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>Message-ID: <0df701c61e08$3fd827a0$0e66970a at ds.mot.com>
>Content-Type: text/plain; charset="iso-8859-1"
>
>Hello All,
>
>I have a question which probably sounds silly.
>
>I am using Asterisk SVN-branch-1.2-r8140 built. To be honest I am
>still confused with this SVN things. The main reason I use it is
>that I think I can get the bugs fixed by updating it regularly,
>which I previously could not do that using Asterisk 1.2.1 built. I
>hope I am right.
>
>My question is when I update my current version, will I get into the
>same level of codes as tags/1.2.2 or tags/1.2.2-netsec? Or should I
>change the branch from branches/1.2 to branches/1.2-netsec?
>
>Cheers,
>
>Anto
>
>----- Original Message -----
>From: "Asterisk Development Team" <asteriskteam at digium.com>
>To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>Sent: Thursday, January 19, 2006 1:05 AM
>Subject: [Asterisk-Dev] Asterisk 1.2.2 Released!
>
>
> > Greetings everyone!
> >
> > The 1.2.2 versions of Asterisk, Zaptel, and Libpri have now been
> > released. The source tarballs are available for download on
> > ftp.digium.com. For details about what has changed, see the ChangeLog
> > for Asterisk, Zaptel, or Libpri.
> >
> > We are also excited to announce the release of a special version of
> > Asterisk 1.2.2, called Asterisk-NetSec. It includes some very exciting
> > features not available in any other version of Asterisk, or even any
> > other related product! Please view the appropriate README and ChangeLog
> > for more details.
> >
> > Asterisk-addons and Asterisk-sounds will remain at version 1.2.1.
> > Previously, all packages were updated to reflect a matching version
> > number, even if no changes have been made. From now on, releases will
> > only be made when changes have actually been made. Even if version
> > numbers do not match, it is safe to use all of these releases together,
> > as long as all of them are the latest version available.
> >
> > Thank you!
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
>
>
>
>------------------------------
>
>Message: 2
>Date: Fri, 20 Jan 2006 13:25:54 -0800
>From: Brian Bell <networkdesign at shaw.ca>
>Subject: [asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 62
>To: asterisk-dev at lists.digium.com
>Message-ID:
> <6.2.3.4.0.20060120132530.02126438 at shawmail.vc.shawcable.net>
>Content-Type: text/plain; charset=us-ascii; format=flowed;
> x-avg-checked=avg-ok-30F828
>
>please stop sending me these emails..brian bellAt 12:53 PM 1/20/2006,
>you wrote:
>
> >Send asterisk-dev mailing list submissions to
> > asterisk-dev at lists.digium.com
> >
> >To subscribe or unsubscribe via the World Wide Web, visit
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >or, via email, send a message with subject or body 'help' to
> > asterisk-dev-request at lists.digium.com
> >
> >You can reach the person managing the list at
> > asterisk-dev-owner at lists.digium.com
> >
> >When replying, please edit your Subject line so it is more specific
> >than "Re: Contents of asterisk-dev digest..."
> >
> >
> >Today's Topics:
> >
> > 1. bounty update $5000.00 - Asterisk bounty PRI 2B channel
> > transfer for NI2 PRI line (voip3 at nibble.net)
> > 2. Re: bounty update $5000.00 - Asterisk bounty PRI 2B channel
> > transfer for NI2 PRI line (Steven Critchfield)
> > 3. Re: bounty update $5000.00 - Asterisk bounty PRI 2B channel
> > transfer for NI2 PRI line (North Antara)
> > 4. Re: bounty update $5000.00 - Asterisk bounty PRI 2B channel
> > transfer for NI2 PRI line (alex at pilosoft.com)
> > 5. RE: bounty update $5000.00 - Asterisk bounty PRI 2Bchannel
> > transfer for NI2 PRI line (Steve Totaro)
> > 6. how to enable app_queue inband call progress to caller
> > (Raymond Chen)
> > 7. ztdummy question (Sean Cook)
> >
> >
> >----------------------------------------------------------------------
> >
> >Message: 1
> >Date: Fri, 20 Jan 2006 13:14:35 -0500 (EST)
> >From: voip3 at nibble.net
> >Subject: [asterisk-dev] bounty update $5000.00 - Asterisk bounty PRI
> > 2B channel transfer for NI2 PRI line
> >To: asterisk-dev at lists.digium.com
> >Message-ID: <49644.64.74.225.131.1137780875.squirrel at 64.74.225.131>
> >Content-Type: text/plain;charset=iso-8859-1
> >
> >Maintainer: Express Line
> >Date opened: January 17, 2006
> >Status: Open
> >Value of bounty: $5000.00
> >Licensing for code: We retain intellectual rights to the underlying source
> >code.
> >
> >We need Asterisk (stable version) to be able to perform a 2B channel
> >transfer for a NI2 B8ZS PRI line. We can't use a channelized T1 at the
> >time for our work. This feature is commonly called a call transfer on
> >analog phone lines. On an analog phone line, the incoming call is
> >answered, a hook-flash is performed to get stutter dialtone, the telephone
> >number to transfer to is dialed, and finally the caller hangs up the phone
> >to complete the call transfer. This frees up the analog phone line to
> >process another call and the central office handles the transfered call.
> >This transfer feature can be done with a channelized winkstart T1, and is
> >possible on a PRI. On a PRI, this feature is called a 2B Channel Transfer.
> >Contact us at voip3 at nibble.net.
> >
> >
> >
> >------------------------------
> >
> >Message: 2
> >Date: Fri, 20 Jan 2006 12:35:01 -0600
> >From: Steven Critchfield <critch at basesys.com>
> >Subject: Re: [asterisk-dev] bounty update $5000.00 - Asterisk bounty
> > PRI 2B channel transfer for NI2 PRI line
> >To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>,
> > voip3 at nibble.net
> >Message-ID: <1137782101.26355.16.camel at localhost.localdomain>
> >Content-Type: text/plain
> >
> >On Fri, 2006-01-20 at 13:14 -0500, voip3 at nibble.net wrote:
> > > Maintainer: Express Line
> > > Date opened: January 17, 2006
> > > Status: Open
> > > Value of bounty: $5000.00
> > > Licensing for code: We retain intellectual rights to the
> underlying source
> > > code.
> >
> >please don't spam this list. So far you have only posted messages that
> >are primarily offtopic since they didn't actually pertain to the code of
> >asterisk but rather solicitation of someone to do the work.
> >
> >I don't want to discourage the use of bounties, but rather I want to
> >encourage better mailinglist ettiquette.
> >--
> >Steven Critchfield <critch at basesys.com>
> >
> >
> >
> >------------------------------
> >
> >Message: 3
> >Date: Fri, 20 Jan 2006 10:39:06 -0800 (PST)
> >From: "North Antara" <north at ntbox.com>
> >Subject: Re: [asterisk-dev] bounty update $5000.00 - Asterisk bounty
> > PRI 2B channel transfer for NI2 PRI line
> >To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> >Message-ID: <28194.159.37.7.93.1137782346.squirrel at 159.37.7.93>
> >Content-Type: text/plain;charset=iso-8859-1
> >
> > > On Fri, 2006-01-20 at 13:14 -0500, voip3 at nibble.net wrote:
> > >> Maintainer: Express Line
> > >> Date opened: January 17, 2006
> > >> Status: Open
> > >> Value of bounty: $5000.00
> > >> Licensing for code: We retain intellectual rights to the underlying
> > >> source
> > >> code.
> > >
> > > please don't spam this list. So far you have only posted messages that
> > > are primarily offtopic since they didn't actually pertain to the code of
> > > asterisk but rather solicitation of someone to do the work.
> > >
> > > I don't want to discourage the use of bounties, but rather I want to
> > > encourage better mailinglist ettiquette.
> > > --
> > > Steven Critchfield <critch at basesys.com>
> > >
> >Indeed. In fact, one should probably be posting these messages to the
> >-biz mailing list instead. That's what it's for, right?
> >
> >
> >------------------------------
> >
> >Message: 4
> >Date: Fri, 20 Jan 2006 14:46:55 -0500 (EST)
> >From: alex at pilosoft.com
> >Subject: Re: [asterisk-dev] bounty update $5000.00 - Asterisk bounty
> > PRI 2B channel transfer for NI2 PRI line
> >To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> >Cc: voip3 at nibble.net
> >Message-ID:
> > <Pine.LNX.4.44.0601201446300.15581-100000 at bawx.pilosoft.com>
> >Content-Type: TEXT/PLAIN; charset=US-ASCII
> >
> >On Fri, 20 Jan 2006, Steven Critchfield wrote:
> >
> > > please don't spam this list. So far you have only posted messages that
> > > are primarily offtopic since they didn't actually pertain to the code of
> > > asterisk but rather solicitation of someone to do the work.
> >Indeed. The proper forum would be -biz list. (or -users, or voip wiki)
> >
> >-alex
> >
> >
> >
> >------------------------------
> >
> >Message: 5
> >Date: Fri, 20 Jan 2006 13:43:44 -0500
> >From: "Steve Totaro" <stotaro at totarotechnologies.com>
> >Subject: RE: [asterisk-dev] bounty update $5000.00 - Asterisk bounty
> > PRI 2Bchannel transfer for NI2 PRI line
> >To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> >Message-ID:
> >
> <BE9DDFB4003EB1499F18B605C4A0E54DDD32 at steves.first-notification.com>
> >Content-Type: text/plain; charset="utf-8"
> >
> >Post to the biz list or here www.asteriskhelpdesk.com
> ><http://www.asteriskhelpdesk.com>
> >
> >
> >
> > -----Original Message-----
> > From: voip3 at nibble.net
> > Sent: Fri 1/20/2006 1:14 PM
> > To: asterisk-dev at lists.digium.com
> > Cc:
> > Subject: [asterisk-dev] bounty update $5000.00 - Asterisk bounty
> >PRI 2Bchannel transfer for NI2 PRI line
> >
> >
> >
> > Maintainer: Express Line
> > Date opened: January 17, 2006
> > Status: Open
> > Value of bounty: $5000.00
> > Licensing for code: We retain intellectual rights to the
> >underlying source
> > code.
> >
> > We need Asterisk (stable version) to be able to perform a 2B
> >channel
> > transfer for a NI2 B8ZS PRI line. We can't use a channelized T1
> >at the
> > time for our work. This feature is commonly called a call
> >transfer on
> > analog phone lines. On an analog phone line, the incoming call
> >is
> > answered, a hook-flash is performed to get stutter dialtone, the
> >telephone
> > number to transfer to is dialed, and finally the caller hangs up
> >the phone
> > to complete the call transfer. This frees up the analog phone
> >line to
> > process another call and the central office handles the
> >transfered call.
> > This transfer feature can be done with a channelized winkstart
> >T1, and is
> > possible on a PRI. On a PRI, this feature is called a 2B Channel
> >Transfer.
> > Contact us at voip3 at nibble.net.
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> >
> >-------------- next part --------------
> >A non-text attachment was scrubbed...
> >Name: not available
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> >Url :
> >http://lists.digium.com/pipermail/asterisk-dev/attachments/20060120
> /9649d8af/attachment-0001.bin
> >
> >------------------------------
> >
> >Message: 6
> >Date: Sat, 21 Jan 2006 02:42:23 -0800
> >From: "Raymond Chen" <rchen at broadz.com>
> >Subject: [asterisk-dev] how to enable app_queue inband call progress
> > to caller
> >To: "'Asterisk Developers Mailing List'"
> > <asterisk-dev at lists.digium.com>
> >Message-ID: <20060120184225.E37ACCBD8 at lists.digium.com>
> >Content-Type: text/plain; charset="us-ascii"
> >
> >
> >
> >Hi all,
> >
> >
> >
> >I would like to have the caller in app_queue to hear inband call progress
> >ringing instead of music on hold. Using options 'r' will enforce false
> >ringtone which is not what I want, I want the app_dial call progress forward
> >to app_queue instead. Can anyone give me some hints on how to make this
> >happen?
> >
> >
> >
> >Thanks
> >
> >
> >
> >Ray
> >
> >
> >
> >
> >
> >-------------- next part --------------
> >An HTML attachment was scrubbed...
> >URL:
> >http://lists.digium.com/pipermail/asterisk-dev/attachments/20060120
> /93da32fd/attachment-0001.htm
> >
> >------------------------------
> >
> >Message: 7
> >Date: Fri, 20 Jan 2006 15:52:25 -0500
> >From: Sean Cook <scook at kinex.net>
> >Subject: [asterisk-dev] ztdummy question
> >To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> >Message-ID: <43D14D89.8010506 at kinex.net>
> >Content-Type: text/plain; charset=ISO-8859-1
> >
> >-----BEGIN PGP SIGNED MESSAGE-----
> >Hash: SHA1
> >
> >with the changes to the ztdummy to rely on rtc vs jiffies, I am now
> >forced to increase the interrupt frequency time by roughly 10x the
> >frequency recommended for the SMP processing systems.
> >
> >Is this wise? Or would it be better to not assume that the CONFIG_HZ ==
> >1000 and base the calculation on what ever HZ is set to?
> >
> >Maybe for me the solution is to not rely on ztdummy at all ( i will be
> >using a te210P in this server ).
> >
> >If I am way off on this question, i apologize... it just seem strange.
> >
> >Sean
> >-----BEGIN PGP SIGNATURE-----
> >Version: GnuPG v1.4.2 (MingW32)
> >Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
> >
> >iD8DBQFD0U2Jy9wPyZpnL2URAn6lAJ4rKPI9u8K0wEqVNFrZqpgU+1agdgCaA9R8
> >zD17R6tA/33wuot8rQE1s3s=
> >=is5A
> >-----END PGP SIGNATURE-----
> >
> >
> >------------------------------
> >
> >_______________________________________________
> >--Bandwidth and Colocation provided by Easynews.com --
> >
> >asterisk-dev mailing list
> >To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> >
> >End of asterisk-dev Digest, Vol 18, Issue 62
> >********************************************
> >
> >
> >
> >--
> >No virus found in this incoming message.
> >Checked by AVG Anti-Virus.
> >Version: 7.1.375 / Virus Database: 267.14.21/236 - Release Date: 1/20/2006
>
>
>--
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>Checked by AVG Anti-Virus.
>Version: 7.1.375 / Virus Database: 267.14.21/236 - Release Date: 1/20/2006
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>
>
>
>------------------------------
>
>Message: 3
>Date: Fri, 20 Jan 2006 17:02:48 -0600
>From: "Kevin P. Fleming" <kpfleming at digium.com>
>Subject: Re: [asterisk-dev] ztdummy question
>To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>Message-ID: <43D16C18.3080900 at digium.com>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>Sean Cook wrote:
>
> > with the changes to the ztdummy to rely on rtc vs jiffies, I am now
> > forced to increase the interrupt frequency time by roughly 10x the
> > frequency recommended for the SMP processing systems.
>
>Isn't that backwards? Using the RTC means we are _not_ relying on the
>frequency of jiffies at all.
>
>
>------------------------------
>
>Message: 4
>Date: Fri, 20 Jan 2006 17:03:54 -0600
>From: "Kevin P. Fleming" <kpfleming at digium.com>
>Subject: Re: [Asterisk-Dev] Asterisk 1.2.2 Released!
>To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>Message-ID: <43D16C5A.1080308 at digium.com>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>Aryanto Rachmad wrote:
>
> > I have a question which probably sounds silly.
> >
> > I am using Asterisk SVN-branch-1.2-r8140 built. To be honest I am
> still confused with this SVN things. The main reason I use it is
> that I think I can get the bugs fixed by updating it regularly,
> which I previously could not do that using Asterisk 1.2.1 built. I
> hope I am right.
> >
> > My question is when I update my current version, will I get into
> the same level of codes as tags/1.2.2 or tags/1.2.2-netsec? Or
> should I change the branch from branches/1.2 to branches/1.2-netsec?
>
>No, using branches/1.2 will get you all the current 1.2.x code (except
>for the netsec stuff), even whatever has been committed since the last
>tarball release was made.
>
>
>------------------------------
>
>Message: 5
>Date: Fri, 20 Jan 2006 15:04:37 -0800
>From: "Dan Austin" <Dan_Austin at Phoenix.com>
>Subject: RE: [asterisk-dev] Re: Bugs that Need Your Input!
>To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>Message-ID:
> <B0CF4196F21DC0448367514774331AB7E83A22 at scl-exch2k3.phoenix.com>
>Content-Type: text/plain; charset="us-ascii"
>
>Don't over estimate my familiarity with the code :-)
>
>I think I see something odd in channel.c, in code not touched
>by this patch. In ast_activate_generator there is a call to
>ast_settimeout(chan, 160, generator_force, chan);
>
>Now it might be just me, but should the 1st and 4th parameters
>be chan?
>
>Dan
>
>-----Original Message-----
>From: asterisk-dev-bounces at lists.digium.com
>[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Koopmann,
>Jan-Peter
>Sent: Friday, January 20, 2006 9:44 AM
>To: Asterisk Developers Mailing List
>Subject: RE: [asterisk-dev] Re: Bugs that Need Your Input!
>
>Oh and IAX jitter buffer seems to bet broken as well with this patch.
>Might have a look at this too.
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>asterisk-dev mailing list
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>------------------------------
>
>Message: 6
>Date: Fri, 20 Jan 2006 15:08:00 -0800 (PST)
>From: "North Antara" <north at ntbox.com>
>Subject: Re: [asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 62
>To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>Message-ID: <33009.198.74.20.118.1137798480.squirrel at 198.74.20.118>
>Content-Type: text/plain;charset=iso-8859-1
>
> >>To subscribe or unsubscribe via the World Wide Web, visit
> >> http://lists.digium.com/mailman/listinfo/asterisk-dev
> >>or, via email, send a message with subject or body 'help' to
> >> asterisk-dev-request at lists.digium.com
> >>
> > please stop sending me these emails..brian bellAt 12:53 PM 1/20/2006,
> > you wrote:
> >
>If you don't want these messages...unsubscribe from the list. It very
>clearly tells you how.
>
>
>------------------------------
>
>Message: 7
>Date: Fri, 20 Jan 2006 17:40:04 -0600
>From: Asterisk Development Team <asteriskteam at digium.com>
>Subject: [asterisk-dev] Asterisk Development and Release Cycle
>To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>Message-ID: <43D174D4.4040208 at digium.com>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>Asterisk 1.2 was released over 1 year after Asterisk 1.0, which resulted
>in many users trying to run the development version of Asterisk in a
>production capacity so that they could take advantage of the new
>features that had been added. This produced a flurry of extraneous bug
>reports and caused extra work for the developers as they could not work
>on changes that would actually cause disruption of the development tree.
>
>In an effort to combat this problem, and to give the community a more
>predictable release cycle, the process is being organized so that such a
>long time between releases will never happen again.
>
>Beginning in January of 2006, we will produce new major Asterisk
>releases on a six month cycle.
>
>The development cycle will be organized in this fashion:
>
>MONTHS 1 - 3
>
>The first three months of the development cycle are when the development
>branch will be changed most drastically. The tree is open to large
>architectural changes as well as new feature enhancements and bug fixes.
>
>MONTHS 4 - 5
>
>For the next two months, the development branch will no longer receive
>architectural changes. New features that are ready to be merged will
>still be accepted at this point.
>
>MONTH 6
>
>The last month is reserved for beta testing. No more features will be
>accepted for the upcoming release. Beta releases will be made on a
>weekly cycle, culminating in one (or two) release candidate releases
>just before the final release.
>
>Asterisk 1.4 is scheduled to be released in the beginning of July, 2006.
>Once the release is made, a branch will be created. This branch will
>then receive maintenance for bug fixes only. At that point, the
>development cycle will start over to prepare for the next major release
>of Asterisk, scheduled for January of 2007.
>
>The Asterisk Development Team
>
>
>
>
>------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>asterisk-dev mailing list
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>End of asterisk-dev Digest, Vol 18, Issue 63
>********************************************
>
>
>
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