From ratguy at insightbb.com  Mon Oct  1 09:07:32 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Mon, 1 Oct 2007 10:07:32 -0400
Subject: [VoIP] Question on connecting old dial telephone
References: <A691C3685A152E4AAC9608B8D8E1131668650A@godzilla.hickory.vyger.net>
	<20071001005812.GI13006@5.7.5.5.6.6.6.6.0.2.1.e164.arpa>
Message-ID: <001401c80434$68e1d420$6700a8c0@bluegrasspals.com>

Hi,

    I think RP under Asterisk would be cool! Especially if two Asterisk
systems could RP to each other, even though that's totally unnecessary. A
question though? I'm wondering, would you have Asterisk pulse as quickly as
possible, or, for old times' sake, implement panel-like timing for each
digit? Thanks.
Jayson

----- Original Message ----- 
From: "Duncan Smith" <duncan.b.smith at gmail.com>
To: "CNET List" <voip at ckts.info>
Cc: "Phil Braverman" <braverma at hotmail.com>
Sent: Sunday, September 30, 2007 8:58 PM
Subject: Re: [VoIP] Question on connecting old dial telephone


> Phil wrote:
> > My name is Phil Braverman and I am very interested in old telephone
> > technology.  I was reading your Website and I found that you can
> > call into the Seattle Museum of communications switches that are
> > connected to the CNET network that you all started a few years ago
> > by using asterisk IP tandom switches.
> >
> > My question is that I have a PAP2 unlocked VOIP adapter taht can
> > accept touchtone or dial pulses.  Is it possible for me to hook up
> > my dial phone in San Jose California to the panel switch at the
> > Seattle Museum of Communications using Voip so I would have a
> > number coming out of the panel switch to my phone.  When I remove
> > the reciever of my telephone it would have a dial tone originating
> > from the switch.  The VOIP would be like a tieline from the switch
> > to my telephone.  For example if I was assigned 722-1701, this
> > would be my telephone number.
>
> Hi Phil!  My name is Duncan Smith.  Among other things, I'm the
> volunteer in charge of Asterisk at the Seattle museum.
>
> We don't have our switches online yet; we're waiting for a T1 card to
> be either bought or donated.  (I am very pleased to announce that this
> week I received two RP units for a D4 channel bank!)  Eventually you
> will be able to call into the EM switches when the Museum is open
> (1XB, 5XB, Panel, and perhaps others including Step and CX-100), as
> well as our 3ESS all the time.
>
> I don't think you _really_ want a line to the Panel switch for a
> couple of reasons:
>
> 1. It's rather crotchety.  Sometimes verticals (what's the proper
>    term, anyway?) get stuck, and have to be manually restored to
>    normal.
>
> 2. We don't run it all the time; around four hours a week is usual.
>
> 3. You won't be able to dial out to CNET, because Panel switches were
>    never fitted for DDD; so far as I know all toll calls were
>    operator-handled.  Thus, a Panel switch can only out-pulse the last
>    five digits of the dialed number.  (Could someone please prove me
>    wrong?)
>
> I don't even think that the Panel switch will be online very soon
> after we get a T1 card!  I'll have to hack up Zaptel to support RP
> signalling, which might take a while.
>
> I understand that you like Panel, though.  I also think it's neat.  :)
>
> The plan is to get some recordings on our Asterisk machine as soon as
> we can, so that you can at least pretend.
>
>
> On Sun, Sep 30, 2007 at 02:23:25PM -0500, Greg Blakely wrote:
> > Phil sent me this inquiry.  I can think of a few ways that might
> > give him most of what he wants, but I don't see how to give him dial
> > tone from the panel office, and, along those lines, I don't see a
> > way for him to listen to the call progress through the panel office
> > as he dials.
>
> The first part is (in my mind) fairly simple; all that's physically
> necessary on the Seattle end is a FXO D4 channel bank card, wired to a
> line presence on the Panel switch.  The simplest solution would be to
> do some sort of pulse conversion in my Asterisk.
>
> Isn't it possible to have Asterisk place a call immediately on
> off-hook?  I don't know if they're using Asterisk, but on the front
> desk at one of my local hospitals is a set which, when picked up, puts
> the caller directly into a queue to speak to an operator.  I would
> think that such a use (as a "hot-line") would be rather common.
>
> -- 
> Duncan Smith  --------\    http://students.washington.edu/f/    /---
>   ()  ascii ribbon     \--- Signed/encrypted mail preferred ---/
>   /\    campaign    [ against html mail ]  [ support open formats ]
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/

From dfroula at sbcglobal.net  Tue Oct  2 08:47:00 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 2 Oct 2007 06:47:00 -0700 (PDT)
Subject: [VoIP] Tandem Stacking over Asterisk
In-Reply-To: <863814.38251.qm@web83215.mail.mud.yahoo.com>
Message-ID: <703492.99021.qm@web83209.mail.mud.yahoo.com>

I changed the setting of "toneduration" in
/etc/zapata.conf from 68 ms. to 100ms. This increases
the duration of the MF tones on the SF trunk group
during call setup, but also increases the length of
regenerated DTMF tones sent down the line. This
enhances the "flash forward" effect on a tandem stack
by extending the propagation time through each link.

The reason one gets a hook-flash effect when dialing
DTMF through the stack is that The ATA receives the
DTMF digit, converts it to a SIP message, Zaptel
converts back to audio on the first SF trunk link,
Zaptel converts back to an internal message on the
receiving end, then back to analog DTMF on the next
link and so on. This adds a 100 ms. delay for each
link on each flash of a DTMF digit down the stack.

Don

--- Donald Froula <dfroula at sbcglobal.net> wrote:

> This actually sound pretty amazing while listening
> on
> the far end, if one dial an extension on the last
> loop. When the originating end goes on-hook, you can
> hear the string of kercheeps as the call
> disconnects,
> one for each link. If the originating end taps a
> touch
> tone digit, you can hear a series of clicks as the
> tone propagates down the stack. Sounds pretty much
> like the Classic Stacking recording!
> 
> If you want to try, here's how:
> 
> - Dial my ProjectMF trunk on 1-762-2601 (or
> 1-762-2602, skipping the net two steps, for those
> without a blue box)
> - Blow off the ringing with 2600
> - After the wink, dial KP-2602-ST
> - Wait for the dial tone
> - Dial 2602 with the DTMF pad of the phone
> - Wait for another dial tone
> - Dial 2602 with the DTMF pad
> - (Repeat X times, up to 23 times, as desired)
> - On the last dial tone, dial a number on CNET with
> 011+CC_Number, again with the DTMF pad
> - Answer the ringing phone. You are now talking to
> yourself over X number of SF trunk links
> - Hit a touch tone digit onthe originating phone.
> Listen to it flash down the stack.
> - Hang up. Hear the stack disconnect with a cheep of
> 2600 for each link
> 
> Notice how the connection takes longer to return the
> dial tone on each loop through the stack, as the
> audio
> delay on the forward MF (which you can't hear) gets
> longer and longer.
> 
> Regardless of how you come in. you can blow the call
> off in the forward direction with 2600 and hear the
> stack disconnect on the far side.
> 
> Much fun!
> 
> Don
> 
> PS - For those without MF and 2600, I set up direct
> access to the SF/DISA arrangement on 1-762-2602
> 
> Note, if others are trying this, your available
> trunk
> pool may be less than 24.
> 
> --- Donald Froula <dfroula at sbcglobal.net> wrote:
> 
> > I hit upon an approach to try this trunk stacking
> > thing through my TDM-Over-Ethernet SF/MF trunks. I
> > set
> > up two extensions to go to DISA over the trunks.
> > What
> > I do is dial the extension that goes over the
> trunks
> > (similar to my 1-762-2601 number), but instead of
> > programming the number that gives ringback in the
> > Dial
> > command, I call another extension with the DISA
> > application that returns a dial tone.
> > 
> > 2602 => 1,Dial(ZAP/g1/9999)
> > 
> > 9999 => 1,DISA(password|from-internal)
> > 
> > I then dial the same number again (2602), which
> > seizes
> > the next line and gives a DISA dial tone again.
> This
> > is all via DTMF, so, yes, I am cheating. However,
> > the
> > call setup on each loop through the trunk group is
> > via
> > 2600 and MF.
> > 
> > I saturated all 24 lines of the trunk group this
> > way,
> > dialing one of my Evan Doorbell recordings on the
> > last
> > trunk. The audio was looping through all 24
> > SF-controlled trunks. Both sides of the connection
> > being on the same switch, I only noticed slight
> > degradation in the audio. I blew off the call with
> a
> > chirp of 2600. This sounded exactly like blowing
> off
> > a
> > call on a single trunk, except for the 24
> disconnect
> > mrssages that flew by on the Asterisk console.
> > 
> > Now, if the looping could be done through another
> > ProjectMF switch, it might sound interesting.
> > 
> > Don
> > 
> > --- Donald Froula <dfroula at sbcglobal.net> wrote:
> > 
> > > Mark, you asked about whether or not a 2600
> audio
> > > blip
> > > might get propagated through the system for 2600
> > > flashes less than rxwink in length. Actually,
> they
> > > would. The 2600 notch filter DSP code takes a
> > > millisecond or so to kick in and out, creating
> an
> > > audible "blip" both on 2600 application and
> > removal.
> > > Although this has no effect on supervision,
> being
> > > purely an audio effect, it might provided
> > something
> > > like the old sound with the latency delay
> through
> > > VOIP.
> > > 
> > > Don
> > > 
> > > --- Donald Froula <dfroula at sbcglobal.net> wrote:
> > > 
> > > > Hi, Mark. Good to hear from you "down under",
> > > > 
> > > > I corrected the trunk-hang problem by a patch
> I
> > > > added
> > > > to Asterisk that restored 2600 supervision
> based
> > > > upon
> > > > steady-state 2600. It sounds odd, but seizure
> of
> > > the
> > > > trunk with the original set of patches was
> based
> > > on
> > > > REMOVAL of steady 2600. Applying steady 2600
> > after
> > > > seizure did not disconnect the call. This was
> to
> > > > allow
> > > > the user to 'box another call after the
> previous
> > > far
> > > > end call disconnected. Evan Doorbell calls
> this
> > a
> > > > "goody" trunk in his recordings. I
> accomplished
> > > the
> > > > "goody" arrangement after my patch by changing
> > > some
> > > > settings in /etc/zaptel.conf. Short answer:
> The
> > > > trunks
> > > > now disconnect correctly from stacking
> attempts.
> > > > 
> > > > It is true that the timing of the minimum 2600
> > > tone
> > > > duration to seize and also the length of the
> > > reverse
> > > > wink can both be set by configuration settings
> > in
> > > > /etc/asterisk/zapata.conf. I have mine set to
> > > > rxwink=50 (requires a minumum 50 ms. of 2600
> to
> > > > seize,
> > > > for talkoff protection) and wink=250 (sets
> > > > acknowledgement wink to 250 ms.). The
> "wink=250"
> > > > setting is purely cosmetic, as the near-end
> > trunks
> > > > are
> > > > using immediate dialing, and is just for
> realism
> > > and
> > > > an indication to the 'box user that the MF
> > > receiver
> > > > has been successfully attached to the far end.
> > > > 
> > > > What you are asking is whether or not 2600
> > bursts
> > > > shorter than the setting of rxwink will cause
> a
> > > > momentary application of supervision that will
> > not
> > > > disconnect the call. This was called "ring
> > > forward"
> > > > or
> > > > "operator flash" in the old days.
> Unfortunately,
> > > > ProjectMF completely ignores any forward
> > > application
> 
=== message truncated ===


From dfroula at sbcglobal.net  Tue Oct  2 15:23:54 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 2 Oct 2007 13:23:54 -0700 (PDT)
Subject: [VoIP] Tandem Stacking over Asterisk
In-Reply-To: <703492.99021.qm@web83209.mail.mud.yahoo.com>
Message-ID: <206423.89966.qm@web83215.mail.mud.yahoo.com>

I also changed rxwink=150. It now takes 150 ms. of
2600 to seize the trunk. This also lengthens the
forward disconnect time on each link in the forward
stack, so you can hear the individual cheeps of 2600
as the stack is torn down. It's slow enough you can
count the links.

Don

--- Donald Froula <dfroula at sbcglobal.net> wrote:

> I changed the setting of "toneduration" in
> /etc/zapata.conf from 68 ms. to 100ms. This
> increases
> the duration of the MF tones on the SF trunk group
> during call setup, but also increases the length of
> regenerated DTMF tones sent down the line. This
> enhances the "flash forward" effect on a tandem
> stack
> by extending the propagation time through each link.
> 
> The reason one gets a hook-flash effect when dialing
> DTMF through the stack is that The ATA receives the
> DTMF digit, converts it to a SIP message, Zaptel
> converts back to audio on the first SF trunk link,
> Zaptel converts back to an internal message on the
> receiving end, then back to analog DTMF on the next
> link and so on. This adds a 100 ms. delay for each
> link on each flash of a DTMF digit down the stack.
> 
> Don
> 
> --- Donald Froula <dfroula at sbcglobal.net> wrote:
> 
> > This actually sound pretty amazing while listening
> > on
> > the far end, if one dial an extension on the last
> > loop. When the originating end goes on-hook, you
> can
> > hear the string of kercheeps as the call
> > disconnects,
> > one for each link. If the originating end taps a
> > touch
> > tone digit, you can hear a series of clicks as the
> > tone propagates down the stack. Sounds pretty much
> > like the Classic Stacking recording!
> > 
> > If you want to try, here's how:
> > 
> > - Dial my ProjectMF trunk on 1-762-2601 (or
> > 1-762-2602, skipping the net two steps, for those
> > without a blue box)
> > - Blow off the ringing with 2600
> > - After the wink, dial KP-2602-ST
> > - Wait for the dial tone
> > - Dial 2602 with the DTMF pad of the phone
> > - Wait for another dial tone
> > - Dial 2602 with the DTMF pad
> > - (Repeat X times, up to 23 times, as desired)
> > - On the last dial tone, dial a number on CNET
> with
> > 011+CC_Number, again with the DTMF pad
> > - Answer the ringing phone. You are now talking to
> > yourself over X number of SF trunk links
> > - Hit a touch tone digit onthe originating phone.
> > Listen to it flash down the stack.
> > - Hang up. Hear the stack disconnect with a cheep
> of
> > 2600 for each link
> > 
> > Notice how the connection takes longer to return
> the
> > dial tone on each loop through the stack, as the
> > audio
> > delay on the forward MF (which you can't hear)
> gets
> > longer and longer.
> > 
> > Regardless of how you come in. you can blow the
> call
> > off in the forward direction with 2600 and hear
> the
> > stack disconnect on the far side.
> > 
> > Much fun!
> > 
> > Don
> > 
> > PS - For those without MF and 2600, I set up
> direct
> > access to the SF/DISA arrangement on 1-762-2602
> > 
> > Note, if others are trying this, your available
> > trunk
> > pool may be less than 24.
> > 
> > --- Donald Froula <dfroula at sbcglobal.net> wrote:
> > 
> > > I hit upon an approach to try this trunk
> stacking
> > > thing through my TDM-Over-Ethernet SF/MF trunks.
> I
> > > set
> > > up two extensions to go to DISA over the trunks.
> > > What
> > > I do is dial the extension that goes over the
> > trunks
> > > (similar to my 1-762-2601 number), but instead
> of
> > > programming the number that gives ringback in
> the
> > > Dial
> > > command, I call another extension with the DISA
> > > application that returns a dial tone.
> > > 
> > > 2602 => 1,Dial(ZAP/g1/9999)
> > > 
> > > 9999 => 1,DISA(password|from-internal)
> > > 
> > > I then dial the same number again (2602), which
> > > seizes
> > > the next line and gives a DISA dial tone again.
> > This
> > > is all via DTMF, so, yes, I am cheating.
> However,
> > > the
> > > call setup on each loop through the trunk group
> is
> > > via
> > > 2600 and MF.
> > > 
> > > I saturated all 24 lines of the trunk group this
> > > way,
> > > dialing one of my Evan Doorbell recordings on
> the
> > > last
> > > trunk. The audio was looping through all 24
> > > SF-controlled trunks. Both sides of the
> connection
> > > being on the same switch, I only noticed slight
> > > degradation in the audio. I blew off the call
> with
> > a
> > > chirp of 2600. This sounded exactly like blowing
> > off
> > > a
> > > call on a single trunk, except for the 24
> > disconnect
> > > mrssages that flew by on the Asterisk console.
> > > 
> > > Now, if the looping could be done through
> another
> > > ProjectMF switch, it might sound interesting.
> > > 
> > > Don
> > > 
> > > --- Donald Froula <dfroula at sbcglobal.net> wrote:
> > > 
> > > > Mark, you asked about whether or not a 2600
> > audio
> > > > blip
> > > > might get propagated through the system for
> 2600
> > > > flashes less than rxwink in length. Actually,
> > they
> > > > would. The 2600 notch filter DSP code takes a
> > > > millisecond or so to kick in and out, creating
> > an
> > > > audible "blip" both on 2600 application and
> > > removal.
> > > > Although this has no effect on supervision,
> > being
> > > > purely an audio effect, it might provided
> > > something
> > > > like the old sound with the latency delay
> > through
> > > > VOIP.
> > > > 
> > > > Don
> > > > 
> > > > --- Donald Froula <dfroula at sbcglobal.net>
> wrote:
> > > > 
> > > > > Hi, Mark. Good to hear from you "down
> under",
> > > > > 
> > > > > I corrected the trunk-hang problem by a
> patch
> > I
> > > > > added
> > > > > to Asterisk that restored 2600 supervision
> > based
> > > > > upon
> > > > > steady-state 2600. It sounds odd, but
> seizure
> > of
> > > > the
> > > > > trunk with the original set of patches was
> > based
> > > > on
> > > > > REMOVAL of steady 2600. Applying steady 2600
> > > after
> > > > > seizure did not disconnect the call. This
> was
> > to
> > > > > allow
> > > > > the user to 'box another call after the
> > previous
> > > > far
> > > > > end call disconnected. Evan Doorbell calls
> > this
> > > a
> 
=== message truncated ===


From dfroula at sbcglobal.net  Tue Oct  2 18:25:39 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 2 Oct 2007 16:25:39 -0700 (PDT)
Subject: [VoIP] Tandem Stacking over Asterisk
In-Reply-To: <206423.89966.qm@web83215.mail.mud.yahoo.com>
Message-ID: <832222.36130.qm@web83207.mail.mud.yahoo.com>

There is a recording of the new timing parameters
applied to a 24 SF trunk stack at the end of the
recording at 1-762-0171. Use # to fast forward, * to
rewind.

Don

--- Donald Froula <dfroula at sbcglobal.net> wrote:

> I also changed rxwink=150. It now takes 150 ms. of
> 2600 to seize the trunk. This also lengthens the
> forward disconnect time on each link in the forward
> stack, so you can hear the individual cheeps of 2600
> as the stack is torn down. It's slow enough you can
> count the links.
> 
> Don
> 
> --- Donald Froula <dfroula at sbcglobal.net> wrote:
> 
> > I changed the setting of "toneduration" in
> > /etc/zapata.conf from 68 ms. to 100ms. This
> > increases
> > the duration of the MF tones on the SF trunk group
> > during call setup, but also increases the length
> of
> > regenerated DTMF tones sent down the line. This
> > enhances the "flash forward" effect on a tandem
> > stack
> > by extending the propagation time through each
> link.
> > 
> > The reason one gets a hook-flash effect when
> dialing
> > DTMF through the stack is that The ATA receives
> the
> > DTMF digit, converts it to a SIP message, Zaptel
> > converts back to audio on the first SF trunk link,
> > Zaptel converts back to an internal message on the
> > receiving end, then back to analog DTMF on the
> next
> > link and so on. This adds a 100 ms. delay for each
> > link on each flash of a DTMF digit down the stack.
> > 
> > Don
> > 
> > --- Donald Froula <dfroula at sbcglobal.net> wrote:
> > 
> > > This actually sound pretty amazing while
> listening
> > > on
> > > the far end, if one dial an extension on the
> last
> > > loop. When the originating end goes on-hook, you
> > can
> > > hear the string of kercheeps as the call
> > > disconnects,
> > > one for each link. If the originating end taps a
> > > touch
> > > tone digit, you can hear a series of clicks as
> the
> > > tone propagates down the stack. Sounds pretty
> much
> > > like the Classic Stacking recording!
> > > 
> > > If you want to try, here's how:
> > > 
> > > - Dial my ProjectMF trunk on 1-762-2601 (or
> > > 1-762-2602, skipping the net two steps, for
> those
> > > without a blue box)
> > > - Blow off the ringing with 2600
> > > - After the wink, dial KP-2602-ST
> > > - Wait for the dial tone
> > > - Dial 2602 with the DTMF pad of the phone
> > > - Wait for another dial tone
> > > - Dial 2602 with the DTMF pad
> > > - (Repeat X times, up to 23 times, as desired)
> > > - On the last dial tone, dial a number on CNET
> > with
> > > 011+CC_Number, again with the DTMF pad
> > > - Answer the ringing phone. You are now talking
> to
> > > yourself over X number of SF trunk links
> > > - Hit a touch tone digit onthe originating
> phone.
> > > Listen to it flash down the stack.
> > > - Hang up. Hear the stack disconnect with a
> cheep
> > of
> > > 2600 for each link
> > > 
> > > Notice how the connection takes longer to return
> > the
> > > dial tone on each loop through the stack, as the
> > > audio
> > > delay on the forward MF (which you can't hear)
> > gets
> > > longer and longer.
> > > 
> > > Regardless of how you come in. you can blow the
> > call
> > > off in the forward direction with 2600 and hear
> > the
> > > stack disconnect on the far side.
> > > 
> > > Much fun!
> > > 
> > > Don
> > > 
> > > PS - For those without MF and 2600, I set up
> > direct
> > > access to the SF/DISA arrangement on 1-762-2602
> > > 
> > > Note, if others are trying this, your available
> > > trunk
> > > pool may be less than 24.
> > > 
> > > --- Donald Froula <dfroula at sbcglobal.net> wrote:
> > > 
> > > > I hit upon an approach to try this trunk
> > stacking
> > > > thing through my TDM-Over-Ethernet SF/MF
> trunks.
> > I
> > > > set
> > > > up two extensions to go to DISA over the
> trunks.
> > > > What
> > > > I do is dial the extension that goes over the
> > > trunks
> > > > (similar to my 1-762-2601 number), but instead
> > of
> > > > programming the number that gives ringback in
> > the
> > > > Dial
> > > > command, I call another extension with the
> DISA
> > > > application that returns a dial tone.
> > > > 
> > > > 2602 => 1,Dial(ZAP/g1/9999)
> > > > 
> > > > 9999 => 1,DISA(password|from-internal)
> > > > 
> > > > I then dial the same number again (2602),
> which
> > > > seizes
> > > > the next line and gives a DISA dial tone
> again.
> > > This
> > > > is all via DTMF, so, yes, I am cheating.
> > However,
> > > > the
> > > > call setup on each loop through the trunk
> group
> > is
> > > > via
> > > > 2600 and MF.
> > > > 
> > > > I saturated all 24 lines of the trunk group
> this
> > > > way,
> > > > dialing one of my Evan Doorbell recordings on
> > the
> > > > last
> > > > trunk. The audio was looping through all 24
> > > > SF-controlled trunks. Both sides of the
> > connection
> > > > being on the same switch, I only noticed
> slight
> > > > degradation in the audio. I blew off the call
> > with
> > > a
> > > > chirp of 2600. This sounded exactly like
> blowing
> > > off
> > > > a
> > > > call on a single trunk, except for the 24
> > > disconnect
> > > > mrssages that flew by on the Asterisk console.
> > > > 
> > > > Now, if the looping could be done through
> > another
> > > > ProjectMF switch, it might sound interesting.
> > > > 
> > > > Don
> > > > 
> > > > --- Donald Froula <dfroula at sbcglobal.net>
> wrote:
> > > > 
> > > > > Mark, you asked about whether or not a 2600
> > > audio
> > > > > blip
> > > > > might get propagated through the system for
> > 2600
> > > > > flashes less than rxwink in length.
> Actually,
> > > they
> > > > > would. The 2600 notch filter DSP code takes
> a
> > > > > millisecond or so to kick in and out,
> creating
> > > an
> > > > > audible "blip" both on 2600 application and
> > > > removal.
> > > > > Although this has no effect on supervision,
> > > being
> > > > > purely an audio effect, it might provided
> > > > something
> > > > > like the old sound with the latency delay
> 
=== message truncated ===


From g4vft at btinternet.com  Fri Oct  5 04:21:07 2007
From: g4vft at btinternet.com (Jonathan Kay)
Date: Fri, 05 Oct 2007 10:21:07 +0100
Subject: [VoIP] Asterisk Org
Message-ID: <47060203.1010102@btinternet.com>


Gents.
I notice that Asterisk 1.2 is no longer obviously available on the 
asterisk.org website.
You can still get to it. But it's as if they are pushing 1.4 now.
Anyone heard any news about retiring 1.2 ???
Jon


From andrew.e.green at gmail.com  Fri Oct  5 05:15:35 2007
From: andrew.e.green at gmail.com (Andrew Green)
Date: Fri, 5 Oct 2007 07:45:35 -0230
Subject: [VoIP] Asterisk Org
In-Reply-To: <47060203.1010102@btinternet.com>
References: <47060203.1010102@btinternet.com>
Message-ID: <c30e256a0710050315g76a080dduc4fa0ef1aec4cf89@mail.gmail.com>

Dir listing:
http://downloads.digium.com/pub/asterisk/old-releases/
And the most recent 1.2:
http://downloads.digium.com/pub/asterisk/old-releases/asterisk-1.2.24.tar.gz

I never heard about Digium pushing 1.4 more than it has before but I
know that customers at the company I work for have less troubles using
1.4 than 1.2 (especially when it comes to it recognizing DTMF
presses.)

-Andrew

On 10/5/07, Jonathan Kay <g4vft at btinternet.com> wrote:
>
> Gents.
> I notice that Asterisk 1.2 is no longer obviously available on the
> asterisk.org website.
> You can still get to it. But it's as if they are pushing 1.4 now.
> Anyone heard any news about retiring 1.2 ???
> Jon
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

From jnovack at stromberg-carlson.org  Fri Oct  5 08:21:35 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 05 Oct 2007 09:21:35 -0400
Subject: [VoIP] Asterisk Org
In-Reply-To: <47060203.1010102@btinternet.com>
References: <47060203.1010102@btinternet.com>
Message-ID: <47063A5F.7010907@stromberg-carlson.org>

1.2 will only have security fixes now.
They ARE pushing people into 1.4
Simple upgrades do not seem possible either, and the last I tried, 1.4 
will not work on the CentOS 3.x Linux I have been using, but requires at 
least CentOS 4.x

For what I am doing, the 1.2 seems to be just fine.
I really don't feel up to making the drastic changes needed right now.
I know some others have though, and it seems to work.

John Novack


Jonathan Kay wrote:
> Gents.
> I notice that Asterisk 1.2 is no longer obviously available on the 
> asterisk.org website.
> You can still get to it. But it's as if they are pushing 1.4 now.
> Anyone heard any news about retiring 1.2 ???
> Jon
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From jnovack at stromberg-carlson.org  Fri Oct  5 14:07:21 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 05 Oct 2007 15:07:21 -0400
Subject: [VoIP] Asterisk Org
In-Reply-To: <47066518.4040609@rudholm.com>
References: <47060203.1010102@btinternet.com>
	<47063A5F.7010907@stromberg-carlson.org>
	<47066518.4040609@rudholm.com>
Message-ID: <47068B69.5040401@stromberg-carlson.org>



Mark Rudholm wrote:
> John Novack wrote:
>   
>> 1.2 will only have security fixes now.
>> They ARE pushing people into 1.4
>> Simple upgrades do not seem possible either, and the last I tried, 1.4 
>> will not work on the CentOS 3.x Linux I have been using, but requires at 
>> least CentOS 4.x
>>     
>
> That seems kind of weird.  Do you know why it won't work (or where
> it fails) on 3.x?-- 
>   
Some obscure ( to me at least ) dependencies with Linux prevent it from 
finishing the compile.
I do not remember now exactly what, nor if it was Zaptel or Asterisk.
And this was a very early release of 1.4

Perhaps Shane Young remembers more, as he was an early adopter.

John Novack

> Dog is my co-pilot

From mark at rudholm.com  Fri Oct  5 11:23:52 2007
From: mark at rudholm.com (Mark Rudholm)
Date: Fri, 05 Oct 2007 09:23:52 -0700
Subject: [VoIP] Asterisk Org
In-Reply-To: <47063A5F.7010907@stromberg-carlson.org>
References: <47060203.1010102@btinternet.com>
	<47063A5F.7010907@stromberg-carlson.org>
Message-ID: <47066518.4040609@rudholm.com>

John Novack wrote:
> 1.2 will only have security fixes now.
> They ARE pushing people into 1.4
> Simple upgrades do not seem possible either, and the last I tried, 1.4 
> will not work on the CentOS 3.x Linux I have been using, but requires at 
> least CentOS 4.x

That seems kind of weird.  Do you know why it won't work (or where
it fails) on 3.x?

From pdwills at cedarknolltelephone.com  Sat Oct  6 12:31:36 2007
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Sat, 06 Oct 2007 13:31:36 -0400
Subject: [VoIP] CNET Status
References: <47060203.1010102@btinternet.com>
	<47063A5F.7010907@stromberg-carlson.org> <47066518.4040609@rudholm.com>
Message-ID: <001701c8083e$bfd39c10$0201a8c0@Main>

An open note to Shane Young:

...and a progress report on a useful little utility.


>From time to time, I will discover that my CNET connection has been down for 
some time.  The audit program Shane has is a nice way to see who's up but I 
have to remember to check it from time to time.

I just came up with a combination of a batch file, wget for Windoze, a small 
C program to search for my status in the downloaded output from Shane's 
auditor and a popup generator to announce that something's wrong.  I can set 
it to run when I boot my computer so I will, at least, have a chance to 
discover something's wrong.

In other words, don't change anything.  The auditor works great!

PDW



From voiptandem at shaneyoung.com  Sat Oct  6 13:23:44 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sat, 06 Oct 2007 13:23:44 -0500
Subject: [VoIP] CNET Status
In-Reply-To: <001701c8083e$bfd39c10$0201a8c0@Main>
References: <47060203.1010102@btinternet.com>
	<47063A5F.7010907@stromberg-carlson.org> <47066518.4040609@rudholm.com>
	<001701c8083e$bfd39c10$0201a8c0@Main>
Message-ID: <20071006132344.4lcz3jlsg844oooo@mail.shaneyoung.com>

I love wget!  I've used it in Unix environments for years to grab data.

In fact, I use wget to make the auditor work :)

Quoting Paul Wills <pdwills at cedarknolltelephone.com>:

> An open note to Shane Young:
>
> ...and a progress report on a useful little utility.
>
>
>> From time to time, I will discover that my CNET connection has been down for
> some time.  The audit program Shane has is a nice way to see who's up but I
> have to remember to check it from time to time.
>
> I just came up with a combination of a batch file, wget for Windoze, a small
> C program to search for my status in the downloaded output from Shane's
> auditor and a popup generator to announce that something's wrong.  I can set
> it to run when I boot my computer so I will, at least, have a chance to
> discover something's wrong.
>
> In other words, don't change anything.  The auditor works great!
>
> PDW
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

--Shane
+1-821-7311 CNET

----------------------------------------------------------------


From john_reads_cnet_via_archives at covert.org  Sun Oct  7 08:22:30 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Sun, 7 Oct 2007 09:22:30 -0400 (EDT)
Subject: [VoIP] Just Imagine (1947)
Message-ID: <20071007132818.F390156317@ns01.ckts.info>

Animated assembly of a Model 300 WeCo phone from 433 parts.

The way the Bell System used to make movies.

http://youtube.com/watch?v=kHOHtNSFPdQ

/john

From chad at maine.edu  Thu Oct 11 20:59:15 2007
From: chad at maine.edu (Chad Perkins)
Date: Thu, 11 Oct 2007 21:59:15 -0400
Subject: [VoIP] Offices +1 700 and 955 back on the air
Message-ID: <470E9CB3.29170.183008@localhost>

Verizon DSL burpage...

From hockd at dteenergy.com  Fri Oct 12 11:57:06 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Fri, 12 Oct 2007 12:57:06 -0400
Subject: [VoIP] Offices +1 700 and 955 back on the air
In-Reply-To: <470E9CB3.29170.183008@localhost>
References: <470E9CB3.29170.183008@localhost>
Message-ID: <OF1D06A799.5C245783-ON85257372.005D1E4D-85257372.005D1E50@dteenergy.com>

Too much root beer?

Dennis Hock

-----voip-bounces at ckts.info wrote: -----


To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
From: "Chad Perkins" <chad at maine.edu>
Sent by: voip-bounces at ckts.info
Date: 10/11/2007 09:59PM
Subject: [VoIP] Offices +1 700 and 955 back on the air

Verizon DSL burpage...
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/

From ratguy at insightbb.com  Sun Oct 14 18:25:39 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Sun, 14 Oct 2007 19:25:39 -0400
Subject: [VoIP] Continuous music on hold?
Message-ID: <000501c80eb9$8881b680$6700a8c0@bluegrasspals.com>

Hi,

     I'm wanting to implement a continuously running music on hold. What I
mean is this. In a traditional Asterisk MOH, when the last person connected
to the MOH hangs up, the music stops, and doesn't continue until someone
else gets put on hold. In other words, if you hang up on an MOH and call
right back, no music or audio will have passed between the time the call was
disconnected and when you connected again. I'd like to have a continuously
running MOH, which runs even with nobody connected to it. This would allow
for the implementation of a certain style of centralized drum recording
machines from back in the day before I was born, where when you were
connected to the recording, it just dropped you on the recording at whatever
point it happened to be, and the recording was running all the time, even
with nobody connected. Therefore, you never knew where you were going to
come in on any given call. I've thought of having something dial an MOH
context and just keep the connection up. However, is there any way for
Asterisk to dial into something within itself when nobody real is actually
making a call?
Thanks.
Jayson

From greg at vyger.net  Sun Oct 14 20:11:00 2007
From: greg at vyger.net (Greg Blakely)
Date: Sun, 14 Oct 2007 20:11:00 -0500
Subject: [VoIP] Continuous music on hold?
Message-ID: <A691C3685A152E4AAC9608B8D8E11316686576@godzilla.hickory.vyger.net>

Along those same lines, I have a key system that sits behind my asterisk
pbx.  I'd like to use that 'continuously running MOH' on my key system
as well.  Would it take a sound card in the asterisk box?  Is it even
possible?
 

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of Jayson Smith
> Sent: Sunday, October 14, 2007 6:26 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: [VoIP] Continuous music on hold?
> 
> Hi,
> 
>      I'm wanting to implement a continuously running music on 
> hold. What I mean is this. In a traditional Asterisk MOH, 
> when the last person connected to the MOH hangs up, the music 
> stops, and doesn't continue until someone else gets put on 
> hold. In other words, if you hang up on an MOH and call right 
> back, no music or audio will have passed between the time the 
> call was disconnected and when you connected again. I'd like 
> to have a continuously running MOH, which runs even with 
> nobody connected to it. This would allow for the 
> implementation of a certain style of centralized drum 
> recording machines from back in the day before I was born, 
> where when you were connected to the recording, it just 
> dropped you on the recording at whatever point it happened to 
> be, and the recording was running all the time, even with 
> nobody connected. Therefore, you never knew where you were 
> going to come in on any given call. I've thought of having 
> something dial an MOH context and just keep the connection 
> up. However, is there any way for Asterisk to dial into 
> something within itself when nobody real is actually making a call?
> Thanks.
> Jayson
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 

From jnovack at stromberg-carlson.org  Sun Oct 14 20:24:23 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 14 Oct 2007 21:24:23 -0400
Subject: [VoIP] Continuous music on hold?
In-Reply-To: <A691C3685A152E4AAC9608B8D8E11316686576@godzilla.hickory.vyger.net>
References: <A691C3685A152E4AAC9608B8D8E11316686576@godzilla.hickory.vyger.net>
Message-ID: <4712C147.7070209@stromberg-carlson.org>

Doesn't this work somewhat differently with different versions of Asterisk?
Later 1.2 has a "native" mode which either plays continuously, or is it 
the other way around?

Or was this a change in 1.4
There was some discussion on the Asterisk users list recently regarding 
this.


John Novack


Greg Blakely wrote:
> Along those same lines, I have a key system that sits behind my asterisk
> pbx.  I'd like to use that 'continuously running MOH' on my key system
> as well.  Would it take a sound card in the asterisk box?  Is it even
> possible?
>  
>
>   
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
>> On Behalf Of Jayson Smith
>> Sent: Sunday, October 14, 2007 6:26 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: [VoIP] Continuous music on hold?
>>
>> Hi,
>>
>>      I'm wanting to implement a continuously running music on 
>> hold. What I mean is this. In a traditional Asterisk MOH, 
>> when the last person connected to the MOH hangs up, the music 
>> stops, and doesn't continue until someone else gets put on 
>> hold. In other words, if you hang up on an MOH and call right 
>> back, no music or audio will have passed between the time the 
>> call was disconnected and when you connected again. I'd like 
>> to have a continuously running MOH, which runs even with 
>> nobody connected to it. This would allow for the 
>> implementation of a certain style of centralized drum 
>> recording machines from back in the day before I was born, 
>> where when you were connected to the recording, it just 
>> dropped you on the recording at whatever point it happened to 
>> be, and the recording was running all the time, even with 
>> nobody connected. Therefore, you never knew where you were 
>> going to come in on any given call. I've thought of having 
>> something dial an MOH context and just keep the connection 
>> up. However, is there any way for Asterisk to dial into 
>> something within itself when nobody real is actually making a call?
>> Thanks.
>> Jayson
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>     
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From kxt at fubegra.net  Sun Oct 14 20:37:57 2007
From: kxt at fubegra.net (Russ Price)
Date: Sun, 14 Oct 2007 20:37:57 -0500
Subject: [VoIP] Continuous music on hold?
In-Reply-To: <A691C3685A152E4AAC9608B8D8E11316686576@godzilla.hickory.vyger.net>
References: <A691C3685A152E4AAC9608B8D8E11316686576@godzilla.hickory.vyger.net>
Message-ID: <4712C475.7080104@fubegra.net>

Greg Blakely wrote:
> Along those same lines, I have a key system that sits behind my asterisk
> pbx.  I'd like to use that 'continuously running MOH' on my key system
> as well.  Would it take a sound card in the asterisk box?  Is it even
> possible?
>  

It might be possible to set up something with a conference bridge that's 
reserved for your continuous audio source.  Asterisk does support using 
a sound card (the console channel), so you could use an Asterisk Manager 
  Interface app to connect the console to the conference bridge.

You could also use a different channel type (e.g. Zap or SIP) for the 
audio source.

Depending on how you set up the conference, you could make your 
recording talkable...

	Russ

From voiptandem at shaneyoung.com  Sun Oct 14 21:10:18 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sun, 14 Oct 2007 21:10:18 -0500
Subject: [VoIP] Continuous music on hold?
In-Reply-To: <4712C147.7070209@stromberg-carlson.org>
References: <A691C3685A152E4AAC9608B8D8E11316686576@godzilla.hickory.vyger.net>
	<4712C147.7070209@stromberg-carlson.org>
Message-ID: <20071014211018.k7k7jgmrtww4ccws@secure.shaneyoung.com>

In 1.4 the music starts from the beginning reading the file directly  
rather than through the mgp123 program used in 1.2

You could use the cosole channel to dial into a MOH extension which  
should get it started and keep it going (you don't need to have  
anything plugged into the console).  Anyone else getting put on hold  
should be dropped into the middle.


Quoting John Novack <jnovack at stromberg-carlson.org>:

> Doesn't this work somewhat differently with different versions of Asterisk?
> Later 1.2 has a "native" mode which either plays continuously, or is it
> the other way around?
>
> Or was this a change in 1.4
> There was some discussion on the Asterisk users list recently regarding
> this.
>
>
> John Novack
>
>
> Greg Blakely wrote:
>> Along those same lines, I have a key system that sits behind my asterisk
>> pbx.  I'd like to use that 'continuously running MOH' on my key system
>> as well.  Would it take a sound card in the asterisk box?  Is it even
>> possible?
>>
>>
>>
>>> -----Original Message-----
>>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info]
>>> On Behalf Of Jayson Smith
>>> Sent: Sunday, October 14, 2007 6:26 PM
>>> To: Voice Over IP Tandem for Analog Switches
>>> Subject: [VoIP] Continuous music on hold?
>>>
>>> Hi,
>>>
>>>      I'm wanting to implement a continuously running music on
>>> hold. What I mean is this. In a traditional Asterisk MOH,
>>> when the last person connected to the MOH hangs up, the music
>>> stops, and doesn't continue until someone else gets put on
>>> hold. In other words, if you hang up on an MOH and call right
>>> back, no music or audio will have passed between the time the
>>> call was disconnected and when you connected again. I'd like
>>> to have a continuously running MOH, which runs even with
>>> nobody connected to it. This would allow for the
>>> implementation of a certain style of centralized drum
>>> recording machines from back in the day before I was born,
>>> where when you were connected to the recording, it just
>>> dropped you on the recording at whatever point it happened to
>>> be, and the recording was running all the time, even with
>>> nobody connected. Therefore, you never knew where you were
>>> going to come in on any given call. I've thought of having
>>> something dial an MOH context and just keep the connection
>>> up. However, is there any way for Asterisk to dial into
>>> something within itself when nobody real is actually making a call?
>>> Thanks.
>>> Jayson
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>
>>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>
> --
> Dog is my co-pilot
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

--Shane
+1-821-7311 CNET

----------------------------------------------------------------


From ratguy at insightbb.com  Mon Oct 15 09:52:25 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Mon, 15 Oct 2007 10:52:25 -0400
Subject: [VoIP] Payphone project?
Message-ID: <000d01c80f3b$0014a820$6700a8c0@bluegrasspals.com>

Hi,

     Mark, any news on the proposed payphone project? I've been talking with
Donald Froula, and he seems to be having trouble coming up with a design for
a coin relay control circuit that doesn't involve modifying the coin relay
itself. I'd love to have a payphone, but imho they'd be no fun without coin
control. I'm planning on buying one from Phil whenever we get this sorted
out. Just curious about any progress, etc.
Jayson

From lee at spenadel.com  Mon Oct 15 16:09:34 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Mon, 15 Oct 2007 17:09:34 -0400
Subject: [VoIP] Payphone project?
In-Reply-To: <000d01c80f3b$0014a820$6700a8c0@bluegrasspals.com>
References: <000d01c80f3b$0014a820$6700a8c0@bluegrasspals.com>
Message-ID: <022701c80f6f$afe472b0$0fad5810$@com>

This is something that perhaps Stan Schreier can help with.  He has built
amazing coin controller units for use with single and three-slot payphones
that do not require the modification of the coin relay.  Stan will need to
be brought up to speed on Asterisk and what's being accomplished.  I have
copied him on this thread to see if we can peak his interest.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Jayson Smith
Sent: Monday, October 15, 2007 10:52 AM
To: Voice Over IP Tandem for Analog Switches
Subject: [VoIP] Payphone project?

Hi,

     Mark, any news on the proposed payphone project? I've been talking with
Donald Froula, and he seems to be having trouble coming up with a design for
a coin relay control circuit that doesn't involve modifying the coin relay
itself. I'd love to have a payphone, but imho they'd be no fun without coin
control. I'm planning on buying one from Phil whenever we get this sorted
out. Just curious about any progress, etc.
Jayson
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/


From g4vft at btinternet.com  Wed Oct 17 06:48:27 2007
From: g4vft at btinternet.com (Jonathan Kay)
Date: Wed, 17 Oct 2007 12:48:27 +0100
Subject: [VoIP] Fedora, CentOS or What?
Message-ID: <4715F68B.4050808@btinternet.com>


Chaps.
 If you started from scratch now, what would be your Linux distro of 
choice for a purely Asterisk box?
I'm building a few machines to pass on at cost price.

CentOS 4, was available in a "Server" edition. I had problems compiling 
the latest zaptel with that and had to upgrade kernel etc. CentOS 5, 
doesn't offer the server edition.

Fedora 5, and 6, I think had kernel version issues, again I had lots of 
trouble compiling Asterisk with FC6. I don't know if 7 is ok?

Ubuntu is great, I use desktop and server versions. But it is Debian 
based, and the community here and on the UK list tend to use flavours of 
Red Hat with Asterisk, so getting CNET support might be more tricky.

Comments welcome !?

Jon

From kxt at fubegra.net  Wed Oct 17 07:17:15 2007
From: kxt at fubegra.net (Russ Price)
Date: Wed, 17 Oct 2007 07:17:15 -0500
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <4715F68B.4050808@btinternet.com>
References: <4715F68B.4050808@btinternet.com>
Message-ID: <4715FD4B.8040104@fubegra.net>

Jonathan Kay wrote:
>  If you started from scratch now, what would be your Linux distro of 
> choice for a purely Asterisk box?

The problem with Fedora is that it is in constant flux, and updates are 
only available for about six months beyond the next release.  After 
that, your system will be a sitting duck for cracking.  Furthermore, 
when you update a Fedora box, there's always a chance that something 
will break - I've been bitten by that on laptops with wireless cards, 
and on my gaming/media system, where sound stopped working after one 
update.  That prompted me to switch it to CentOS 5.

CentOS 3 end of life is Oct. 31, 2010; for CentOS 4, Feb. 29, 2012; 
version 5 will last until Mar. 31, 2014.  I would consider CentOS 4 to 
be just fine - as long as you keep current with upgrades.

The CentOS 4 server CD is a 4.4 version, so it would need to be upgraded 
to 4.5 immediately once you install it; a "yum update" would suffice.

I haven't tried setting up a CentOS 5 Asterisk box - mine is a 4.5 system.

If you decide to go with Ubuntu, use a Long Term Support edition, and it 
would probably be best to compile Asterisk from source for CNET purposes.

	Russ

From dfroula at sbcglobal.net  Wed Oct 17 07:23:34 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Wed, 17 Oct 2007 05:23:34 -0700 (PDT)
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <4715F68B.4050808@btinternet.com>
Message-ID: <952953.46470.qm@web83213.mail.mud.yahoo.com>

I've had no problems with Debian "Sarge" and Asterisk.
Compiling with ProjectMF patches has been
problem-free.

Don

--- Jonathan Kay <g4vft at btinternet.com> wrote:

> 
> Chaps.
>  If you started from scratch now, what would be your
> Linux distro of 
> choice for a purely Asterisk box?
> I'm building a few machines to pass on at cost
> price.
> 
> CentOS 4, was available in a "Server" edition. I had
> problems compiling 
> the latest zaptel with that and had to upgrade
> kernel etc. CentOS 5, 
> doesn't offer the server edition.
> 
> Fedora 5, and 6, I think had kernel version issues,
> again I had lots of 
> trouble compiling Asterisk with FC6. I don't know if
> 7 is ok?
> 
> Ubuntu is great, I use desktop and server versions.
> But it is Debian 
> based, and the community here and on the UK list
> tend to use flavours of 
> Red Hat with Asterisk, so getting CNET support might
> be more tricky.
> 
> Comments welcome !?
> 
> Jon
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From voiptandem at shaneyoung.com  Wed Oct 17 07:44:30 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Wed, 17 Oct 2007 07:44:30 -0500
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <4715F68B.4050808@btinternet.com>
References: <4715F68B.4050808@btinternet.com>
Message-ID: <20071017074430.pyry5k3esg0k80c0@secure.shaneyoung.com>

Quoting Jonathan Kay <g4vft at btinternet.com>:

>
> Chaps.
>  If you started from scratch now, what would be your Linux distro of
> choice for a purely Asterisk box?
> I'm building a few machines to pass on at cost price.

I have recently (in the last two years) started using Gentoo.  It's a  
little tricky to install the first 2 or 3 times, but after a while  
I've found I'm almost as comfortable installing it as I was with  
Slackware.  Previously (for the last 13 or so years) Slackware was my  
preferred distro.

I'm a little supprized that there isn't much discussion about  
AstLinux.  The whole thing can run on a 64meg compact flash card (or  
from a regular hard drive).  If you have a new enough machine that  
will boot from USB, you can install it on a thumbdrive.

It's Linux built just for running Asterisk with very little overhead.

I'm in the process now of building some machines using AstLinux to try  
it out.  I'll have two external compact flash card slots.  One for  
AstLinux and one for the configuration files.  If the AstLinux card  
fails, I just build a new one on my PC (about a 5 minute process),  
replace the old one and reboot.  All of the configuration is read from  
the config card.

I'm curious what you mean by "getting CNET support might be more  
tricky".  Are you referring to getting support from other CNET members?

--Shane
+1-821-7311 CNET

----------------------------------------------------------------


From jnovack at stromberg-carlson.org  Wed Oct 17 09:02:40 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 17 Oct 2007 10:02:40 -0400
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <4715F68B.4050808@btinternet.com>
References: <4715F68B.4050808@btinternet.com>
Message-ID: <47161600.10307@stromberg-carlson.org>

Boy, you Brits really know how to start a religious argument!!

My choice has been, for better or worse, CentOS.
CentOS version 3 works well on 1.2 versions
CentOS 4/5 seems to be required on the 1.4 versions
So far I have seen no reason to move to 1.4, though some have for 
various reasons
Many on the Asterisk users list consider 1.4 still unstable, though I 
doubt any of the Cnet uses will be affected.
I always install everything ( moans from the Linux gurus here ) but I 
have found it easier to install everything and then turn off services I 
don't need, rather than later battle some cryptic message regarding a 
missing dependency that I can't fix or know where to stick a file.
I also use the full, rather than the server version, and do an update 
from yum before proceeding to Asterisk.
I have cloned several hard drives for others on CNET with fair success, 
saving having to start from scratch every time.
Easier to change a machine name and such than reinstall everything from 
scratch.
Acronis version 9 handles Linux disks well, even when the geometry isn't 
the same.

Others can comment on the other religions. I suppose it is down to 
whatever we have been imprinted with in our early Linux days, and what 
we have become familiar with.
I have dabbled in the Debian pool, but have not met with any success.

John Novack


Jonathan Kay wrote:
> Chaps.
>  If you started from scratch now, what would be your Linux distro of 
> choice for a purely Asterisk box?
> I'm building a few machines to pass on at cost price.
>
> CentOS 4, was available in a "Server" edition. I had problems compiling 
> the latest zaptel with that and had to upgrade kernel etc. CentOS 5, 
> doesn't offer the server edition.
>
> Fedora 5, and 6, I think had kernel version issues, again I had lots of 
> trouble compiling Asterisk with FC6. I don't know if 7 is ok?
>
> Ubuntu is great, I use desktop and server versions. But it is Debian 
> based, and the community here and on the UK list tend to use flavours of 
> Red Hat with Asterisk, so getting CNET support might be more tricky.
>
> Comments welcome !?
>
> Jon
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From jnovack at stromberg-carlson.org  Wed Oct 17 09:08:38 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 17 Oct 2007 10:08:38 -0400
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <20071017074430.pyry5k3esg0k80c0@secure.shaneyoung.com>
References: <4715F68B.4050808@btinternet.com>
	<20071017074430.pyry5k3esg0k80c0@secure.shaneyoung.com>
Message-ID: <47161766.4030104@stromberg-carlson.org>



Shane Young wrote:
> Quoting Jonathan Kay <g4vft at btinternet.com>:
>
>   
>> Chaps.
>>  If you started from scratch now, what would be your Linux distro of
>> choice for a purely Asterisk box?
>> I'm building a few machines to pass on at cost price.
>>     
> <snip>
> I'm a little supprized that there isn't much discussion about  
> AstLinux.  The whole thing can run on a 64meg compact flash card (or  
> from a regular hard drive).  If you have a new enough machine that  
> will boot from USB, you can install it on a thumbdrive.
>
> It's Linux built just for running Asterisk with very little overhead.
>
> I'm in the process now of building some machines using AstLinux to try  
> it out.  I'll have two external compact flash card slots.  One for  
> AstLinux and one for the configuration files.  If the AstLinux card  
> fails, I just build a new one on my PC (about a 5 minute process),  
> replace the old one and reboot.  All of the configuration is read from  
> the config card.
>   
I wonder how this would work with the cheaply available HP "thin Client" 
units?
64 Meg of flash, 128 meg of ram, 800 Meg processor?

Can you supply any details on installation? Either on or off list? The 
Flash on these units is plug compatible ( though of the wrong sex ) with 
the 2.5 Hard drive. With the right cable a HD can be substituted for the 
flash memory. Main memory on some can be upped, but not the 5520.
These may be an interesting alternative to the "Asterisk in a Router" 
some have been using.

Lets explore this further.

John Novack

-- 
Dog is my co-pilot


From voiptandem at shaneyoung.com  Wed Oct 17 09:53:41 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Wed, 17 Oct 2007 09:53:41 -0500
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <47161766.4030104@stromberg-carlson.org>
References: <4715F68B.4050808@btinternet.com>
	<20071017074430.pyry5k3esg0k80c0@secure.shaneyoung.com>
	<47161766.4030104@stromberg-carlson.org>
Message-ID: <20071017095341.gpiq7j9e4gowcws4@secure.shaneyoung.com>

It's a pretty cool thing.

Get a compact flash card and reader for your PC.
Download the files from sourceforge.
Run the command that builds a filesystem on the flash drive from the  
files you download.

remove the flash drive from your pc.
Insert it into the new device.
Boot.

It prefers to find a second device to write it's configs to.

The compact flash to IDE reader is about $10-25.  It just looks like  
an IDE drive to the system.

The documentation is a little lacking, but it is pretty nifty.


Quoting John Novack <jnovack at stromberg-carlson.org>:

>
>
> Shane Young wrote:
>> Quoting Jonathan Kay <g4vft at btinternet.com>:
>>
>>
>>> Chaps.
>>>  If you started from scratch now, what would be your Linux distro of
>>> choice for a purely Asterisk box?
>>> I'm building a few machines to pass on at cost price.
>>>
>> <snip>
>> I'm a little supprized that there isn't much discussion about
>> AstLinux.  The whole thing can run on a 64meg compact flash card (or
>> from a regular hard drive).  If you have a new enough machine that
>> will boot from USB, you can install it on a thumbdrive.
>>
>> It's Linux built just for running Asterisk with very little overhead.
>>
>> I'm in the process now of building some machines using AstLinux to try
>> it out.  I'll have two external compact flash card slots.  One for
>> AstLinux and one for the configuration files.  If the AstLinux card
>> fails, I just build a new one on my PC (about a 5 minute process),
>> replace the old one and reboot.  All of the configuration is read from
>> the config card.
>>
> I wonder how this would work with the cheaply available HP "thin Client"
> units?
> 64 Meg of flash, 128 meg of ram, 800 Meg processor?
>
> Can you supply any details on installation? Either on or off list? The
> Flash on these units is plug compatible ( though of the wrong sex ) with
> the 2.5 Hard drive. With the right cable a HD can be substituted for the
> flash memory. Main memory on some can be upped, but not the 5520.
> These may be an interesting alternative to the "Asterisk in a Router"
> some have been using.
>
> Lets explore this further.
>
> John Novack
>
> --
> Dog is my co-pilot
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

--Shane
+1-821-7311 CNET

----------------------------------------------------------------


From jnovack at stromberg-carlson.org  Wed Oct 17 10:05:23 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 17 Oct 2007 11:05:23 -0400
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <20071017095341.gpiq7j9e4gowcws4@secure.shaneyoung.com>
References: <4715F68B.4050808@btinternet.com>	<20071017074430.pyry5k3esg0k80c0@secure.shaneyoung.com>	<47161766.4030104@stromberg-carlson.org>
	<20071017095341.gpiq7j9e4gowcws4@secure.shaneyoung.com>
Message-ID: <471624B3.4000001@stromberg-carlson.org>

Unfortunately, these thin client machines don't have a "standard" CF card.
They have a female socket that would mate to a 2.5 inch HD.
I have managed to find a cable that allows me to substitute a 2.5 inch 
hard drive for the circuit board, but I can't yet find something that 
would allow me to substitute this card for a HD.
All the sexes of the cables are pointing the wrong way!!
I suppose IF I am able to write AstLinux to the flash, then I could use 
a pen drive on the machine to store configs and sound files??
Right now I could install AstLinux to a 4 gig 2.5 drive, but that seems 
like cheating.
Do you see the 128 meg of Ram a problem?

Stay tuned


John Novack

Shane Young wrote:
> It's a pretty cool thing.
>
> Get a compact flash card and reader for your PC.
> Download the files from sourceforge.
> Run the command that builds a filesystem on the flash drive from the  
> files you download.
>
> remove the flash drive from your pc.
> Insert it into the new device.
> Boot.
>
> It prefers to find a second device to write it's configs to.
>
> The compact flash to IDE reader is about $10-25.  It just looks like  
> an IDE drive to the system.
>
> The documentation is a little lacking, but it is pretty nifty.
>
>
> Quoting John Novack <jnovack at stromberg-carlson.org>:
>
>   
>> Shane Young wrote:
>>     
>>> Quoting Jonathan Kay <g4vft at btinternet.com>:
>>>
>>>
>>>       
>>>> Chaps.
>>>>  If you started from scratch now, what would be your Linux distro of
>>>> choice for a purely Asterisk box?
>>>> I'm building a few machines to pass on at cost price.
>>>>
>>>>         
>>> <snip>
>>> I'm a little supprized that there isn't much discussion about
>>> AstLinux.  The whole thing can run on a 64meg compact flash card (or
>>> from a regular hard drive).  If you have a new enough machine that
>>> will boot from USB, you can install it on a thumbdrive.
>>>
>>> It's Linux built just for running Asterisk with very little overhead.
>>>
>>> I'm in the process now of building some machines using AstLinux to try
>>> it out.  I'll have two external compact flash card slots.  One for
>>> AstLinux and one for the configuration files.  If the AstLinux card
>>> fails, I just build a new one on my PC (about a 5 minute process),
>>> replace the old one and reboot.  All of the configuration is read from
>>> the config card.
>>>
>>>       
>> I wonder how this would work with the cheaply available HP "thin Client"
>> units?
>> 64 Meg of flash, 128 meg of ram, 800 Meg processor?
>>
>> Can you supply any details on installation? Either on or off list? The
>> Flash on these units is plug compatible ( though of the wrong sex ) with
>> the 2.5 Hard drive. With the right cable a HD can be substituted for the
>> flash memory. Main memory on some can be upped, but not the 5520.
>> These may be an interesting alternative to the "Asterisk in a Router"
>> some have been using.
>>
>> Lets explore this further.
>>
>> John Novack
>>
>> --
>> Dog is my co-pilot
>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>     
>
> --Shane
> +1-821-7311 CNET
>
> ----------------------------------------------------------------
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From david at josephson.com  Thu Oct 18 00:45:15 2007
From: david at josephson.com (David Josephson)
Date: Thu, 18 Oct 2007 01:45:15 -0400
Subject: [VoIP] Office code assignment wiki hacked
Message-ID: <4716F2EB.1080006@josephson.com>

Sometimes these things just happen, I guess two years is a good record.

The CNET wiki page for office code assignments was hacked yesterday by a 
very determined robot, whi't just  appeared to be simultaneously in 
China, India and in unassigned IP address space. So, instead of 
www.altaphon.com/cgi-bin/f8n.pl?TelCollNet, change the f8n to etz1k (the 
fourth character is a numeral 1) ... we have recovered the changes.

And while I'm here, I was surprised last week to get an actual call 
through the crossbar exchange from a list member in San Jose. I think 
this may have been the first actual CNET call to an actual listed 
extension, that wasn't just to a test number on the switch, even though 
it's been online for almost a year. 555-2368 does indeed ring on my desk 
(through a WE 202 set, no less) and it is not simulated ringback tone.

Cheers

--
David Josephson

From ian at uax.org.uk  Thu Oct 18 04:09:14 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Thu, 18 Oct 2007 10:09:14 +0100
Subject: [VoIP] Office code assignment wiki hacked
References: <4716F2EB.1080006@josephson.com>
Message-ID: <001e01c81166$8e0b49e0$0c01a8c0@acer1dd0bbc6d0>

Might it be an idea such that only 'registered' CNET members can get access 
to the WiKi ?

After all those interested enough will have 'registered' so we should have 
some idea as to who they are. In the UK all the CNET members except one are 
know to at least one of the main CNET users.  On this side of the 'pond', 
some of us had thought that it was rather open to abuse.

Shouldn't be too difficult to link the ckts.info password should it?

Ian Jolly

+44 (0)  352      82 26 (via a 1929 GPO Rural Automatic eXchange!) from 
CNET - the Heritage Telephone Network
+44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from  Public 
Telephone Network
FWD Telephone No  83 2230


----- Original Message ----- 
From: "David Josephson" <david at josephson.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
Sent: Thursday, October 18, 2007 6:45 AM
Subject: [VoIP] Office code assignment wiki hacked


> Sometimes these things just happen, I guess two years is a good record.
>
> The CNET wiki page for office code assignments was hacked yesterday by a
> very determined robot, whi't just  appeared to be simultaneously in
> China, India and in unassigned IP address space. So, instead of
> www.altaphon.com/cgi-bin/f8n.pl?TelCollNet, change the f8n to etz1k (the
> fourth character is a numeral 1) ... we have recovered the changes.
>
> And while I'm here, I was surprised last week to get an actual call
> through the crossbar exchange from a list member in San Jose. I think
> this may have been the first actual CNET call to an actual listed
> extension, that wasn't just to a test number on the switch, even though
> it's been online for almost a year. 555-2368 does indeed ring on my desk
> (through a WE 202 set, no less) and it is not simulated ringback tone.
>
> Cheers
>
> --
> David Josephson
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: 
> 16/10/2007 14:14
> 


From greg at vyger.net  Thu Oct 18 07:15:24 2007
From: greg at vyger.net (Greg Blakely)
Date: Thu, 18 Oct 2007 07:15:24 -0500
Subject: [VoIP] Office code assignment wiki hacked
Message-ID: <A691C3685A152E4AAC9608B8D8E11316686596@godzilla.hickory.vyger.net>

We could certainly do some sort of behind-the-scenes authentication, and
that may actually be the best route.  Instead of the 'reserve, register,
and activate' scenario, it could end up being 'register, reserve, and
activate.'


> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of Ian Jolly
> Sent: Thursday, October 18, 2007 4:09 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Office code assignment wiki hacked
> 
> Might it be an idea such that only 'registered' CNET members 
> can get access to the WiKi ?
> 
> After all those interested enough will have 'registered' so 
> we should have some idea as to who they are. In the UK all 
> the CNET members except one are know to at least one of the 
> main CNET users.  On this side of the 'pond', some of us had 
> thought that it was rather open to abuse.
> 
> Shouldn't be too difficult to link the ckts.info password should it?
> 
> Ian Jolly
> 
> +44 (0)  352      82 26 (via a 1929 GPO Rural Automatic 
> eXchange!) from 
> CNET - the Heritage Telephone Network
> +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from  
> +Public
> Telephone Network
> FWD Telephone No  83 2230
> 
> 
> ----- Original Message -----
> From: "David Josephson" <david at josephson.com>
> To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
> Sent: Thursday, October 18, 2007 6:45 AM
> Subject: [VoIP] Office code assignment wiki hacked
> 
> 
> > Sometimes these things just happen, I guess two years is a 
> good record.
> >
> > The CNET wiki page for office code assignments was hacked 
> yesterday by a
> > very determined robot, whi't just  appeared to be simultaneously in
> > China, India and in unassigned IP address space. So, instead of
> > www.altaphon.com/cgi-bin/f8n.pl?TelCollNet, change the f8n 
> to etz1k (the
> > fourth character is a numeral 1) ... we have recovered the changes.
> >
> > And while I'm here, I was surprised last week to get an actual call
> > through the crossbar exchange from a list member in San 
> Jose. I think
> > this may have been the first actual CNET call to an actual listed
> > extension, that wasn't just to a test number on the switch, 
> even though
> > it's been online for almost a year. 555-2368 does indeed 
> ring on my desk
> > (through a WE 202 set, no less) and it is not simulated 
> ringback tone.
> >
> > Cheers
> >
> > --
> > David Josephson
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
> > -- 
> > No virus found in this incoming message.
> > Checked by AVG Free Edition.
> > Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: 
> > 16/10/2007 14:14
> > 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 

From lee at spenadel.com  Thu Oct 18 09:33:22 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Thu, 18 Oct 2007 10:33:22 -0400
Subject: [VoIP] Office code assignment wiki hacked
In-Reply-To: <A691C3685A152E4AAC9608B8D8E11316686596@godzilla.hickory.vyger.net>
References: <A691C3685A152E4AAC9608B8D8E11316686596@godzilla.hickory.vyger.net>
Message-ID: <002401c81193$d6f2f310$84d8d930$@com>

While this is on a slightly different note, is there a way to get the
person's name back in the user listings on the directory pages?  I find it
difficult to look up another CNET member and get their number with the new
page format.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Greg Blakely
Sent: Thursday, October 18, 2007 8:15 AM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Office code assignment wiki hacked

We could certainly do some sort of behind-the-scenes authentication, and
that may actually be the best route.  Instead of the 'reserve, register,
and activate' scenario, it could end up being 'register, reserve, and
activate.'


> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of Ian Jolly
> Sent: Thursday, October 18, 2007 4:09 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Office code assignment wiki hacked
> 
> Might it be an idea such that only 'registered' CNET members 
> can get access to the WiKi ?
> 
> After all those interested enough will have 'registered' so 
> we should have some idea as to who they are. In the UK all 
> the CNET members except one are know to at least one of the 
> main CNET users.  On this side of the 'pond', some of us had 
> thought that it was rather open to abuse.
> 
> Shouldn't be too difficult to link the ckts.info password should it?
> 
> Ian Jolly
> 
> +44 (0)  352      82 26 (via a 1929 GPO Rural Automatic 
> eXchange!) from 
> CNET - the Heritage Telephone Network
> +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from  
> +Public
> Telephone Network
> FWD Telephone No  83 2230
> 
> 
> ----- Original Message -----
> From: "David Josephson" <david at josephson.com>
> To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
> Sent: Thursday, October 18, 2007 6:45 AM
> Subject: [VoIP] Office code assignment wiki hacked
> 
> 
> > Sometimes these things just happen, I guess two years is a 
> good record.
> >
> > The CNET wiki page for office code assignments was hacked 
> yesterday by a
> > very determined robot, whi't just  appeared to be simultaneously in
> > China, India and in unassigned IP address space. So, instead of
> > www.altaphon.com/cgi-bin/f8n.pl?TelCollNet, change the f8n 
> to etz1k (the
> > fourth character is a numeral 1) ... we have recovered the changes.
> >
> > And while I'm here, I was surprised last week to get an actual call
> > through the crossbar exchange from a list member in San 
> Jose. I think
> > this may have been the first actual CNET call to an actual listed
> > extension, that wasn't just to a test number on the switch, 
> even though
> > it's been online for almost a year. 555-2368 does indeed 
> ring on my desk
> > (through a WE 202 set, no less) and it is not simulated 
> ringback tone.
> >
> > Cheers
> >
> > --
> > David Josephson
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
> > -- 
> > No virus found in this incoming message.
> > Checked by AVG Free Edition.
> > Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: 
> > 16/10/2007 14:14
> > 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/


From ian at uax.org.uk  Thu Oct 18 12:26:12 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Thu, 18 Oct 2007 18:26:12 +0100
Subject: [VoIP] "Owner" column (was - Office code assignment wiki
	hacked)
References: <A691C3685A152E4AAC9608B8D8E11316686596@godzilla.hickory.vyger.net>
	<002401c81193$d6f2f310$84d8d930$@com>
Message-ID: <00e001c811ab$fbb879f0$0c01a8c0@acer1dd0bbc6d0>

In 'Europe', we've haven't had an 'owner' column for a while -  to give more 
room for 'description' and 'comments'. We still a have a number of people 
who haven't ensured that their name is not in the description column.  There 
shouldn't be a need for the name to appear may be 50 times in a column if 
people adjust their entry to include one or two of their 'contact' numbers 
in the description column.  The 'owner' of any code is available by clicking 
on the STD/Area code at the start of the number saving all the repetition 
and approx 10% of the screen area. The problem on the North American page 
(as with the European page) could soon be solved if the owner's name could 
be included against their main contact number.

Greg I think has our page set up OK now. The description column is limited 
to one line per number - so if the name is included in that column, it is 
easy to scan down the column to find a person. The 'Comments column can be 
more than one line hence it takes a bit more reading as you scroll down the 
page.

Even now as CNET expands here in the UK, the number of entries will only 
increase.  Currently we have 15 pages of numbers with more appearing as time 
goes on. Another Asterisk has come on line under test and probably more 
numbers with it.  Our different pattern of numbering probably accounts for 
the increase of codes/numbers in the UK.  The New Zealand page also includes 
all the names & contact numbers of  those involved bar one.

Ian Jolly

+44 (0)  352      82 26 (via a 1929 GPO Rural Automatic eXchange!) from 
CNET - the Heritage Telephone Network
+44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from  Public 
Telephone Network
FWD Telephone No  83 2230


----- Original Message ----- 
From: "Lee Spenadel" <lee at spenadel.com>
To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
Sent: Thursday, October 18, 2007 3:33 PM
Subject: Re: [VoIP] Office code assignment wiki hacked


> While this is on a slightly different note, is there a way to get the
> person's name back in the user listings on the directory pages?  I find it
> difficult to look up another CNET member and get their number with the new
> page format.
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Greg Blakely
> Sent: Thursday, October 18, 2007 8:15 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Office code assignment wiki hacked
>
> We could certainly do some sort of behind-the-scenes authentication, and
> that may actually be the best route.  Instead of the 'reserve, register,
> and activate' scenario, it could end up being 'register, reserve, and
> activate.'
>
>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info]
>> On Behalf Of Ian Jolly
>> Sent: Thursday, October 18, 2007 4:09 AM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Office code assignment wiki hacked
>>
>> Might it be an idea such that only 'registered' CNET members
>> can get access to the WiKi ?
>>
>> After all those interested enough will have 'registered' so
>> we should have some idea as to who they are. In the UK all
>> the CNET members except one are know to at least one of the
>> main CNET users.  On this side of the 'pond', some of us had
>> thought that it was rather open to abuse.
>>
>> Shouldn't be too difficult to link the ckts.info password should it?
>>
>> Ian Jolly
>>
>> +44 (0)  352      82 26 (via a 1929 GPO Rural Automatic
>> eXchange!) from
>> CNET - the Heritage Telephone Network
>> +44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from
>> +Public
>> Telephone Network
>> FWD Telephone No  83 2230
>>
>>
>> ----- Original Message -----
>> From: "David Josephson" <david at josephson.com>
>> To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
>> Sent: Thursday, October 18, 2007 6:45 AM
>> Subject: [VoIP] Office code assignment wiki hacked
>>
>>
>> > Sometimes these things just happen, I guess two years is a
>> good record.
>> >
>> > The CNET wiki page for office code assignments was hacked
>> yesterday by a
>> > very determined robot, whi't just  appeared to be simultaneously in
>> > China, India and in unassigned IP address space. So, instead of
>> > www.altaphon.com/cgi-bin/f8n.pl?TelCollNet, change the f8n
>> to etz1k (the
>> > fourth character is a numeral 1) ... we have recovered the changes.
>> >
>> > And while I'm here, I was surprised last week to get an actual call
>> > through the crossbar exchange from a list member in San
>> Jose. I think
>> > this may have been the first actual CNET call to an actual listed
>> > extension, that wasn't just to a test number on the switch,
>> even though
>> > it's been online for almost a year. 555-2368 does indeed
>> ring on my desk
>> > (through a WE 202 set, no less) and it is not simulated
>> ringback tone.
>> >
>> > Cheers
>> >
>> > --
>> > David Josephson
>> > _______________________________________________
>> > VoIP mailing list
>> > VoIP at ckts.info
>> > http://lists.ckts.info/mailman/listinfo/voip
>> > Project Web Page: http://www.ckts.info/
>> >
>> >
>> > -- 
>> > No virus found in this incoming message.
>> > Checked by AVG Free Edition.
>> > Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date:
>> > 16/10/2007 14:14
>> >
>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.488 / Virus Database: 269.15.0/1076 - Release Date: 
> 17/10/2007 19:53
>
> 


From greg at vyger.net  Thu Oct 18 20:18:57 2007
From: greg at vyger.net (Greg Blakely)
Date: Thu, 18 Oct 2007 20:18:57 -0500
Subject: [VoIP] "Owner" column
Message-ID: <A691C3685A152E4AAC9608B8D8E1131668659D@godzilla.hickory.vyger.net>

I think it'd be nice to have a line drawn after all the entries of one
office code, and then a heading at the beginning of the next one that
includes the member's name.

I haven't yet figured out how to do it, but it'd be a fine idea, if I
ever do... 



From voiptandem at shaneyoung.com  Thu Oct 18 22:43:10 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Thu, 18 Oct 2007 22:43:10 -0500
Subject: [VoIP] Office code assignment wiki hacked
In-Reply-To: <4716F2EB.1080006@josephson.com>
References: <4716F2EB.1080006@josephson.com>
Message-ID: <20071018224310.ps2qs1tyecwcc8cg@secure.shaneyoung.com>

My script which does the simple status checking uses the data from  
this page.  Since it moved, my script broke and the status page was  
pretty hosed.

I've updated my stuff with the new URL and kicked it so the status  
page is back to being "normal"  :)


Quoting David Josephson <david at josephson.com>:

> Sometimes these things just happen, I guess two years is a good record.
>
> The CNET wiki page for office code assignments was hacked yesterday by a
> very determined robot, whi't just  appeared to be simultaneously in
> China, India and in unassigned IP address space. ringback tone.

--Shane
+1-821-7311 CNET

----------------------------------------------------------------


From dfroula at sbcglobal.net  Fri Oct 19 07:14:37 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 19 Oct 2007 05:14:37 -0700 (PDT)
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <47161600.10307@stromberg-carlson.org>
Message-ID: <285116.65435.qm@web83204.mail.mud.yahoo.com>

John, what application did you use to clone your
drives? I have several 6 Gbyte drives identical to the
one I hacked into my Wyse thin client, and would like
to clone the drive for easy recovery, if the drive
fails.

I've tried some of the free Linux code available, but
none have successfully created an image to my external
USB drive.

Thanks,

Don
--- John Novack <jnovack at stromberg-carlson.org> wrote:

> Boy, you Brits really know how to start a religious
> argument!!
> 
> My choice has been, for better or worse, CentOS.
> CentOS version 3 works well on 1.2 versions
> CentOS 4/5 seems to be required on the 1.4 versions
> So far I have seen no reason to move to 1.4, though
> some have for 
> various reasons
> Many on the Asterisk users list consider 1.4 still
> unstable, though I 
> doubt any of the Cnet uses will be affected.
> I always install everything ( moans from the Linux
> gurus here ) but I 
> have found it easier to install everything and then
> turn off services I 
> don't need, rather than later battle some cryptic
> message regarding a 
> missing dependency that I can't fix or know where to
> stick a file.
> I also use the full, rather than the server version,
> and do an update 
> from yum before proceeding to Asterisk.
> I have cloned several hard drives for others on CNET
> with fair success, 
> saving having to start from scratch every time.
> Easier to change a machine name and such than
> reinstall everything from 
> scratch.
> Acronis version 9 handles Linux disks well, even
> when the geometry isn't 
> the same.
> 
> Others can comment on the other religions. I suppose
> it is down to 
> whatever we have been imprinted with in our early
> Linux days, and what 
> we have become familiar with.
> I have dabbled in the Debian pool, but have not met
> with any success.
> 
> John Novack
> 
> 
> Jonathan Kay wrote:
> > Chaps.
> >  If you started from scratch now, what would be
> your Linux distro of 
> > choice for a purely Asterisk box?
> > I'm building a few machines to pass on at cost
> price.
> >
> > CentOS 4, was available in a "Server" edition. I
> had problems compiling 
> > the latest zaptel with that and had to upgrade
> kernel etc. CentOS 5, 
> > doesn't offer the server edition.
> >
> > Fedora 5, and 6, I think had kernel version
> issues, again I had lots of 
> > trouble compiling Asterisk with FC6. I don't know
> if 7 is ok?
> >
> > Ubuntu is great, I use desktop and server
> versions. But it is Debian 
> > based, and the community here and on the UK list
> tend to use flavours of 
> > Red Hat with Asterisk, so getting CNET support
> might be more tricky.
> >
> > Comments welcome !?
> >
> > Jon
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >   
> 
> -- 
> Dog is my co-pilot
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From dfroula at sbcglobal.net  Fri Oct 19 07:19:02 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 19 Oct 2007 05:19:02 -0700 (PDT)
Subject: [VoIP] Cisco 3810
Message-ID: <16016.84374.qm@web83212.mail.mud.yahoo.com>

I've decided to purchase a Cisco 3810 for some extra
FXS and FXO ports and to experiment with the T1
interface and some test equipment I have. It seems the
Ebay source for these has dried up, and the
information on memory and other options is difficult
to get on what I see offered by others. Any suitable
units gathering dust out there?

Thanks,

Don

From dfroula at sbcglobal.net  Fri Oct 19 07:24:52 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 19 Oct 2007 05:24:52 -0700 (PDT)
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <285116.65435.qm@web83204.mail.mud.yahoo.com>
Message-ID: <660716.40089.qm@web83203.mail.mud.yahoo.com>

Hmmm....Just saw it in you original post. Thanks!

Don

--- Donald Froula <dfroula at sbcglobal.net> wrote:

> > Acronis version 9 handles Linux disks well, even
> > when the geometry isn't 
> > the same.
> > 


From jnovack at stromberg-carlson.org  Fri Oct 19 08:14:13 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 19 Oct 2007 09:14:13 -0400
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <285116.65435.qm@web83204.mail.mud.yahoo.com>
References: <285116.65435.qm@web83204.mail.mud.yahoo.com>
Message-ID: <4718ADA5.2080401@stromberg-carlson.org>

I use Acronis True Image version 9.0
I am not sure if it works with USB drives, though
I use a PIII that can boot from CD-ROM, booting the Acronis disk created 
once you install it on a Windows machine, then it sees two IDE drives 
and you can go from there.
It handles different geometry, and will expand to a larger drive 
proportionately, or you can manually adjust partition sizes.
The cloned drive should boot and appear as if it was the original, if 
all goes well.
I have had problems with the software on some Pentium IV machines.

Version 9 is not a current version, so I cannot speak to the later ones.
Other cloning software doesn't handle Linux drives properly if at all

John Novack

Donald Froula wrote:
> John, what application did you use to clone your
> drives? I have several 6 Gbyte drives identical to the
> one I hacked into my Wyse thin client, and would like
> to clone the drive for easy recovery, if the drive
> fails.
>
> I've tried some of the free Linux code available, but
> none have successfully created an image to my external
> USB drive.
>
> Thanks,
>
> Don
> --- John Novack <jnovack at stromberg-carlson.org> wrote:
>
>   
>> Boy, you Brits really know how to start a religious
>> argument!!
>>
>> My choice has been, for better or worse, CentOS.
>> CentOS version 3 works well on 1.2 versions
>> CentOS 4/5 seems to be required on the 1.4 versions
>> So far I have seen no reason to move to 1.4, though
>> some have for 
>> various reasons
>> Many on the Asterisk users list consider 1.4 still
>> unstable, though I 
>> doubt any of the Cnet uses will be affected.
>> I always install everything ( moans from the Linux
>> gurus here ) but I 
>> have found it easier to install everything and then
>> turn off services I 
>> don't need, rather than later battle some cryptic
>> message regarding a 
>> missing dependency that I can't fix or know where to
>> stick a file.
>> I also use the full, rather than the server version,
>> and do an update 
>> from yum before proceeding to Asterisk.
>> I have cloned several hard drives for others on CNET
>> with fair success, 
>> saving having to start from scratch every time.
>> Easier to change a machine name and such than
>> reinstall everything from 
>> scratch.
>> Acronis version 9 handles Linux disks well, even
>> when the geometry isn't 
>> the same.
>>
>> Others can comment on the other religions. I suppose
>> it is down to 
>> whatever we have been imprinted with in our early
>> Linux days, and what 
>> we have become familiar with.
>> I have dabbled in the Debian pool, but have not met
>> with any success.
>>
>> John Novack
>>
>>
>> Jonathan Kay wrote:
>>     
>>> Chaps.
>>>  If you started from scratch now, what would be
>>>       
>> your Linux distro of 
>>     
>>> choice for a purely Asterisk box?
>>> I'm building a few machines to pass on at cost
>>>       
>> price.
>>     
>>> CentOS 4, was available in a "Server" edition. I
>>>       
>> had problems compiling 
>>     
>>> the latest zaptel with that and had to upgrade
>>>       
>> kernel etc. CentOS 5, 
>>     
>>> doesn't offer the server edition.
>>>
>>> Fedora 5, and 6, I think had kernel version
>>>       
>> issues, again I had lots of 
>>     
>>> trouble compiling Asterisk with FC6. I don't know
>>>       
>> if 7 is ok?
>>     
>>> Ubuntu is great, I use desktop and server
>>>       
>> versions. But it is Debian 
>>     
>>> based, and the community here and on the UK list
>>>       
>> tend to use flavours of 
>>     
>>> Red Hat with Asterisk, so getting CNET support
>>>       
>> might be more tricky.
>>     
>>> Comments welcome !?
>>>
>>> Jon
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>   
>>>       
>> -- 
>> Dog is my co-pilot
>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>     
>
>
>   

-- 
Dog is my co-pilot


From voiptandem at shaneyoung.com  Fri Oct 19 08:24:08 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Fri, 19 Oct 2007 08:24:08 -0500
Subject: [VoIP] Fedora, CentOS or What?
In-Reply-To: <4718ADA5.2080401@stromberg-carlson.org>
References: <285116.65435.qm@web83204.mail.mud.yahoo.com>
	<4718ADA5.2080401@stromberg-carlson.org>
Message-ID: <20071019082408.89yw8a00k0cws4oc@secure.shaneyoung.com>

With identical drives, it's easy to do in Linux.  I've never done it  
with drives of different geometry.

I would do it this way (though it's been a long time since I've done it)

Have a system which is not booting from the drive you want to clone.

Attach the source drive as the primary on your second IDE controller.
Attach the destination drive as the secondary on your second IDE controller.

Boot the system and log in.

Type dd if=/dev/hdc of=/dev/hdd

This should copy the formating, partitioning, file structure  
(everything, really) from the first drive to the second drive.


I used to do this a lot when I needed to create a bootable floppy disk  
from a disk image.



Quoting John Novack <jnovack at stromberg-carlson.org>:

> I use Acronis True Image version 9.0
> I am not sure if it works with USB drives, though
> I use a PIII that can boot from CD-ROM, booting the Acronis disk created
> once you install it on a Windows machine, then it sees two IDE drives
> and you can go from there.
> It handles different geometry, and will expand to a larger drive
> proportionately, or you can manually adjust partition sizes.
> The cloned drive should boot and appear as if it was the original, if
> all goes well.
> I have had problems with the software on some Pentium IV machines.
>
> Version 9 is not a current version, so I cannot speak to the later ones.
> Other cloning software doesn't handle Linux drives properly if at all
>
> John Novack
>
> Donald Froula wrote:
>> John, what application did you use to clone your
>> drives? I have several 6 Gbyte drives identical to the
>> one I hacked into my Wyse thin client, and would like
>> to clone the drive for easy recovery, if the drive
>> fails.
>>
>> I've tried some of the free Linux code available, but
>> none have successfully created an image to my external
>> USB drive.
>>
>> Thanks,
>>
>> Don
>> --- John Novack <jnovack at stromberg-carlson.org> wrote:
>>
>>
>>> Boy, you Brits really know how to start a religious
>>> argument!!
>>>
>>> My choice has been, for better or worse, CentOS.
>>> CentOS version 3 works well on 1.2 versions
>>> CentOS 4/5 seems to be required on the 1.4 versions
>>> So far I have seen no reason to move to 1.4, though
>>> some have for
>>> various reasons
>>> Many on the Asterisk users list consider 1.4 still
>>> unstable, though I
>>> doubt any of the Cnet uses will be affected.
>>> I always install everything ( moans from the Linux
>>> gurus here ) but I
>>> have found it easier to install everything and then
>>> turn off services I
>>> don't need, rather than later battle some cryptic
>>> message regarding a
>>> missing dependency that I can't fix or know where to
>>> stick a file.
>>> I also use the full, rather than the server version,
>>> and do an update
>>> from yum before proceeding to Asterisk.
>>> I have cloned several hard drives for others on CNET
>>> with fair success,
>>> saving having to start from scratch every time.
>>> Easier to change a machine name and such than
>>> reinstall everything from
>>> scratch.
>>> Acronis version 9 handles Linux disks well, even
>>> when the geometry isn't
>>> the same.
>>>
>>> Others can comment on the other religions. I suppose
>>> it is down to
>>> whatever we have been imprinted with in our early
>>> Linux days, and what
>>> we have become familiar with.
>>> I have dabbled in the Debian pool, but have not met
>>> with any success.
>>>
>>> John Novack
>>>
>>>
>>> Jonathan Kay wrote:
>>>
>>>> Chaps.
>>>>  If you started from scratch now, what would be
>>>>
>>> your Linux distro of
>>>
>>>> choice for a purely Asterisk box?
>>>> I'm building a few machines to pass on at cost
>>>>
>>> price.
>>>
>>>> CentOS 4, was available in a "Server" edition. I
>>>>
>>> had problems compiling
>>>
>>>> the latest zaptel with that and had to upgrade
>>>>
>>> kernel etc. CentOS 5,
>>>
>>>> doesn't offer the server edition.
>>>>
>>>> Fedora 5, and 6, I think had kernel version
>>>>
>>> issues, again I had lots of
>>>
>>>> trouble compiling Asterisk with FC6. I don't know
>>>>
>>> if 7 is ok?
>>>
>>>> Ubuntu is great, I use desktop and server
>>>>
>>> versions. But it is Debian
>>>
>>>> based, and the community here and on the UK list
>>>>
>>> tend to use flavours of
>>>
>>>> Red Hat with Asterisk, so getting CNET support
>>>>
>>> might be more tricky.
>>>
>>>> Comments welcome !?
>>>>
>>>> Jon
>>>> _______________________________________________
>>>> VoIP mailing list
>>>> VoIP at ckts.info
>>>> http://lists.ckts.info/mailman/listinfo/voip
>>>> Project Web Page: http://www.ckts.info/
>>>>
>>>>
>>>>
>>> --
>>> Dog is my co-pilot
>>>
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>
>>
>>
>>
>
> --
> Dog is my co-pilot
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

--Shane
+1-821-7311 CNET

----------------------------------------------------------------


From jjones3601 at yahoo.com  Fri Oct 19 08:31:38 2007
From: jjones3601 at yahoo.com (john jones)
Date: Fri, 19 Oct 2007 06:31:38 -0700 (PDT)
Subject: [VoIP] Cisco 3810
Message-ID: <979655.49143.qm@web34307.mail.mud.yahoo.com>

This one will work but doesn't have many FXS/FXO ports

http://cgi.ebay.com/New-open-box-Cisco-MC3810-V3-MC3810-MFT-T1-modules_W0QQitemZ180171161173QQihZ008QQcategoryZ73321QQssPageNameZWDVWQQrdZ1QQcmdZViewItem

This was the source.  Maybe ask the seller if he has any more.

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=140123501044&ssPageName=STRK:MEWA:IT&ih=004

John

----- Original Message ----
From: Donald Froula <dfroula at sbcglobal.net>
To: voip at ckts.info
Sent: Friday, October 19, 2007 8:19:02 AM
Subject: [VoIP] Cisco 3810


I've decided to purchase a Cisco 3810 for some extra
FXS and FXO ports and to experiment with the T1
interface and some test equipment I have. It seems the
Ebay source for these has dried up, and the
information on memory and other options is difficult
to get on what I see offered by others. Any suitable
units gathering dust out there?

Thanks,

Don
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/




From jjones3601 at yahoo.com  Fri Oct 19 08:38:01 2007
From: jjones3601 at yahoo.com (john jones)
Date: Fri, 19 Oct 2007 06:38:01 -0700 (PDT)
Subject: [VoIP] Office code assignment wiki hacked
Message-ID: <759172.49745.qm@web34301.mail.mud.yahoo.com>

Shane,

Can I get a copy of your updated script?

Thanks!

John

----- Original Message ----
From: Shane Young <voiptandem at shaneyoung.com>
To: voip at ckts.info
Sent: Thursday, October 18, 2007 11:43:10 PM
Subject: Re: [VoIP] Office code assignment wiki hacked


My script which does the simple status checking uses the data from  
this page.  Since it moved, my script broke and the status page was  
pretty hosed.

I've updated my stuff with the new URL and kicked it so the status  
page is back to being "normal"  :)


Quoting David Josephson <david at josephson.com>:

> Sometimes these things just happen, I guess two years is a good
 record.
>
> The CNET wiki page for office code assignments was hacked yesterday
 by a
> very determined robot, whi't just  appeared to be simultaneously in
> China, India and in unassigned IP address space. ringback tone.

--Shane
+1-821-7311 CNET

----------------------------------------------------------------

_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/




From jnovack at stromberg-carlson.org  Fri Oct 19 08:38:23 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 19 Oct 2007 09:38:23 -0400
Subject: [VoIP] Cisco 3810
In-Reply-To: <16016.84374.qm@web83212.mail.mud.yahoo.com>
References: <16016.84374.qm@web83212.mail.mud.yahoo.com>
Message-ID: <4718B34F.8040304@stromberg-carlson.org>

Do a search on MC3810

Best choice:
MC3810V-3
MC3810 and MC3810V can't take 64 K memory
64K ram
16K flash
AVM
some number of FXO/FXS/E&M
Some of us have either surplus port boards or are willing to do a swap.
Just make sure the AVM is there.
I don't remember now who was accepting a best offer of $75 plus 
shipping, perhaps someone else does. There may still be more there, just 
not listed.
Memory and flash upgrades are too expensive, so be patient and set up a 
search and have eBay e-mail you every day there are new listings, and 
something will show up soon.

John Novack


Donald Froula wrote:
> I've decided to purchase a Cisco 3810 for some extra
> FXS and FXO ports and to experiment with the T1
> interface and some test equipment I have. It seems the
> Ebay source for these has dried up, and the
> information on memory and other options is difficult
> to get on what I see offered by others. Any suitable
> units gathering dust out there?
>
> Thanks,
>
> Don
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From dfroula at sbcglobal.net  Fri Oct 19 08:48:42 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 19 Oct 2007 06:48:42 -0700 (PDT)
Subject: [VoIP] Cisco 3810
In-Reply-To: <4718B34F.8040304@stromberg-carlson.org>
Message-ID: <495708.39961.qm@web83208.mail.mud.yahoo.com>

Thanks!

I've done all that. I've written the guy who had the
boxes a few weeks ago, but no reply.

Thanks for the information on the older units not
accepting the larger memory upgrade. I understand that
16 Mbytes is required for th 12.3 image that has SIP
functionality, so that seems important.

Don

--- John Novack <jnovack at stromberg-carlson.org> wrote:

> Do a search on MC3810
> 
> Best choice:
> MC3810V-3
> MC3810 and MC3810V can't take 64 K memory
> 64K ram
> 16K flash
> AVM
> some number of FXO/FXS/E&M
> Some of us have either surplus port boards or are
> willing to do a swap.
> Just make sure the AVM is there.
> I don't remember now who was accepting a best offer
> of $75 plus 
> shipping, perhaps someone else does. There may still
> be more there, just 
> not listed.
> Memory and flash upgrades are too expensive, so be
> patient and set up a 
> search and have eBay e-mail you every day there are
> new listings, and 
> something will show up soon.
> 
> John Novack
> 
> 
> Donald Froula wrote:
> > I've decided to purchase a Cisco 3810 for some
> extra
> > FXS and FXO ports and to experiment with the T1
> > interface and some test equipment I have. It seems
> the
> > Ebay source for these has dried up, and the
> > information on memory and other options is
> difficult
> > to get on what I see offered by others. Any
> suitable
> > units gathering dust out there?
> >
> > Thanks,
> >
> > Don
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >   
> 
> -- 
> Dog is my co-pilot
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From dfroula at sbcglobal.net  Fri Oct 19 08:54:32 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 19 Oct 2007 06:54:32 -0700 (PDT)
Subject: [VoIP] 2600 Flashing Supervision
Message-ID: <733384.26606.qm@web83212.mail.mud.yahoo.com>

I added a flashing 2600 test number to my box at:

1-762-2600/2601
<2600> KP+199+ST

This number plays back 2600 at .5 second on/.5 second
off for 8 seconds, then .25 on/.25 off for another 60
seconds, kind of like the old supervision tests.

When dialing direct on Asterisk from a local extension
(not through the SF trunk group), it sounds like 2600
Hz. busy and reorder.

However, when dialing through the SF trunks, one can
clearly hear the action of the SF notch filter.

What one hears is a cheep of 2600 when the tone is
applied, another cheep when the tone is removed, and
the old warbling noise in-between. The warbling is due
to the rather high level of 2600 and imperfect tone
filtering of the notch filter.

This confirms that the cheeps we sometimes hear while
a call completes and supervision is returned through
ProjectMF are not real winks, but artifacts of the
tone filter operation.

Note that the reverse 2600 will not blow off the call,
as I have the near-end trunk group "goodied" through
the Zaptel settings. However, the notch filters still
work fine, making this observation possible.

Best,

Don



From dfroula at sbcglobal.net  Fri Oct 19 09:14:15 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 19 Oct 2007 07:14:15 -0700 (PDT)
Subject: [VoIP] 2600 Flashing Supervision
In-Reply-To: <733384.26606.qm@web83212.mail.mud.yahoo.com>
Message-ID: <455447.33902.qm@web83212.mail.mud.yahoo.com>

I set up a little demo of this at some additional
numbers at my extension:

1-762-0198 plays the flashing 2600 direct, not through
the SF-controlled trunks.

1-762-0199 routes the call through the SF-controlled
trunk group, then to the flashing 2600. The notch
filter action can be heard.

No Blue Box required!

Don

--- Donald Froula <dfroula at sbcglobal.net> wrote:

> I added a flashing 2600 test number to my box at:
> 
> 1-762-2600/2601
> <2600> KP+199+ST
> 
> This number plays back 2600 at .5 second on/.5
> second
> off for 8 seconds, then .25 on/.25 off for another
> 60
> seconds, kind of like the old supervision tests.
> 
> When dialing direct on Asterisk from a local
> extension
> (not through the SF trunk group), it sounds like
> 2600
> Hz. busy and reorder.
> 
> However, when dialing through the SF trunks, one can
> clearly hear the action of the SF notch filter.
> 
> What one hears is a cheep of 2600 when the tone is
> applied, another cheep when the tone is removed, and
> the old warbling noise in-between. The warbling is
> due
> to the rather high level of 2600 and imperfect tone
> filtering of the notch filter.
> 
> This confirms that the cheeps we sometimes hear
> while
> a call completes and supervision is returned through
> ProjectMF are not real winks, but artifacts of the
> tone filter operation.
> 
> Note that the reverse 2600 will not blow off the
> call,
> as I have the near-end trunk group "goodied" through
> the Zaptel settings. However, the notch filters
> still
> work fine, making this observation possible.
> 
> Best,
> 
> Don
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From stfkerman at jps.net  Fri Oct 19 15:08:48 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 19 Oct 2007 16:08:48 -0400
Subject: [VoIP] Cisco 3810
In-Reply-To: <4718B34F.8040304@stromberg-carlson.org>
References: <16016.84374.qm@web83212.mail.mud.yahoo.com>
	<4718B34F.8040304@stromberg-carlson.org>
Message-ID: <47190ED0.1040100@jps.net>

How does one identify with certainty whether a unit is a MC3810V-3, 
MC3810V or MC3810? 

Is there a reliable ID label or is it possible for a newer housing to 
contain an older circuit board? 

Are there mechanical differences that would permit reliable identification?

Can one rely on a sign on message at the console or is the sign on 
message generic and not indicate the version?

I assume JN meant 16MB flash.  Is this the rare case of a safe assumption?

Steph

John Novack wrote:
> Do a search on MC3810
>
> Best choice:
> MC3810V-3
> MC3810 and MC3810V can't take 64 K memory
> 64K ram
> 16K flash
> AVM
> some number of FXO/FXS/E&M
> Some of us have either surplus port boards or are willing to do a swap.
> Just make sure the AVM is there.
> I don't remember now who was accepting a best offer of $75 plus 
> shipping, perhaps someone else does. There may still be more there, just 
> not listed.
> Memory and flash upgrades are too expensive, so be patient and set up a 
> search and have eBay e-mail you every day there are new listings, and 
> something will show up soon.
>
> John Novack
>
>
> Donald Froula wrote:
>   
>> I've decided to purchase a Cisco 3810 for some extra
>> FXS and FXO ports and to experiment with the T1
>> interface and some test equipment I have. It seems the
>> Ebay source for these has dried up, and the
>> information on memory and other options is difficult
>> to get on what I see offered by others. Any suitable
>> units gathering dust out there?
>>
>> Thanks,
>>
>> Don
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>   
>>     
>
>   

From jjones3601 at yahoo.com  Fri Oct 19 15:17:53 2007
From: jjones3601 at yahoo.com (john jones)
Date: Fri, 19 Oct 2007 13:17:53 -0700 (PDT)
Subject: [VoIP] Cisco 3810
Message-ID: <985077.5447.qm@web34307.mail.mud.yahoo.com>

The physical difference are unreliable.  The outside pictures may show the part number.  A V3 version IMHO is guaranteed to have a boot prom which will recognize 16M flash ( You are correct, this is 16M not 16k) and 64M DRAM.  Unless you see a "show version" output on a pre-V3 router, you may not be able to upgrade the memory. The boot prom seems impossible to get.

Your best approach is to see the output of the show ver.  It not only shows the memory, but also the analog modules installed.  Note that the 16k of config register memory is fixed and is always 16k so no need to worry about it.



cisco MC3810 (MPC860) processor (revision 06.07) with 61440K/4096K bytes of memory.
Processor board ID 10118324
PPC860 PowerQUICC, partnum 0x0000, version A03(0x0013)
Channelized E1, Version 1.0.
Bridging software.
X.25 software, Version 3.0.0.
SuperLAT software (copyright 1990 by Meridian Technology Corp).
TN3270 Emulation software.
MC3810 SCB board (v05.A0)
1 Six-Slot Analog Voice Module (v03.K0)
1    Analog FXS voice interface (v03.K0) port 1/1
1    Analog FXS voice interface (v03.K0) port 1/2
1    Analog E&M voice interface (v03.K0) port 1/4
1    Analog FXO voice interface (v03.K0) port 1/6
1 3-DSP(slot2) Voice Compression Module(v01.A0)
1 Ethernet/IEEE 802.3 interface(s)
1 Serial network interface(s)
2 Serial(sync/async) network interface(s)
256K bytes of non-volatile configuration memory.
16384K bytes of processor board System flash (AMD29F016) 

Configuration register is 0x2102

Hope this helps!

John


----- Original Message ----
From: Steph Kerman <stfkerman at jps.net>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Friday, October 19, 2007 4:08:48 PM
Subject: Re: [VoIP] Cisco 3810


How does one identify with certainty whether a unit is a MC3810V-3, 
MC3810V or MC3810? 

Is there a reliable ID label or is it possible for a newer housing to 
contain an older circuit board? 

Are there mechanical differences that would permit reliable
 identification?

Can one rely on a sign on message at the console or is the sign on 
message generic and not indicate the version?

I assume JN meant 16MB flash.  Is this the rare case of a safe
 assumption?

Steph

John Novack wrote:
> Do a search on MC3810
>
> Best choice:
> MC3810V-3
> MC3810 and MC3810V can't take 64 K memory
> 64K ram
> 16K flash
> AVM
> some number of FXO/FXS/E&M
> Some of us have either surplus port boards or are willing to do a
 swap.
> Just make sure the AVM is there.
> I don't remember now who was accepting a best offer of $75 plus 
> shipping, perhaps someone else does. There may still be more there,
 just 
> not listed.
> Memory and flash upgrades are too expensive, so be patient and set up
 a 
> search and have eBay e-mail you every day there are new listings, and
 
> something will show up soon.
>
> John Novack
>
>
> Donald Froula wrote:
>   
>> I've decided to purchase a Cisco 3810 for some extra
>> FXS and FXO ports and to experiment with the T1
>> interface and some test equipment I have. It seems the
>> Ebay source for these has dried up, and the
>> information on memory and other options is difficult
>> to get on what I see offered by others. Any suitable
>> units gathering dust out there?
>>
>> Thanks,
>>
>> Don
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>   
>>     
>
>   
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/




From stfkerman at jps.net  Fri Oct 19 15:39:34 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 19 Oct 2007 16:39:34 -0400
Subject: [VoIP] Cisco 3810
In-Reply-To: <985077.5447.qm@web34307.mail.mud.yahoo.com>
References: <985077.5447.qm@web34307.mail.mud.yahoo.com>
Message-ID: <47191606.3020504@jps.net>

john jones wrote:
> The physical difference are unreliable. The outside pictures may show 
> the part number. A V3 version IMHO is guaranteed to have a boot prom 
> which will recognize 16M flash ( You are correct, this is 16M not 16k) 
> and 64M DRAM. Unless you see a "show version" output on a pre-V3 
> router, you may not be able to upgrade the memory. The boot prom seems 
> impossible to get.
What is the part #?  Is it socketed?  Why can't it be copied?
> Your best approach is to see the output of the show ver. It not only 
> shows the memory, but also the analog modules installed. Note that the 
> 16k of config register memory is fixed and is always 16k so no need to 
> worry about it.
Thanks John!
Steph
> cisco MC3810 (MPC860) processor (revision 06.07) with 61440K/4096K 
> bytes of memory.
> Processor board ID 10118324
> PPC860 PowerQUICC, partnum 0x0000, version A03(0x0013)
> Channelized E1, Version 1.0.
> Bridging software.
> X.25 software, Version 3.0.0.
> SuperLAT software (copyright 1990 by Meridian Technology Corp).
> TN3270 Emulation software.
> MC3810 SCB board (v05.A0)
> 1 Six-Slot Analog Voice Module (v03.K0)
> 1 Analog FXS voice interface (v03.K0) port 1/1
> 1 Analog FXS voice interface (v03.K0) port 1/2
> 1 Analog E&M voice interface (v03.K0) port 1/4
> 1 Analog FXO voice interface (v03.K0) port 1/6
> 1 3-DSP(slot2) Voice Compression Module(v01.A0)
> 1 Ethernet/IEEE 802.3 interface(s)
> 1 Serial network interface(s)
> 2 Serial(sync/async) network interface(s)
> 256K bytes of non-volatile configuration memory.
> 16384K bytes of processor board System flash (AMD29F016)
>
> Configuration register is 0x2102
>
> Hope this helps!
>
> John
>
>
> ----- Original Message ----
> From: Steph Kerman <stfkerman at jps.net>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Friday, October 19, 2007 4:08:48 PM
> Subject: Re: [VoIP] Cisco 3810
>
>
> How does one identify with certainty whether a unit is a MC3810V-3,
> MC3810V or MC3810?
>
> Is there a reliable ID label or is it possible for a newer housing to
> contain an older circuit board?
>
> Are there mechanical differences that would permit reliable
> identification?
>
> Can one rely on a sign on message at the console or is the sign on
> message generic and not indicate the version?
>
> I assume JN meant 16MB flash. Is this the rare case of a safe
> assumption?
>
> Steph
>
> John Novack wrote:
>> Do a search on MC3810
>>
>> Best choice:
>> MC3810V-3
>> MC3810 and MC3810V can't take 64 K memory
>> 64K ram
>> 16K flash
>> AVM
>> some number of FXO/FXS/E&M
>> Some of us have either surplus port boards or are willing to do a
>>     
>  swap.
>   
>> Just make sure the AVM is there.
>> I don't remember now who was accepting a best offer of $75 plus 
>> shipping, perhaps someone else does. There may still be more there,
>>     
>  just 
>   
>> not listed.
>> Memory and flash upgrades are too expensive, so be patient and set up
>>     
>  a 
>   
>> search and have eBay e-mail you every day there are new listings, and
>>     
>  
>   
>> something will show up soon.
>>
>> John Novack
>>
>>
>> Donald Froula wrote:
>>   
>>     
>>> I've decided to purchase a Cisco 3810 for some extra
>>> FXS and FXO ports and to experiment with the T1
>>> interface and some test equipment I have. It seems the
>>> Ebay source for these has dried up, and the
>>> information on memory and other options is difficult
>>> to get on what I see offered by others. Any suitable
>>> units gathering dust out there?
>>>
>>> Thanks,
>>>
>>> Don
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>   
>>>     
>>>       
>>   
>>     
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From jjones3601 at yahoo.com  Fri Oct 19 15:52:01 2007
From: jjones3601 at yahoo.com (john jones)
Date: Fri, 19 Oct 2007 13:52:01 -0700 (PDT)
Subject: [VoIP] Cisco 3810
Message-ID: <477873.82068.qm@web34303.mail.mud.yahoo.com>

It is socketed and the part number is something like BOOT-381 .  It is a PLCC format so I guess it could be copied but I don't know anybody with a PLCC copier.   If you google for MC-3810 memory, I'm sure you'll find the part number.

John

----- Original Message ----
From: Steph Kerman <stfkerman at jps.net>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Friday, October 19, 2007 4:39:34 PM
Subject: Re: [VoIP] Cisco 3810


john jones wrote:
> The physical difference are unreliable. The outside pictures may show
 
> the part number. A V3 version IMHO is guaranteed to have a boot prom 
> which will recognize 16M flash ( You are correct, this is 16M not
 16k) 
> and 64M DRAM. Unless you see a "show version" output on a pre-V3 
> router, you may not be able to upgrade the memory. The boot prom
 seems 
> impossible to get.
What is the part #?  Is it socketed?  Why can't it be copied?
> Your best approach is to see the output of the show ver. It not only 
> shows the memory, but also the analog modules installed. Note that
 the 
> 16k of config register memory is fixed and is always 16k so no need
 to 
> worry about it.
Thanks John!
Steph
> cisco MC3810 (MPC860) processor (revision 06.07) with 61440K/4096K 
> bytes of memory.
> Processor board ID 10118324
> PPC860 PowerQUICC, partnum 0x0000, version A03(0x0013)
> Channelized E1, Version 1.0.
> Bridging software.
> X.25 software, Version 3.0.0.
> SuperLAT software (copyright 1990 by Meridian Technology Corp).
> TN3270 Emulation software.
> MC3810 SCB board (v05.A0)
> 1 Six-Slot Analog Voice Module (v03.K0)
> 1 Analog FXS voice interface (v03.K0) port 1/1
> 1 Analog FXS voice interface (v03.K0) port 1/2
> 1 Analog E&M voice interface (v03.K0) port 1/4
> 1 Analog FXO voice interface (v03.K0) port 1/6
> 1 3-DSP(slot2) Voice Compression Module(v01.A0)
> 1 Ethernet/IEEE 802.3 interface(s)
> 1 Serial network interface(s)
> 2 Serial(sync/async) network interface(s)
> 256K bytes of non-volatile configuration memory.
> 16384K bytes of processor board System flash (AMD29F016)
>
> Configuration register is 0x2102
>
> Hope this helps!
>
> John
>
>
> ----- Original Message ----
> From: Steph Kerman <stfkerman at jps.net>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Friday, October 19, 2007 4:08:48 PM
> Subject: Re: [VoIP] Cisco 3810
>
>
> How does one identify with certainty whether a unit is a MC3810V-3,
> MC3810V or MC3810?
>
> Is there a reliable ID label or is it possible for a newer housing to
> contain an older circuit board?
>
> Are there mechanical differences that would permit reliable
> identification?
>
> Can one rely on a sign on message at the console or is the sign on
> message generic and not indicate the version?
>
> I assume JN meant 16MB flash. Is this the rare case of a safe
> assumption?
>
> Steph
>
> John Novack wrote:
>> Do a search on MC3810
>>
>> Best choice:
>> MC3810V-3
>> MC3810 and MC3810V can't take 64 K memory
>> 64K ram
>> 16K flash
>> AVM
>> some number of FXO/FXS/E&M
>> Some of us have either surplus port boards or are willing to do a
>>     
>  swap.
>   
>> Just make sure the AVM is there.
>> I don't remember now who was accepting a best offer of $75 plus 
>> shipping, perhaps someone else does. There may still be more there,
>>     
>  just 
>   
>> not listed.
>> Memory and flash upgrades are too expensive, so be patient and set
 up
>>     
>  a 
>   
>> search and have eBay e-mail you every day there are new listings,
 and
>>     
>  
>   
>> something will show up soon.
>>
>> John Novack
>>
>>
>> Donald Froula wrote:
>>   
>>     
>>> I've decided to purchase a Cisco 3810 for some extra
>>> FXS and FXO ports and to experiment with the T1
>>> interface and some test equipment I have. It seems the
>>> Ebay source for these has dried up, and the
>>> information on memory and other options is difficult
>>> to get on what I see offered by others. Any suitable
>>> units gathering dust out there?
>>>
>>> Thanks,
>>>
>>> Don
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>   
>>>     
>>>       
>>   
>>     
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/




From dfroula at sbcglobal.net  Fri Oct 19 15:53:00 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 19 Oct 2007 13:53:00 -0700 (PDT)
Subject: [VoIP] Cisco 3810
In-Reply-To: <985077.5447.qm@web34307.mail.mud.yahoo.com>
Message-ID: <96822.57260.qm@web83207.mail.mud.yahoo.com>

Thanks, John. I'll resume my search armed with this
info!

Don

--- john jones <jjones3601 at yahoo.com> wrote:

> The physical difference are unreliable.  The outside
> pictures may show the part number.  A V3 version
> IMHO is guaranteed to have a boot prom which will
> recognize 16M flash ( You are correct, this is 16M
> not 16k) and 64M DRAM.  Unless you see a "show
> version" output on a pre-V3 router, you may not be
> able to upgrade the memory. The boot prom seems
> impossible to get.
> 
> Your best approach is to see the output of the show
> ver.  It not only shows the memory, but also the
> analog modules installed.  Note that the 16k of
> config register memory is fixed and is always 16k so
> no need to worry about it.
> 
> 
> 
> cisco MC3810 (MPC860) processor (revision 06.07)
> with 61440K/4096K bytes of memory.
> Processor board ID 10118324
> PPC860 PowerQUICC, partnum 0x0000, version
> A03(0x0013)
> Channelized E1, Version 1.0.
> Bridging software.
> X.25 software, Version 3.0.0.
> SuperLAT software (copyright 1990 by Meridian
> Technology Corp).
> TN3270 Emulation software.
> MC3810 SCB board (v05.A0)
> 1 Six-Slot Analog Voice Module (v03.K0)
> 1    Analog FXS voice interface (v03.K0) port 1/1
> 1    Analog FXS voice interface (v03.K0) port 1/2
> 1    Analog E&M voice interface (v03.K0) port 1/4
> 1    Analog FXO voice interface (v03.K0) port 1/6
> 1 3-DSP(slot2) Voice Compression Module(v01.A0)
> 1 Ethernet/IEEE 802.3 interface(s)
> 1 Serial network interface(s)
> 2 Serial(sync/async) network interface(s)
> 256K bytes of non-volatile configuration memory.
> 16384K bytes of processor board System flash
> (AMD29F016) 
> 
> Configuration register is 0x2102
> 
> Hope this helps!
> 
> John
> 
> 
> ----- Original Message ----
> From: Steph Kerman <stfkerman at jps.net>
> To: Voice Over IP Tandem for Analog Switches
> <voip at ckts.info>
> Sent: Friday, October 19, 2007 4:08:48 PM
> Subject: Re: [VoIP] Cisco 3810
> 
> 
> How does one identify with certainty whether a unit
> is a MC3810V-3, 
> MC3810V or MC3810? 
> 
> Is there a reliable ID label or is it possible for a
> newer housing to 
> contain an older circuit board? 
> 
> Are there mechanical differences that would permit
> reliable
>  identification?
> 
> Can one rely on a sign on message at the console or
> is the sign on 
> message generic and not indicate the version?
> 
> I assume JN meant 16MB flash.  Is this the rare case
> of a safe
>  assumption?
> 
> Steph
> 
> John Novack wrote:
> > Do a search on MC3810
> >
> > Best choice:
> > MC3810V-3
> > MC3810 and MC3810V can't take 64 K memory
> > 64K ram
> > 16K flash
> > AVM
> > some number of FXO/FXS/E&M
> > Some of us have either surplus port boards or are
> willing to do a
>  swap.
> > Just make sure the AVM is there.
> > I don't remember now who was accepting a best
> offer of $75 plus 
> > shipping, perhaps someone else does. There may
> still be more there,
>  just 
> > not listed.
> > Memory and flash upgrades are too expensive, so be
> patient and set up
>  a 
> > search and have eBay e-mail you every day there
> are new listings, and
>  
> > something will show up soon.
> >
> > John Novack
> >
> >
> > Donald Froula wrote:
> >   
> >> I've decided to purchase a Cisco 3810 for some
> extra
> >> FXS and FXO ports and to experiment with the T1
> >> interface and some test equipment I have. It
> seems the
> >> Ebay source for these has dried up, and the
> >> information on memory and other options is
> difficult
> >> to get on what I see offered by others. Any
> suitable
> >> units gathering dust out there?
> >>
> >> Thanks,
> >>
> >> Don
> >> _______________________________________________
> >> VoIP mailing list
> >> VoIP at ckts.info
> >> http://lists.ckts.info/mailman/listinfo/voip
> >> Project Web Page: http://www.ckts.info/
> >>
> >>   
> >>     
> >
> >   
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From donw at engineeringinc.com  Fri Oct 19 15:53:06 2007
From: donw at engineeringinc.com (Don E. Wisdom)
Date: Fri, 19 Oct 2007 14:53:06 -0600
Subject: [VoIP] Cisco 3810
In-Reply-To: <477873.82068.qm@web34303.mail.mud.yahoo.com>
References: <477873.82068.qm@web34303.mail.mud.yahoo.com>
Message-ID: <490F06C61AA78F4AB1FD2081EDE0ED7F1BCE36@backupsvr.corporate.engineeringinc.org>

If someone has access to a enterprise partner CCO account you can
download new bootrom images (mine isn't enterprise partner :( )

--Don










-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
Of john jones
Sent: Friday, October 19, 2007 2:52 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Cisco 3810

It is socketed and the part number is something like BOOT-381 .  It is a
PLCC format so I guess it could be copied but I don't know anybody with
a PLCC copier.   If you google for MC-3810 memory, I'm sure you'll find
the part number.

John

----- Original Message ----
From: Steph Kerman <stfkerman at jps.net>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Friday, October 19, 2007 4:39:34 PM
Subject: Re: [VoIP] Cisco 3810


john jones wrote:
> The physical difference are unreliable. The outside pictures may show
 
> the part number. A V3 version IMHO is guaranteed to have a boot prom 
> which will recognize 16M flash ( You are correct, this is 16M not
 16k) 
> and 64M DRAM. Unless you see a "show version" output on a pre-V3 
> router, you may not be able to upgrade the memory. The boot prom
 seems 
> impossible to get.
What is the part #?  Is it socketed?  Why can't it be copied?
> Your best approach is to see the output of the show ver. It not only 
> shows the memory, but also the analog modules installed. Note that
 the 
> 16k of config register memory is fixed and is always 16k so no need
 to 
> worry about it.
Thanks John!
Steph
> cisco MC3810 (MPC860) processor (revision 06.07) with 61440K/4096K 
> bytes of memory.
> Processor board ID 10118324
> PPC860 PowerQUICC, partnum 0x0000, version A03(0x0013)
> Channelized E1, Version 1.0.
> Bridging software.
> X.25 software, Version 3.0.0.
> SuperLAT software (copyright 1990 by Meridian Technology Corp).
> TN3270 Emulation software.
> MC3810 SCB board (v05.A0)
> 1 Six-Slot Analog Voice Module (v03.K0)
> 1 Analog FXS voice interface (v03.K0) port 1/1
> 1 Analog FXS voice interface (v03.K0) port 1/2
> 1 Analog E&M voice interface (v03.K0) port 1/4
> 1 Analog FXO voice interface (v03.K0) port 1/6
> 1 3-DSP(slot2) Voice Compression Module(v01.A0)
> 1 Ethernet/IEEE 802.3 interface(s)
> 1 Serial network interface(s)
> 2 Serial(sync/async) network interface(s)
> 256K bytes of non-volatile configuration memory.
> 16384K bytes of processor board System flash (AMD29F016)
>
> Configuration register is 0x2102
>
> Hope this helps!
>
> John
>
>
> ----- Original Message ----
> From: Steph Kerman <stfkerman at jps.net>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Friday, October 19, 2007 4:08:48 PM
> Subject: Re: [VoIP] Cisco 3810
>
>
> How does one identify with certainty whether a unit is a MC3810V-3,
> MC3810V or MC3810?
>
> Is there a reliable ID label or is it possible for a newer housing to
> contain an older circuit bo