Send asterisk-dev mailing list submissions to
asterisk-dev@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
[EMAIL PROTECTED]
You can reach the person managing the list at
[EMAIL PROTECTED]
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" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Dev] Asterisk 1.2.2 Released!
To: "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
To: "Asterisk Developers Mailing List" <asterisk-dev@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 <[EMAIL PROTECTED]>
Subject: [asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 62
To: asterisk-dev@lists.digium.com
Message-ID:
<[EMAIL PROTECTED]>
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@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
> [EMAIL PROTECTED]
>
>You can reach the person managing the list at
> [EMAIL PROTECTED]
>
>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 ([EMAIL PROTECTED])
> 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 ([EMAIL PROTECTED])
> 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: [EMAIL PROTECTED]
>Subject: [asterisk-dev] bounty update $5000.00 - Asterisk bounty PRI
> 2B channel transfer for NI2 PRI line
>To: asterisk-dev@lists.digium.com
>Message-ID: <[EMAIL PROTECTED]>
>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 [EMAIL PROTECTED]
>
>
>
>------------------------------
>
>Message: 2
>Date: Fri, 20 Jan 2006 12:35:01 -0600
>From: Steven Critchfield <[EMAIL PROTECTED]>
>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@lists.digium.com>,
> [EMAIL PROTECTED]
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain
>
>On Fri, 2006-01-20 at 13:14 -0500, [EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>
>
>
>
>------------------------------
>
>Message: 3
>Date: Fri, 20 Jan 2006 10:39:06 -0800 (PST)
>From: "North Antara" <[EMAIL PROTECTED]>
>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@lists.digium.com>
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain;charset=iso-8859-1
>
> > On Fri, 2006-01-20 at 13:14 -0500, [EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>
> >
>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: [EMAIL PROTECTED]
>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@lists.digium.com>
>Cc: [EMAIL PROTECTED]
>Message-ID:
> <[EMAIL PROTECTED]>
>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" <[EMAIL PROTECTED]>
>Subject: RE: [asterisk-dev] bounty update $5000.00 - Asterisk bounty
> PRI 2Bchannel transfer for NI2 PRI line
>To: "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com>
>Message-ID:
>
<[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="utf-8"
>
>Post to the biz list or here www.asteriskhelpdesk.com
><http://www.asteriskhelpdesk.com>
>
>
>
> -----Original Message-----
> From: [EMAIL PROTECTED]
> Sent: Fri 1/20/2006 1:14 PM
> To: asterisk-dev@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 [EMAIL PROTECTED]
>
> _______________________________________________
> --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
>Type: application/ms-tnef
>Size: 5314 bytes
>Desc: not available
>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" <[EMAIL PROTECTED]>
>Subject: [asterisk-dev] how to enable app_queue inband call progress
> to caller
>To: "'Asterisk Developers Mailing List'"
> <asterisk-dev@lists.digium.com>
>Message-ID: <[EMAIL PROTECTED]>
>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 <[EMAIL PROTECTED]>
>Subject: [asterisk-dev] ztdummy question
>To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com>
>Message-ID: <[EMAIL PROTECTED]>
>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
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.375 / Virus Database: 267.14.21/236 - Release Date: 1/20/2006
------------------------------
Message: 3
Date: Fri, 20 Jan 2006 17:02:48 -0600
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-dev] ztdummy question
To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Dev] Asterisk 1.2.2 Released!
To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-dev] Re: Bugs that Need Your Input!
To: "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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" <[EMAIL PROTECTED]>
Subject: Re: [asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 62
To: "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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
>> [EMAIL PROTECTED]
>>
> 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 <[EMAIL PROTECTED]>
Subject: [asterisk-dev] Asterisk Development and Release Cycle
To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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
********************************************
--
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
