[asterisk-dev] Re: asterisk-dev Digest, Vol 18, Issue 62
Brian Bell
networkdesign at shaw.ca
Fri Jan 20 14:25:54 MST 2006
please stop sending me these emails..brian bellAt 12:53 PM 1/20/2006,
you wrote:
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>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.
>
> _______________________________________________
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>------------------------------
>
>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
>
>
>
>
>
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>------------------------------
>
>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-----
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>
>------------------------------
>
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>End of asterisk-dev Digest, Vol 18, Issue 62
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