From dheimer at plantationcable.net  Sat Mar  1 00:24:48 2008
From: dheimer at plantationcable.net (Dave H.)
Date: Sat, 1 Mar 2008 01:24:48 -0500
Subject: [VoIP] call to my *box a short while ago.
References: <47C8BECF.7010003@topletter.com>
Message-ID: <000601c87b64$f353f190$0201a8c0@your94e826b122>

Yo' Brian,

Thanx for the email.  Sorry, I was a bit confused when you answered.  I had 
a buddy of mine over with his two kids and was showing-off the telephone 
stuph and was dialing "mind boggleing"  & was kinda at a loss for words whan 
a human answered.  To further confuse thing I though that you were Ian Jolly 
(no idea why)  The thing cut off in mid sentence.  I'll give ya a holler 
tomorrow at a decent hour.
Apologies & Regardz, Dave H.

Dave Heimer
2160 Armor Bridge Road
Greensboro, GA 30642
706-467-3926 - Home
706-318-9577 - Cell
dheimer at plantationcable.net
www.heimer.siteblast.com
----- Original Message ----- 
From: "windmill" <windmill at topletter.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Friday, February 29, 2008 9:26 PM
Subject: [VoIP] call to my *box a short while ago.


>
>
> Hi there,
>
> I had a call a short while ago as follows;
>
> caller is "Dave Heimer" <2792275> called extension is 4452271696
>
> So hello Dave, I'm not sure exactly what happened but after I started to
> explain how you reached me instead of what you dialled my phone went dead!
>
> Firstly may I say that you did not disturb me, I was watching TV, but as
> stated in my directory entries I will answer 24/7 if I am awake and hear
> the phone ringing, it is not a problem at all.
>
> You did not misdial, currently there are just two lines available on my
> *box, the one that you dialled and 4452271501 which is me in person. The
> situation is that I am rewriting my dialplan from the ground up as it
> has become a sprawl of vast proportions with more than 100 extensions so
> at the moment while the new dialplan is under construction any calls
> reaching my *box from CNET get stuffed into a [temporary] context where
> they are switched alternately to 501 or 696 !
>
> So if you are calling 696 and get ringing instead just hang up and
> redial if you don't want to sya 'hello'!
>
> Brian
> _______________________________________________
> 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: 7.5.516 / Virus Database: 269.21.2/1304 - Release Date: 2/29/2008 
> 8:18 AM
>
> 


From ian at uax.org.uk  Sat Mar  1 02:49:27 2008
From: ian at uax.org.uk (Ian Jolly)
Date: Sat, 1 Mar 2008 08:49:27 -0000
Subject: [VoIP] call to my *box a short while ago.
In-Reply-To: <000601c87b64$f353f190$0201a8c0@your94e826b122>
References: <47C8BECF.7010003@topletter.com>
	<000601c87b64$f353f190$0201a8c0@your94e826b122>
Message-ID: <6134D5C6A6594EB7B7795CCF5CE2EFAA@IanPC>

"To further confuse thing I though that you were Ian Jolly (no idea why)! "

Oh dear !   What have I done to be confused with Brian  :-))


----- Original Message ----- 
From: "Dave H." <dheimer at plantationcable.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Saturday, March 01, 2008 6:24 AM
Subject: Re: [VoIP] call to my *box a short while ago.


> Yo' Brian,
>
> Thanx for the email.  Sorry, I was a bit confused when you answered.  I 
> had
> a buddy of mine over with his two kids and was showing-off the telephone
> stuph and was dialing "mind boggleing"  & was kinda at a loss for words 
> whan
> a human answered.  To further confuse thing I though that you were Ian 
> Jolly
> (no idea why)  The thing cut off in mid sentence.  I'll give ya a holler
> tomorrow at a decent hour.
> Apologies & Regardz, Dave H.
>
> Dave Heimer
> 2160 Armor Bridge Road
> Greensboro, GA 30642
> 706-467-3926 - Home
> 706-318-9577 - Cell
> dheimer at plantationcable.net
> www.heimer.siteblast.com
> ----- Original Message ----- 
> From: "windmill" <windmill at topletter.com>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Friday, February 29, 2008 9:26 PM
> Subject: [VoIP] call to my *box a short while ago.
>
>
>>
>>
>> Hi there,
>>
>> I had a call a short while ago as follows;
>>
>> caller is "Dave Heimer" <2792275> called extension is 4452271696
>>
>> So hello Dave, I'm not sure exactly what happened but after I started to
>> explain how you reached me instead of what you dialled my phone went 
>> dead!
>>
>> Firstly may I say that you did not disturb me, I was watching TV, but as
>> stated in my directory entries I will answer 24/7 if I am awake and hear
>> the phone ringing, it is not a problem at all.
>>
>> You did not misdial, currently there are just two lines available on my
>> *box, the one that you dialled and 4452271501 which is me in person. The
>> situation is that I am rewriting my dialplan from the ground up as it
>> has become a sprawl of vast proportions with more than 100 extensions so
>> at the moment while the new dialplan is under construction any calls
>> reaching my *box from CNET get stuffed into a [temporary] context where
>> they are switched alternately to 501 or 696 !
>>
>> So if you are calling 696 and get ringing instead just hang up and
>> redial if you don't want to sya 'hello'!
>>
>> Brian
>> _______________________________________________
>> 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: 7.5.516 / Virus Database: 269.21.2/1304 - Release Date: 
>> 2/29/2008
>> 8:18 AM
>>
>>
>
> _______________________________________________
> 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  Sat Mar  1 09:28:19 2008
From: jjones3601 at yahoo.com (john jones)
Date: Sat, 1 Mar 2008 07:28:19 -0800 (PST)
Subject: [VoIP] Cisco 3810 and CAC channel bank available
Message-ID: <996361.86910.qm@web34306.mail.mud.yahoo.com>

I do but other than a few E&M cards, no APM's (FXO/FXS).

John

----- Original Message ----
From: Tammy A. Wisdom <tammy-lists at wiztech.biz>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Tuesday, February 26, 2008 12:49:31 AM
Subject: Re: [VoIP] Cisco 3810 and CAC channel bank available


John,
Do 
you 
have 
any 
3810's 
left 
with 
the 
AVM-6 
cards?
--Tammy



----- 
Original 
Message 
-----
From: 
"john 
jones" 
<jjones3601 at yahoo.com>
To: 
voip at ckts.info
Sent: 
Sunday, 
February 
17, 
2008 
8:51:37 
AM 
GMT 
-07:00 
US/Canada 
Mountain
Subject: 
[VoIP] 
Cisco 
3810 
and 
CAC 
channel 
bank 
available

I 
have 
a 
3810 
available 
attached 
to 
a 
CAC 
24 
port 
FXS 
channel 
bank 
available 
if 
anyone 
is 
interested.  
It 
has 
an 
IOS 
image 
that 
works 
with 
Asterisk.  
It 
does 
NOT 
have 
the 
AVM6 
board 
and 
the 
channel 
bank 
only 
has 
FXS 
ports 
so, 
short 
of 
buying 
an 
AVM 
and 
some 
FXO 
cards, 
only 
has 
a 
single 
type 
of 
access 
available.  
This 
router 
has 
32M 
DRAM 
and 
8 
M 
flash.  
This 
would 
be 
perfect 
for 
a 
collector 
of 
phones 
who 
wants 
to 
1) 
make  
up 
to 
24 
phones 
work 
and 
be 
able 
to 
call 
each 
other  
( 
No 
Asterisk 
required!) 
or 
join 
CNET 
(Asterisk 
required).

The 
equipment 
works 
fine 
but 
due 
to 
age, 
no 
warranty 
is 
offered.  
 
It 
includes 
a 
console 
cable 
for 
the 
router 
and 
a 
T1 
cable 
to 
interconnect 
the 
two 
devices.  
Cost 
is 
$110 
delivered 
in 
the 
US.  
 
I 
also 
have 
some 
3810's 
with 
AVM-6's 
but 
somehow, 
have 
very 
few  
FXS/FXO 
cards  
but 
have 
several 
E&M.

Configuration 
and 
troubleshooting 
assistance 
gladly 
provided! 

Please 
let 
me 
know 
if 
interested.

John 

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

******************************************************************************
Disclaimer:

This 
e-mail 
may 
contain 
trade 
secrets 
or 
privileged, 
undisclosed 
or 
otherwise 
confidential 
information. 
If 
you 
have 
received 
this 
e-mail 
in 
error, 
you 
are 
hereby 
notified 
that 
any 
review, 
copying 
or 
distribution 
of 
it 
is 
strictly 
prohibited. 
Please 
inform 
us 
immediately 
and 
destroy 
the 
original 
transmittal. 
Thank 
you 
for 
your 
cooperation.

******************************************************************************

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




From dheimer at plantationcable.net  Sat Mar  1 16:51:03 2008
From: dheimer at plantationcable.net (Dave H.)
Date: Sat, 1 Mar 2008 17:51:03 -0500
Subject: [VoIP] call to my *box a short while ago.
References: <47C8BECF.7010003@topletter.com><000601c87b64$f353f190$0201a8c0@your94e826b122>
	<6134D5C6A6594EB7B7795CCF5CE2EFAA@IanPC>
Message-ID: <004c01c87bee$ba782cd0$0201a8c0@your94e826b122>

Yo' Ian,
Don't take it personally.  As with many of my screw-ups a bit of alcohol was 
involved. (o.k. maybe more than a bit).

Regardz, Dave H

Dave Heimer
2160 Armor Bridge Road
Greensboro, GA 30642
706-467-3926 - Home
706-318-9577 - Cell
dheimer at plantationcable.net
www.heimer.siteblast.com


----- Original Message ----- 
From: "Ian Jolly" <ian at uax.org.uk>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Saturday, March 01, 2008 3:49 AM
Subject: Re: [VoIP] call to my *box a short while ago.


> "To further confuse thing I though that you were Ian Jolly (no idea why)! 
> "
>
> Oh dear !   What have I done to be confused with Brian  :-))
>
>
> ----- Original Message ----- 
> From: "Dave H." <dheimer at plantationcable.net>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Saturday, March 01, 2008 6:24 AM
> Subject: Re: [VoIP] call to my *box a short while ago.
>
>
>> Yo' Brian,
>>
>> Thanx for the email.  Sorry, I was a bit confused when you answered.  I
>> had
>> a buddy of mine over with his two kids and was showing-off the telephone
>> stuph and was dialing "mind boggleing"  & was kinda at a loss for words
>> whan
>> a human answered.  To further confuse thing I though that you were Ian
>> Jolly
>> (no idea why)  The thing cut off in mid sentence.  I'll give ya a holler
>> tomorrow at a decent hour.
>> Apologies & Regardz, Dave H.
>>
>> Dave Heimer
>> 2160 Armor Bridge Road
>> Greensboro, GA 30642
>> 706-467-3926 - Home
>> 706-318-9577 - Cell
>> dheimer at plantationcable.net
>> www.heimer.siteblast.com
>> ----- Original Message ----- 
>> From: "windmill" <windmill at topletter.com>
>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>> Sent: Friday, February 29, 2008 9:26 PM
>> Subject: [VoIP] call to my *box a short while ago.
>>
>>
>>>
>>>
>>> Hi there,
>>>
>>> I had a call a short while ago as follows;
>>>
>>> caller is "Dave Heimer" <2792275> called extension is 4452271696
>>>
>>> So hello Dave, I'm not sure exactly what happened but after I started to
>>> explain how you reached me instead of what you dialled my phone went
>>> dead!
>>>
>>> Firstly may I say that you did not disturb me, I was watching TV, but as
>>> stated in my directory entries I will answer 24/7 if I am awake and hear
>>> the phone ringing, it is not a problem at all.
>>>
>>> You did not misdial, currently there are just two lines available on my
>>> *box, the one that you dialled and 4452271501 which is me in person. The
>>> situation is that I am rewriting my dialplan from the ground up as it
>>> has become a sprawl of vast proportions with more than 100 extensions so
>>> at the moment while the new dialplan is under construction any calls
>>> reaching my *box from CNET get stuffed into a [temporary] context where
>>> they are switched alternately to 501 or 696 !
>>>
>>> So if you are calling 696 and get ringing instead just hang up and
>>> redial if you don't want to sya 'hello'!
>>>
>>> Brian
>>> _______________________________________________
>>> 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: 7.5.516 / Virus Database: 269.21.2/1304 - Release Date:
>>> 2/29/2008
>>> 8:18 AM
>>>
>>>
>>
>> _______________________________________________
>> 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: 7.5.516 / Virus Database: 269.21.2/1305 - Release Date: 2/29/2008 
> 6:32 PM
>
> 


From ikj1234i at yahoo.com  Sat Mar  1 18:06:52 2008
From: ikj1234i at yahoo.com (ikjtel)
Date: Sat, 1 Mar 2008 16:06:52 -0800 (PST)
Subject: [VoIP] Teenage Hacker Is Blind,
	Brash and in the Crosshairs of the FBI
In-Reply-To: <DD319340F28C421A9E331D528B1EBB27@IanPC>
Message-ID: <802311.67662.qm@web51604.mail.re2.yahoo.com>


One of our very own is quoted in this article...

Hopefully the idiotic yahoo-mail code won't mangle the
URL, which is

http://www.wired.com/politics/law/news/2008/02/blind_hacker?currentPage=all

"...A federal Joint Terrorism Task Force would later
conclude that Gasper had been the victim of a new type
of nasty hoax, called "swatting," that was spreading
across the United States. Pranksters were phoning
police with fake murders and hostage crises, spoofing
their caller IDs so the calls appear to be coming from
inside the target's home. The result: police SWAT
teams rolling to the scene, sometimes bursting into
homes, guns drawn........."


      ____________________________________________________________________________________
Never miss a thing.  Make Yahoo your home page. 
http://www.yahoo.com/r/hs

From doctor.jack.ryan at gmail.com  Mon Mar  3 09:37:51 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Tue, 4 Mar 2008 02:07:51 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
Message-ID: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>

Hi all,

I can't get asterisk to play voice prompts like those in voice-mail.

Fedora 8
Asterisk 1.4.18 downloaded, compiled and installed successfully. Voice-mail
options on.
Sound files located in /var/lib/asterisk/sounds (just the english gsm files)

 If asterisk needs a file I get (for example):

File vm-login does not exist in any format
Unable to open vm-login (format 0x4 (ulaw)): No such file or directory
Couldn't stream login file

asterisk doesn't say where it is looking but the file in question is in
/var/lib/asterisk/sounds.

It can only be a question of "where" or "attributes" (both as per default
bild and install).

What did I do wrong?

Thanks

Jack

From richardlane at exemail.com.au  Mon Mar  3 14:53:34 2008
From: richardlane at exemail.com.au (Richard Lane)
Date: Tue, 04 Mar 2008 07:53:34 +1100
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
Message-ID: <47CC654E.9030904@exemail.com.au>

Hi Jack

Using a bash prompt, goto /var/lib/asterisk/sounds and do a "ls -l ". 
This will list the files. Let us know what is says. It will be something 
like 741 or similar.

Richard

Jack Ryan wrote:
> Hi all,
>
> I can't get asterisk to play voice prompts like those in voice-mail.
>
> Fedora 8
> Asterisk 1.4.18 downloaded, compiled and installed successfully. Voice-mail
> options on.
> Sound files located in /var/lib/asterisk/sounds (just the english gsm files)
>
>  If asterisk needs a file I get (for example):
>
> File vm-login does not exist in any format
> Unable to open vm-login (format 0x4 (ulaw)): No such file or directory
> Couldn't stream login file
>
> asterisk doesn't say where it is looking but the file in question is in
> /var/lib/asterisk/sounds.
>
> It can only be a question of "where" or "attributes" (both as per default
> bild and install).
>
> What did I do wrong?
>
> Thanks
>
> Jack
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   


From doctor.jack.ryan at gmail.com  Mon Mar  3 16:52:40 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Tue, 4 Mar 2008 09:22:40 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <47CC654E.9030904@exemail.com.au>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
Message-ID: <dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>

Hi Richard

The permissions are currently 644 but I have tried other values (that made
sense). The owner and group are "1000" - that is what the asterisk install
set. I have also tried root and asterisk (from the sound file download
instructions).

And yet...silence.

Jack




>
> Using a bash prompt, goto /var/lib/asterisk/sounds and do a "ls -l ".
> This will list the files. Let us know what is says. It will be something
> like 741 or similar.
>
> Richard
>

From spenadel at gmail.com  Mon Mar  3 17:00:23 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Mon, 3 Mar 2008 18:00:23 -0500
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>	<47CC654E.9030904@exemail.com.au>
	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>
Message-ID: <00c301c87d82$5d129ae0$1737d0a0$@com>

Why not set the directory and files to 777?

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Jack Ryan
Sent: Monday, March 03, 2008 5:53 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Asterisk won't play sound (voice prompt) files

Hi Richard

The permissions are currently 644 but I have tried other values (that made
sense). The owner and group are "1000" - that is what the asterisk install
set. I have also tried root and asterisk (from the sound file download
instructions).

And yet...silence.

Jack




>
> Using a bash prompt, goto /var/lib/asterisk/sounds and do a "ls -l ".
> This will list the files. Let us know what is says. It will be something
> like 741 or similar.
>
> Richard
>
_______________________________________________
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  Mon Mar  3 17:30:24 2008
From: voiptandem at shaneyoung.com (Shane Young)
Date: Mon, 03 Mar 2008 17:30:24 -0600
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <47CC654E.9030904@exemail.com.au>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
Message-ID: <20080303173024.8icg4vcdckwccoow@secure.shaneyoung.com>

It looks like it's trying to force you to use ULAW files rather than GSM.

I've never compiled Asterisk without GSM support, but the error  
message doesn't indicate that it's trying to play a GSM file.

Can you install the ulaw files?

Quoting Richard Lane <richardlane at exemail.com.au>:

> Hi Jack
>
> Using a bash prompt, goto /var/lib/asterisk/sounds and do a "ls -l ".
> This will list the files. Let us know what is says. It will be something
> like 741 or similar.
>
> Richard
>
> Jack Ryan wrote:
>> Hi all,
>>
>> I can't get asterisk to play voice prompts like those in voice-mail.
>>
>> Fedora 8
>> Asterisk 1.4.18 downloaded, compiled and installed successfully. Voice-mail
>> options on.
>> Sound files located in /var/lib/asterisk/sounds (just the english gsm files)
>>
>>  If asterisk needs a file I get (for example):
>>
>> File vm-login does not exist in any format
>> Unable to open vm-login (format 0x4 (ulaw)): No such file or directory
>> Couldn't stream login file
>>
>> asterisk doesn't say where it is looking but the file in question is in
>> /var/lib/asterisk/sounds.
>>
>> It can only be a question of "where" or "attributes" (both as per default
>> bild and install).
>>
>> What did I do wrong?
>>
>> Thanks
>>
>> Jack
>> _______________________________________________
>> 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/
>

--Shane
+1-821-7311 CNET

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


From doctor.jack.ryan at gmail.com  Mon Mar  3 17:50:58 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Tue, 4 Mar 2008 10:20:58 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <00c301c87d82$5d129ae0$1737d0a0$@com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>
	<00c301c87d82$5d129ae0$1737d0a0$@com>
Message-ID: <dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>

Lee,

I tried read, write and even execute - silence.

Jack



On 04/03/2008, Lee Spenadel <spenadel at gmail.com> wrote:
>
> Why not set the directory and files to 777?
>
> Lee
>

From doctor.jack.ryan at gmail.com  Mon Mar  3 17:57:38 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Tue, 4 Mar 2008 10:27:38 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <20080303173024.8icg4vcdckwccoow@secure.shaneyoung.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
	<20080303173024.8icg4vcdckwccoow@secure.shaneyoung.com>
Message-ID: <dd7d28450803031557l1676f394sfe066b22ede4e87c@mail.gmail.com>

Shane,

I have had all formats present. The gsm files were put there by the install
program. I removed what I had originally and used the install program
because my manual installation resulted in silence.

I think the error message is confusing because all formats are tried and at
the end, if nothing is found, the error reports the last format tried. The
error was the same when all formats were present.

Jack


On 04/03/2008, Shane Young <voiptandem at shaneyoung.com> wrote:
>
> It looks like it's trying to force you to use ULAW files rather than GSM.
>
> I've never compiled Asterisk without GSM support, but the error
> message doesn't indicate that it's trying to play a GSM file.
>
> Can you install the ulaw files?
>

From mark at rudholm.com  Mon Mar  3 17:31:01 2008
From: mark at rudholm.com (Mark Rudholm)
Date: Mon, 03 Mar 2008 15:31:01 -0800
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>	<47CC654E.9030904@exemail.com.au>	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>	<00c301c87d82$5d129ae0$1737d0a0$@com>
	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>
Message-ID: <47CC8A35.4050708@rudholm.com>

Try downloading a ulaw or g721 file from the internet
and see if it can play that (note that when you specify
the filename in the Playback command, you'll have to
include the filename extension, since that defaults to
gsm).

This way at least you can tell that your Asterisk can
play files back into call channels at all.

Jack Ryan wrote:
> Lee,
> 
> I tried read, write and even execute - silence.
> 
> Jack
> 
> 
> 
> On 04/03/2008, Lee Spenadel <spenadel at gmail.com> wrote:
>> Why not set the directory and files to 777?
>>
>> Lee
>>
> _______________________________________________
> 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  Mon Mar  3 18:04:31 2008
From: voiptandem at shaneyoung.com (Shane Young)
Date: Mon, 03 Mar 2008 18:04:31 -0600
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <dd7d28450803031557l1676f394sfe066b22ede4e87c@mail.gmail.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
	<20080303173024.8icg4vcdckwccoow@secure.shaneyoung.com>
	<dd7d28450803031557l1676f394sfe066b22ede4e87c@mail.gmail.com>
Message-ID: <20080303180431.ir1brkwokcs8s8k0@secure.shaneyoung.com>

Do you have a firewal enabled on your Linux system?

Quoting Jack Ryan <doctor.jack.ryan at gmail.com>:

> Shane,
>
> I have had all formats present. The gsm files were put there by the install
> program. I removed what I had originally and used the install program
> because my manual installation resulted in silence.
>
> I think the error message is confusing because all formats are tried and at
> the end, if nothing is found, the error reports the last format tried. The
> error was the same when all formats were present.
>
> Jack
>
>
> On 04/03/2008, Shane Young <voiptandem at shaneyoung.com> wrote:
>>
>> It looks like it's trying to force you to use ULAW files rather than GSM.
>>
>> I've never compiled Asterisk without GSM support, but the error
>> message doesn't indicate that it's trying to play a GSM file.
>>
>> Can you install the ulaw files?
>>
> _______________________________________________
> 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 doctor.jack.ryan at gmail.com  Mon Mar  3 19:01:25 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Tue, 4 Mar 2008 11:31:25 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <47CC8A35.4050708@rudholm.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>
	<00c301c87d82$5d129ae0$1737d0a0$@com>
	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>
	<47CC8A35.4050708@rudholm.com>
Message-ID: <dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>

I have since installed all sound file formats.

Playback() and SayDigits() result in silence and the same error messages.
The logic of searching for an existing format seems to be common to all
sound output applications. Even so, I tried adding the file extension and
the complete path. The result is the same.

Grrr

Jack

On 04/03/2008, Mark Rudholm <mark at rudholm.com> wrote:
>
> Try downloading a ulaw or g721 file from the internet
> and see if it can play that (note that when you specify
> the filename in the Playback command, you'll have to
> include the filename extension, since that defaults to
> gsm).
>
> This way at least you can tell that your Asterisk can
> play files back into call channels at all.
>

From novackster at gmail.com  Mon Mar  3 21:21:25 2008
From: novackster at gmail.com (John Novack)
Date: Mon, 03 Mar 2008 22:21:25 -0500
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>	<47CC654E.9030904@exemail.com.au>	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>	<00c301c87d82$5d129ae0$1737d0a0$@com>	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>	<47CC8A35.4050708@rudholm.com>
	<dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>
Message-ID: <47CCC035.5010802@stromberg-carlson.org>

What is the version and release of Asterisk?

Not sure about this, but MANY have had problems with Fedora Core X and 
it generally is troublesome in a number of ways.
Search the Asterisk users list for details
I can't be sure it relates to your present problem.

CentOs is the same price and trouble free.

John Novack


Jack Ryan wrote:
> I have since installed all sound file formats.
>
> Playback() and SayDigits() result in silence and the same error messages.
> The logic of searching for an existing format seems to be common to all
> sound output applications. Even so, I tried adding the file extension and
> the complete path. The result is the same.
>
> Grrr
>
> Jack
>
> On 04/03/2008, Mark Rudholm <mark at rudholm.com> wrote:
>   
>> Try downloading a ulaw or g721 file from the internet
>> and see if it can play that (note that when you specify
>> the filename in the Playback command, you'll have to
>> include the filename extension, since that defaults to
>> gsm).
>>
>> This way at least you can tell that your Asterisk can
>> play files back into call channels at all.
>>
>>     
> _______________________________________________
> 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 doctor.jack.ryan at gmail.com  Mon Mar  3 21:44:13 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Tue, 4 Mar 2008 14:14:13 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <47CCC035.5010802@stromberg-carlson.org>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>
	<00c301c87d82$5d129ae0$1737d0a0$@com>
	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>
	<47CC8A35.4050708@rudholm.com>
	<dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>
	<47CCC035.5010802@stromberg-carlson.org>
Message-ID: <dd7d28450803031944o2aa27951r7b0c5fb32a219ea3@mail.gmail.com>

Hi John,

Asterisk is version 1.4.18. I downloaded, built and installed yesterday as a
way of "starting again". Asterisk started again but it did it silently
again.

The error is: "the file is not there."

So either asterisk is looking somewhere else (not in
/var/lib/asterisk/sounds) or the "find a compatible file type" logic doesn't
work. Either way I would have thought someone else would have the same
problem.

I don't like being special.

Jack


On 04/03/2008, John Novack <novackster at gmail.com> wrote:
>
> What is the version and release of Asterisk?
>
> Not sure about this, but MANY have had problems with Fedora Core X and
> it generally is troublesome in a number of ways.
> Search the Asterisk users list for details
> I can't be sure it relates to your present problem.
>
> CentOs is the same price and trouble free.
>
> John Novack
>
>
> Jack Ryan wrote:
> > I have since installed all sound file formats.
> >
> > Playback() and SayDigits() result in silence and the same error
> messages.
> > The logic of searching for an existing format seems to be common to all
> > sound output applications. Even so, I tried adding the file extension
> and
> > the complete path. The result is the same.
> >
> > Grrr
> >
> > Jack
> >
> > On 04/03/2008, Mark Rudholm <mark at rudholm.com> wrote:
> >
> >> Try downloading a ulaw or g721 file from the internet
> >> and see if it can play that (note that when you specify
> >> the filename in the Playback command, you'll have to
> >> include the filename extension, since that defaults to
> >> gsm).
> >>
> >> This way at least you can tell that your Asterisk can
> >> play files back into call channels at all.
> >>
> >>
> > _______________________________________________
> > 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 novackster at gmail.com  Tue Mar  4 12:24:37 2008
From: novackster at gmail.com (John Novack)
Date: Tue, 04 Mar 2008 13:24:37 -0500
Subject: [VoIP] Thin Clients
Message-ID: <47CD93E5.1080009@stromberg-carlson.org>

There are several HP 5710 Thin Clients on eBay, in case anyone is interested

These are working well for several CNET members, by replacing the 
firmware with AstLinux

Search either for HP t5700 Thin Client or seller is reclamere02

John Novack

-- 
Dog is my co-pilot


From t2600 at sbcglobal.net  Tue Mar  4 12:53:58 2008
From: t2600 at sbcglobal.net (Jerry Petrizze)
Date: Tue, 4 Mar 2008 10:53:58 -0800
Subject: [VoIP] Thin Clients
References: <47CD93E5.1080009@stromberg-carlson.org>
Message-ID: <00f401c87e29$1ae47d80$0501a8c0@pentium4>

John what sort of system size will these handle?
Are they just good for an interface like for 1 each FXS, FXO??
Jerry Petrizze

----- Original Message ----- 
From: "John Novack" <novackster at gmail.com>
To: "Voice Over IP" <voip at ckts.info>
Sent: Tuesday, March 04, 2008 10:24 AM
Subject: [VoIP] Thin Clients


> There are several HP 5710 Thin Clients on eBay, in case anyone is
interested
>
> These are working well for several CNET members, by replacing the
> firmware with AstLinux
>
> Search either for HP t5700 Thin Client or seller is reclamere02
>
> 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/


From novackster at gmail.com  Tue Mar  4 14:39:45 2008
From: novackster at gmail.com (John Novack)
Date: Tue, 04 Mar 2008 15:39:45 -0500
Subject: [VoIP] Thin Clients
In-Reply-To: <00f401c87e29$1ae47d80$0501a8c0@pentium4>
References: <47CD93E5.1080009@stromberg-carlson.org>
	<00f401c87e29$1ae47d80$0501a8c0@pentium4>
Message-ID: <47CDB391.80801@stromberg-carlson.org>

These are primarily for SIP and IAX, as the PCI slot is generally 
inaccessible to use a half size PCI card. There seems to be no expansion 
kits available that would allow the use of a TDM400.
For external devices, however, after loading AstLinux they are quite usable.

John  Novack


Jerry Petrizze wrote:
> John what sort of system size will these handle?
> Are they just good for an interface like for 1 each FXS, FXO??
> Jerry Petrizze
>
> ----- Original Message ----- 
> From: "John Novack" <novackster at gmail.com>
> To: "Voice Over IP" <voip at ckts.info>
> Sent: Tuesday, March 04, 2008 10:24 AM
> Subject: [VoIP] Thin Clients
>
>
>   
>> There are several HP 5710 Thin Clients on eBay, in case anyone is
>>     
> interested
>   
>> These are working well for several CNET members, by replacing the
>> firmware with AstLinux
>>
>> Search either for HP t5700 Thin Client or seller is reclamere02
>>
>> 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/
>>     
>
>
>   

-- 
Dog is my co-pilot


From doctor.jack.ryan at gmail.com  Wed Mar  5 04:13:13 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Wed, 5 Mar 2008 20:43:13 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <dd7d28450803031944o2aa27951r7b0c5fb32a219ea3@mail.gmail.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>
	<00c301c87d82$5d129ae0$1737d0a0$@com>
	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>
	<47CC8A35.4050708@rudholm.com>
	<dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>
	<47CCC035.5010802@stromberg-carlson.org>
	<dd7d28450803031944o2aa27951r7b0c5fb32a219ea3@mail.gmail.com>
Message-ID: <dd7d28450803050213o77cb2837m154c29e6433477f@mail.gmail.com>

Well, I rolled up my sleeves and delved into the source code in an attempt
to solve the problem. As I am new to asterisk and to linux I don't know how
previous versions worked but it seems that someone has decided to change the
location the sound files are read from without telling those who setup the
default configuration files.

In asterisk.conf I changed

astdatadir => /usr/share/asterisk

to

astdatadir => /var/lib/asterisk

and behold, there was sound.

I'm sure there will be side effects but what would life be without
surprises?

Jack


On 04/03/2008, Jack Ryan <doctor.jack.ryan at gmail.com> wrote:
>
> Hi John,
>
> Asterisk is version 1.4.18. I downloaded, built and installed yesterday as
> a way of "starting again". Asterisk started again but it did it silently
> again.
>
> The error is: "the file is not there."
>
> So either asterisk is looking somewhere else (not in
> /var/lib/asterisk/sounds) or the "find a compatible file type" logic doesn't
> work. Either way I would have thought someone else would have the same
> problem.
>
> I don't like being special.
>
> Jack
>
>

From spenadel at gmail.com  Wed Mar  5 06:40:28 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Wed, 5 Mar 2008 07:40:28 -0500
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <dd7d28450803050213o77cb2837m154c29e6433477f@mail.gmail.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>	<47CC654E.9030904@exemail.com.au>	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>	<00c301c87d82$5d129ae0$1737d0a0$@com>	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>	<47CC8A35.4050708@rudholm.com>	<dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>	<47CCC035.5010802@stromberg-carlson.org>	<dd7d28450803031944o2aa27951r7b0c5fb32a219ea3@mail.gmail.com>
	<dd7d28450803050213o77cb2837m154c29e6433477f@mail.gmail.com>
Message-ID: <001d01c87ebe$180dc220$48294660$@com>

Nice work Jack, but with Open Source code, not surprising.

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Jack Ryan
Sent: Wednesday, March 05, 2008 5:13 AM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Asterisk won't play sound (voice prompt) files

Well, I rolled up my sleeves and delved into the source code in an attempt
to solve the problem. As I am new to asterisk and to linux I don't know how
previous versions worked but it seems that someone has decided to change the
location the sound files are read from without telling those who setup the
default configuration files.

In asterisk.conf I changed

astdatadir => /usr/share/asterisk

to

astdatadir => /var/lib/asterisk

and behold, there was sound.

I'm sure there will be side effects but what would life be without
surprises?

Jack


On 04/03/2008, Jack Ryan <doctor.jack.ryan at gmail.com> wrote:
>
> Hi John,
>
> Asterisk is version 1.4.18. I downloaded, built and installed yesterday as
> a way of "starting again". Asterisk started again but it did it silently
> again.
>
> The error is: "the file is not there."
>
> So either asterisk is looking somewhere else (not in
> /var/lib/asterisk/sounds) or the "find a compatible file type" logic
doesn't
> work. Either way I would have thought someone else would have the same
> problem.
>
> I don't like being special.
>
> Jack
>
>
_______________________________________________
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 Mar  5 07:20:33 2008
From: voiptandem at shaneyoung.com (Shane Young)
Date: Wed, 05 Mar 2008 07:20:33 -0600
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <001d01c87ebe$180dc220$48294660$@com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<47CC654E.9030904@exemail.com.au>
	<dd7d28450803031452vaadd8e6if7663e5030a8be97@mail.gmail.com>
	<00c301c87d82$5d129ae0$1737d0a0$@com>
	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>
	<47CC8A35.4050708@rudholm.com>
	<dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>
	<47CCC035.5010802@stromberg-carlson.org>
	<dd7d28450803031944o2aa27951r7b0c5fb32a219ea3@mail.gmail.com>
	<dd7d28450803050213o77cb2837m154c29e6433477f@mail.gmail.com>
	<001d01c87ebe$180dc220$48294660$@com>
Message-ID: <20080305072033.zqh6ypofk0ww8o0w@secure.shaneyoung.com>

Jack

It seems like /usr/share/asterisk is used on some systems which have  
their own asterisk package such as Debian, Ubuntu, etc.

Did you use a package for your specifc distribution, or did you just  
download the source and compile it?

--Shane


Quoting Lee Spenadel <spenadel at gmail.com>:

> Nice work Jack, but with Open Source code, not surprising.
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Jack Ryan
> Sent: Wednesday, March 05, 2008 5:13 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Asterisk won't play sound (voice prompt) files
>
> Well, I rolled up my sleeves and delved into the source code in an attempt
> to solve the problem. As I am new to asterisk and to linux I don't know how
> previous versions worked but it seems that someone has decided to change the
> location the sound files are read from without telling those who setup the
> default configuration files.
>
> In asterisk.conf I changed
>
> astdatadir => /usr/share/asterisk
>
> to
>
> astdatadir => /var/lib/asterisk
>
> and behold, there was sound.
>
> I'm sure there will be side effects but what would life be without
> surprises?
>
> Jack
>
>
> On 04/03/2008, Jack Ryan <doctor.jack.ryan at gmail.com> wrote:
>>
>> Hi John,
>>
>> Asterisk is version 1.4.18. I downloaded, built and installed yesterday as
>> a way of "starting again". Asterisk started again but it did it silently
>> again.
>>
>> The error is: "the file is not there."
>>
>> So either asterisk is looking somewhere else (not in
>> /var/lib/asterisk/sounds) or the "find a compatible file type" logic
> doesn't
>> work. Either way I would have thought someone else would have the same
>> problem.
>>
>> I don't like being special.
>>
>> Jack
>>
>>
> _______________________________________________
> 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/
>

--Shane
+1-821-7311 CNET

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


From novackster at gmail.com  Wed Mar  5 15:21:21 2008
From: novackster at gmail.com (John Novack)
Date: Wed, 05 Mar 2008 16:21:21 -0500
Subject: [VoIP] Mason Show - Keith Hlavacs
Message-ID: <47CF0ED1.30506@stromberg-carlson.org>

 Keith Hlavacs invites anyone interested in switching to visit him after 
the TCI Spring show in Mason.
He expects his switch move to be complete enough by April to have visitors.
Keith will also be at the show, and will provide directions at that time.
Keith is not yet on CNET, but hopes to be in the future.

John Novack

-- 
Dog is my co-pilot


From doctor.jack.ryan at gmail.com  Wed Mar  5 16:47:33 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Thu, 6 Mar 2008 09:17:33 +1030
Subject: [VoIP] Asterisk won't play sound (voice prompt) files
In-Reply-To: <20080305072033.zqh6ypofk0ww8o0w@secure.shaneyoung.com>
References: <dd7d28450803030737p49ab1adcu11bf693817c908a6@mail.gmail.com>
	<00c301c87d82$5d129ae0$1737d0a0$@com>
	<dd7d28450803031550n15b93970r66ac835b8b2a1c72@mail.gmail.com>
	<47CC8A35.4050708@rudholm.com>
	<dd7d28450803031701n15e8362ek6a4423f7acac1ee1@mail.gmail.com>
	<47CCC035.5010802@stromberg-carlson.org>
	<dd7d28450803031944o2aa27951r7b0c5fb32a219ea3@mail.gmail.com>
	<dd7d28450803050213o77cb2837m154c29e6433477f@mail.gmail.com>
	<001d01c87ebe$180dc220$48294660$@com>
	<20080305072033.zqh6ypofk0ww8o0w@secure.shaneyoung.com>
Message-ID: <dd7d28450803051447i31e13a46u9019923235a53d2a@mail.gmail.com>

Hi Shane,

I used the package for Fedora but it wouldn't play sounds. That was why I
removed it and installed the compiled version. Still, the error was made by
the Fedora people. Their configuration files were modified to use

astdatadir => /usr/share/asterisk

but their binaries use

astdatadir => /var/lib/asterisk

After I removed Fedora's binaries I was still using their configuration
files hence the compiled asterisk still would not play sounds.

Why do they bother customising it?

Jack



On 05/03/2008, Shane Young <voiptandem at shaneyoung.com> wrote:
>
> Jack
>
> It seems like /usr/share/asterisk is used on some systems which have
> their own asterisk package such as Debian, Ubuntu, etc.
>
> Did you use a package for your specifc distribution, or did you just
> download the source and compile it?
>
> --Shane
>
>
> Quoting Lee Spenadel <spenadel at gmail.com>:
>
> > Nice work Jack, but with Open Source code, not surprising.
> >
> > -----Original Message-----
> > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
> Of
> > Jack Ryan
> > Sent: Wednesday, March 05, 2008 5:13 AM
> > To: Voice Over IP Tandem for Analog Switches
> > Subject: Re: [VoIP] Asterisk won't play sound (voice prompt) files
> >
> > Well, I rolled up my sleeves and delved into the source code in an
> attempt
> > to solve the problem. As I am new to asterisk and to linux I don't know
> how
> > previous versions worked but it seems that someone has decided to change
> the
> > location the sound files are read from without telling those who setup
> the
> > default configuration files.
> >
> > In asterisk.conf I changed
> >
> > astdatadir => /usr/share/asterisk
> >
> > to
> >
> > astdatadir => /var/lib/asterisk
> >
> > and behold, there was sound.
> >
> > I'm sure there will be side effects but what would life be without
> > surprises?
> >
> > Jack
> >
> >
> > On 04/03/2008, Jack Ryan <doctor.jack.ryan at gmail.com> wrote:
> >>
> >> Hi John,
> >>
> >> Asterisk is version 1.4.18. I downloaded, built and installed yesterday
> as
> >> a way of "starting again". Asterisk started again but it did it
> silently
> >> again.
> >>
> >> The error is: "the file is not there."
> >>
> >> So either asterisk is looking somewhere else (not in
> >> /var/lib/asterisk/sounds) or the "find a compatible file type" logic
> > doesn't
> >> work. Either way I would have thought someone else would have the same
> >> problem.
> >>
> >> I don't like being special.
> >>
> >> Jack
> >>
> >>
> > _______________________________________________
> > 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/
> >
>
> --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 novackster at gmail.com  Thu Mar  6 15:22:16 2008
From: novackster at gmail.com (John Novack)
Date: Thu, 06 Mar 2008 16:22:16 -0500
Subject: [VoIP] More 5710's up on eBay
Message-ID: <47D06088.5040501@stromberg-carlson.org>

More thin client HP 5710's up on eBay - include power supply
Picture is misleading, shows 5 but auction is for one

130203813544

For anyone who wants to join CNET in a SIP/IAX only setup, these are a 
nice unit. At least three current members have them now. A SIP phone 
and/or a Cisco 3810 and a little work and you have a low power 
consumption and no fan. The Cisco unit has a small fan.

Not my auction.

John Novack

-- 
Dog is my co-pilot


From doctor.jack.ryan at gmail.com  Thu Mar  6 21:57:13 2008
From: doctor.jack.ryan at gmail.com (Jack Ryan)
Date: Fri, 7 Mar 2008 14:27:13 +1030
Subject: [VoIP] Linksys 3102/Linksys3000
In-Reply-To: <0873E4A31E5B4391BBD58DBD6A5CC1C4@IanPC>
References: <0873E4A31E5B4391BBD58DBD6A5CC1C4@IanPC>
Message-ID: <dd7d28450803061957u43b4c397k9a832478dcd0480b@mail.gmail.com>

Hi Ian,

Did you sort this out? I anwered it in a recent e-mail to you without
realising you had posted this question.

Its in the "Missing sound files" thread

Regards

Jack



On 23/01/2008, Ian Jolly <ian at uax.org.uk> wrote:
>
> I've been trying without luck to set up one of these devices (3102) to
> connect to my Asterisk.  I've got this far -
>
> FXS port working without any problem .  I can dial in/out to/from a
> telephone connected to the FXS port.
>
> FXO port - Grrrrrrrrrh !
>
> Currently I'm only interested in getting it such that the FXO accepts
> incoming calls from a line on my old Strowger exchange which puts out AC
> ringing.  When I call into the FXO from the line, the FXO answers and gives
> dialling tone but nothing beyond that. The call then drops out after a
> timeout.
>
> In my sip.conf, I've got the following -
>
> [3102FXO]
> type=friend ;
> host=dynamic
> context=3102FXO
> username=3102FXO
> secret=700484
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
>
> At the CLI> it appears to be registering -
>
> asterisk*CLI> sip show peers
> Name/username                  Host            Dyn Nat ACL
> Port        Status
> 3102FXO/3102FXO            192.168.1.16     D   N          5061     OK (10
> ms)
>
> That seems to be OK so far.
>
> The problem appears to be in the ATA.
>
> I've tried that various setting for the FXO and its dialplan in the '3102
> FXO setup' suggested on various forums/wiki's when working to an Asterisk -
> but to no avail :-(
>
> Basically I want it to route the incoming call to a single number on my
> Asterisk.
>
> Anyone any ideas?
>
> Has anyone any setting for the Linksys 3000/3102 ATA when it is connected
> as per the 'ATA' model for connecting a 'heritage' switch as per
> http://www.ckts.info/fx-model.pdf  ??
>
> Ian Jolly
>
> Ian Jolly
> _______________________________________________
> 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  Sat Mar  8 09:14:41 2008
From: dfroula at sbcglobal.net (Donald Froula)
Date: Sat, 8 Mar 2008 07:14:41 -0800 (PST)
Subject: [VoIP] 1-762 Down for Maintenance
Message-ID: <78943.48422.qm@web83205.mail.mud.yahoo.com>

1-762 (projectmf.homelinux.com) will be down for
maintenance this morning while I use the system to
clone several backups of my working hard drive.

I've come by a few more of the Wyse thin client boxes
that I have been using for my Asterisk/ProjectMF
system for the last 6 months.

These are Wyse 941Gs. They look more like a mini PC.
They are convection-cooled, run at 1 GHz, and have 512
Mbyte of RAM installed. They run Windows XPe
(XP-embedded) from a small FLASH memory module plugged
directly into a standard IDE controller slot. They
also have a usable PCI slot that I use for a dual
Ethernet card for the ProjectMF TDMoE trunks.

I've been pulling the module, connecting a 6 Gbyte IDE
drive, and changing the BIOS to boot from an external
USB CD drive and then the hard drive. There is no CD
ROM installed. I am using a USB CDROM that connects to
2 of the available 4 USB ports (the extra is for
power).

The standard BIOS works fine for detecting these
devices. The BIOS is password protected ("Fireport").
If the backup battery coin cell is removed, the BIOS
reverts to factory defaults of booting from the FLASH
module, and password locks again. I keep the password
written inside the case to refresh my failing memory!

I'm cloning my operating disk with a free LINUX
utility called PING (Partimage is not Ghost). I booted
off the CD, made with the .iso image on the web site.
It attached an external USB drive I used for backup
and allowed me to backup all partitions, even the BIOS
settings. I backed up to a USB drive formatted as
FAT32. PING even broke the backup files into smaller
chunks to accommodate the 4 Gbyte file size limit of
FAT32.

It is also possible to make a series of DVDs with the
backup images and restore a system from those. This
might be the ticket for getting CNET members up and
running quickly.

D.

From stfkerman at jps.net  Sat Mar  8 12:11:32 2008
From: stfkerman at jps.net (Steph Kerman)
Date: Sat, 08 Mar 2008 13:11:32 -0500
Subject: [VoIP] More of the usual Verizon lameness
Message-ID: <47D2D6D4.6070808@jps.net>

Does anyone have a user guide for the Westell B90-327W15-06 ADSL modem & 
wireless gateway?  This is a Verizon version.  The Westell website says:
> This product was specifically designed for Verizon. You will need to 
> contact Verizon for any support issues. Their support number is 
> 800-567-6789.  
As one would expect, even when told that the Westell website instructs 
you to contact Verizon, Verizon insists that Westell must be contacted 
for the manual, that Vz does not provide manuals or PDFs, but that they 
will support the modem live in real time.   Frankly, given the quality 
of their tech support help, it would probably be a lot more productive 
to just "do it myself" with the manual.

Of the nine 327W series modems listed on the Westell site, 3 of the 9 
have the above cop-out.  A 4th produces a "not found" error when manual 
downloading is attempted.  The other 3 produce this:        
> Westell technical support is offered to Internet Service Providers, 
> independent phone companies, and distributors.  If you are an end 
> user, please contact your Internet Service Provider for technical 
> support.    
or this:         
> This product was specifically designed for Bell South.  You will need 
> to contact Bell South for any support issues.  Their support number is 
> 888-321-2375 (888-321-ADSL).  
I downloaded the 2 available manuals, (A90-327W10-06 and 
D90-327W11-06).  At least superficially, they differ from each other 
only with respect to the presence or absence of the USB port.  How they 
differ from the B90-327W15-06  is anyone's guess. 

Given that the two available manuals are 187 and 157 pages respectively, 
I suspect that they contain many more pearls of wisdom than can be 
conveyed in a lifetime verbally by a Vz "tech support" person and would 
like to get the correct manual for this modem.   Has anyone succeeded in 
getting the correct PDF for the B90-327W15-06?

Thanks
Steph

               
            
               
               


From richardlane at exemail.com.au  Sat Mar  8 15:32:54 2008
From: richardlane at exemail.com.au (Richard Lane)
Date: Sun, 09 Mar 2008 08:32:54 +1100
Subject: [VoIP] Asterisk & Bluetooth
Message-ID: <47D30606.4080701@exemail.com.au>

Hi All,

I have a technical question for all you asterisk boffins out there.

Has anyone setup bluetooth connectivity to their asterisk units to allow 
a mobile phone to act as a FXO port?

I have a bluetooth USB dongle which I thought I would like to configure 
in this way. The dongle is a HP BT450 USB unit.

Richard

From dfroula at sbcglobal.net  Sat Mar  8 17:21:35 2008
From: dfroula at sbcglobal.net (Donald Froula)
Date: Sat, 8 Mar 2008 15:21:35 -0800 (PST)
Subject: [VoIP] 1-762 Down for Maintenance
In-Reply-To: <78943.48422.qm@web83205.mail.mud.yahoo.com>
Message-ID: <351428.22242.qm@web83204.mail.mud.yahoo.com>

Back on-line.

I cloned four copies of my drive. The freeware PING
worked perfectly for cloning, except for one thing.
The swap partition on the copied drive was
successfully created, but for some reason without a
valid signature. When booting on the copied drive, a
swapon error occurred. I had to issue the "mkswap
/dev/hda5" command to rebuild the signature. On a
reboot, all worked normally.


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

> 1-762 (projectmf.homelinux.com) will be down for
> maintenance this morning while I use the system to
> clone several backups of my working hard drive.
> 
> I've come by a few more of the Wyse thin client
> boxes
> that I have been using for my Asterisk/ProjectMF
> system for the last 6 months.
> 
> These are Wyse 941Gs. They look more like a mini PC.
> They are convection-cooled, run at 1 GHz, and have
> 512
> Mbyte of RAM installed. They run Windows XPe
> (XP-embedded) from a small FLASH memory module
> plugged
> directly into a standard IDE controller slot. They
> also have a usable PCI slot that I use for a dual
> Ethernet card for the ProjectMF TDMoE trunks.
> 
> I've been pulling the module, connecting a 6 Gbyte
> IDE
> drive, and changing the BIOS to boot from an
> external
> USB CD drive and then the hard drive. There is no CD
> ROM installed. I am using a USB CDROM that connects
> to
> 2 of the available 4 USB ports (the extra is for
> power).
> 
> The standard BIOS works fine for detecting these
> devices. The BIOS is password protected
> ("Fireport").
> If the backup battery coin cell is removed, the BIOS
> reverts to factory defaults of booting from the
> FLASH
> module, and password locks again. I keep the
> password
> written inside the case to refresh my failing
> memory!
> 
> I'm cloning my operating disk with a free LINUX
> utility called PING (Partimage is not Ghost). I
> booted
> off the CD, made with the .iso image on the web
> site.
> It attached an external USB drive I used for backup
> and allowed me to backup all partitions, even the
> BIOS
> settings. I backed up to a USB drive formatted as
> FAT32. PING even broke the backup files into smaller
> chunks to accommodate the 4 Gbyte file size limit of
> FAT32.
> 
> It is also possible to make a series of DVDs with
> the
> backup images and restore a system from those. This
> might be the ticket for getting CNET members up and
> running quickly.
> 
> D.
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From mitya89 at aim.com  Sat Mar  8 19:13:13 2008
From: mitya89 at aim.com (Dimitri Ressetar)
Date: Sat, 08 Mar 2008 20:13:13 -0500
Subject: [VoIP] More of the usual Verizon lameness
Message-ID: <47D339A9.1080407@aim.com>

I have PDF copies of the Westell 2110, 2200, and 6100 DSL Modem/Router
User Guides. These modems aren't the ones with the wireless built into
them, but they do act as a Ethernet router when connected to a switch,
so this might not be much help to you.

I found these in the Modems/Westell/Install directory of the CD Verizon
sent me along with the modem and other DSL equipment. If you haven't
checked, poke around and see what you can find on your CD.
-Dimitri Ressetar

From stfkerman at jps.net  Sat Mar  8 19:22:41 2008
From: stfkerman at jps.net (Steph Kerman)
Date: Sat, 08 Mar 2008 20:22:41 -0500
Subject: [VoIP] More of the usual Verizon lameness
In-Reply-To: <47D339A9.1080407@aim.com>
References: <47D339A9.1080407@aim.com>
Message-ID: <47D33BE1.2080904@jps.net>

Thanks.  I did a search for PDFs on the CDROM before posting to the 
list.  I found no PDFs at all.  I did find 258 .HTM files.

Even uglier yet, the CDROM produces a Windows error message when 
inserted in the drive because it is frustrated in its attempt to find a 
flash player on my system so that it can dazzle me with what is probably 
nothing more than useless B.S.

Steph

Dimitri Ressetar wrote:
> I have PDF copies of the Westell 2110, 2200, and 6100 DSL Modem/Router
> User Guides. These modems aren't the ones with the wireless built into
> them, but they do act as a Ethernet router when connected to a switch,
> so this might not be much help to you.
>
> I found these in the Modems/Westell/Install directory of the CD Verizon
> sent me along with the modem and other DSL equipment. If you haven't
> checked, poke around and see what you can find on your CD.
> -Dimitri Ressetar
>
>
>   

From mitya89 at aim.com  Sat Mar  8 19:25:42 2008
From: mitya89 at aim.com (Dimitri Ressetar)
Date: Sat, 08 Mar 2008 20:25:42 -0500
Subject: [VoIP] Upgrading my Server & Asterisk 1.4 Troubles
Message-ID: <47D33C96.9090501@aim.com>

Hi everyone!

I've had my switch offline for quite a while, and I'm doing a lot of
upgrades to it. I got a T1 interface card and a 24 FXS channel bank, so
I can actually plug in real phones to my Asterisk system! I've also
completely upgraded the server, it is now a Celeron 2.5 GHz with 768MB
of RAM (much better than my P2 300 MHz).

Right now, I'm running into the problem many of you have already run
into upgrading to Asterisk 1.4 -- ENUM support. I know that the 1.2
version won't work, but do we have a relatively standardized version of
cnet ENUM dialing for 1.4? I've been able to update the rest of my
dialplan to 1.4 without much trouble, but I don't know how ENUM and ENUM
dialing in Asterisk work. I'm not even sure if I used the original
example from the ckts.info website, I might have copied the one from an
old version of TrixBox, back before I switched to coding my own dialplan.

Maybe there should be a page somewhere on the ckts.info website with all
the info that each Cnet Office needs to know...the basic ENUM, IAX, SIP,
and CallerID configurations to work on the network, as well as the usual
test line numbering, etc. Just a thought.

Thanks in advance to anyone who can help!
-Dimitri Ressetar

From spenadel at gmail.com  Sat Mar  8 19:29:40 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Sat, 8 Mar 2008 20:29:40 -0500
Subject: [VoIP] Upgrading my Server & Asterisk 1.4 Troubles
In-Reply-To: <47D33C96.9090501@aim.com>
References: <47D33C96.9090501@aim.com>
Message-ID: <00de01c88185$0bf68d40$23e3a7c0$@com>

Here's what I implemented when I upgraded to 1.4

Lee


;*************************************************************************
[macro-dialcnet]
;
; Collectors' Network
;
exten => s,1,Set(E164NETWORKS=std.ckts.info)
;
; Check to see if ARG1 is preceded by a "+"
exten => s,2,GotoIf($[ ${ARG1:0:1} = "+"]?startloop) ;  ; Skip next line if
it already is prefixed by a plus; 
							; Otherwise, add one
to the beginning.
exten => s,3,Set(ARG1=+${ARG1})
exten =>
s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,1,std.ckts.info)})
;exten =>
s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1}.std.ckts.info,ALL,,std.ckts.inf
o)})
;
; Sanity check the return, make sure there's something in there.
; If not, send it to CONTINUE.
exten => s,5,GotoIf($[${LEN(${ENUM})} = 0 ]?continue) ; ; If the return
includes the word SIP, go to SIPURI 
exten => s,6,GotoIf($[${ENUM:0:3} = sip ]?sipuri) ; ; Otherwise, if the
return includes the word IAX, go to IAXURI 
exten => s,7,GotoIf($[${ENUM:0:3} = iax ]?iaxuri) ; ; And, if the return
includes the word H323, send it to H323URI 
exten => s,8,GotoIf($[${ENUM:0:3} = h32 ]?h323uri) ; ; If we're here, it's
not a protocol we know about. Let's increment the pointer 
							; and if it's more
than ENUMCOUNT, we know we've run out of options.
;
; This is the generic CANT BE ROUTED spot 
exten => s,9(continue),Macro(invalid-office-code,${ARG1}) 
exten => s,10,Wait(5) 
exten => s,11,Hangup ; ; If the prefix is 'sip:'...
exten => s,12(sipuri),Set(DIALSTR=SIP/${ENUM:5})
exten => s,13,Goto(dodial)
;
; If it's IAX2...
exten => s,14(iaxuri),Set(DIALSTR=IAX2/${ENUM:5})
exten => s,15,Goto(dodial)
;
; Or even if it's H323.
exten => s,16(h323uri),Set(DIALSTR=H323/${ENUM:5})
;
; And this is where we end up if we actually CAN route the call.
exten => s,17(dodial),Dial(${DIALSTR})
exten => s,18,Hangup
exten => s,118,Busy
;*************************************************************************

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Dimitri Ressetar
Sent: Saturday, March 08, 2008 8:26 PM
To: voip at ckts.info
Subject: [VoIP] Upgrading my Server & Asterisk 1.4 Troubles

Hi everyone!

I've had my switch offline for quite a while, and I'm doing a lot of
upgrades to it. I got a T1 interface card and a 24 FXS channel bank, so
I can actually plug in real phones to my Asterisk system! I've also
completely upgraded the server, it is now a Celeron 2.5 GHz with 768MB
of RAM (much better than my P2 300 MHz).

Right now, I'm running into the problem many of you have already run
into upgrading to Asterisk 1.4 -- ENUM support. I know that the 1.2
version won't work, but do we have a relatively standardized version of
cnet ENUM dialing for 1.4? I've been able to update the rest of my
dialplan to 1.4 without much trouble, but I don't know how ENUM and ENUM
dialing in Asterisk work. I'm not even sure if I used the original
example from the ckts.info website, I might have copied the one from an
old version of TrixBox, back before I switched to coding my own dialplan.

Maybe there should be a page somewhere on the ckts.info website with all
the info that each Cnet Office needs to know...the basic ENUM, IAX, SIP,
and CallerID configurations to work on the network, as well as the usual
test line numbering, etc. Just a thought.

Thanks in advance to anyone who can help!
-Dimitri Ressetar
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/


From mitya89 at aim.com  Sat Mar  8 20:43:08 2008
From: mitya89 at aim.com (Dimitri Ressetar)
Date: Sat, 08 Mar 2008 21:43:08 -0500
Subject: [VoIP]  Upgrading my Server & Asterisk 1.4 Troubles
References: 47D33C96.9090501@aim.com
Message-ID: <47D34EBC.5010401@aim.com>

wow, you're fast! thanks so much, i'll try it out!

From novackster at gmail.com  Sat Mar  8 22:31:43 2008
From: novackster at gmail.com (John Novack)
Date: Sat, 08 Mar 2008 23:31:43 -0500
Subject: [VoIP] Asterisk & Bluetooth
In-Reply-To: <47D30606.4080701@exemail.com.au>
References: <47D30606.4080701@exemail.com.au>
Message-ID: <47D3682F.5070404@stromberg-carlson.org>

Some time back I remember discussions on the Asterisk users list 
regarding bluetooth. Not much good either. The Bluetooth code seemed to 
have problems, and the issue hasn't been raised recently
You may want to do a Google search through the mailing list and see 
whats what.

John Novack


Richard Lane wrote:
> Hi All,
>
> I have a technical question for all you asterisk boffins out there.
>
> Has anyone setup bluetooth connectivity to their asterisk units to allow 
> a mobile phone to act as a FXO port?
>
> I have a bluetooth USB dongle which I thought I would like to configure 
> in this way. The dongle is a HP BT450 USB unit.
>
> Richard
> _______________________________________________
> 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 chad at maine.edu  Sun Mar  9 12:07:01 2008
From: chad at maine.edu (Chad Perkins)
Date: Sun, 09 Mar 2008 13:07:01 -0400
Subject: [VoIP] More of the usual Verizon lameness
In-Reply-To: <47D2D6D4.6070808@jps.net>
Message-ID: <47D3E0F5.6395.12D6836@localhost>

I have a VZ supplied A90-327W15-06 (among others) and am 
familiar with their products.  Did you have question about something 
the product, or just lamenting the lack of provided documentation?

Chad
+1 955-9924

On 8 Mar 2008 at 13:11, Steph Kerman wrote:
> Does anyone have a user guide for the Westell B90-327W15-06 ADSL modem
> & wireless gateway?  This is a Verizon version.  The Westell website
> says: > This product was specifically designed for Verizon. You will
> need to > contact Verizon for any support issues. Their support number
> is > 800-567-6789.  As one would expect, even when told that the
> Westell website instructs you to contact Verizon, Verizon insists that
> Westell must be contacted for the manual, that Vz does not provide
> manuals or PDFs, but that they will support the modem live in real
> time.   Frankly, given the quality of their tech support help, it
> would probably be a lot more productive to just "do it myself" with
> the manual.
> 
> Of the nine 327W series modems listed on the Westell site, 3 of the 9
> have the above cop-out.  A 4th produces a "not found" error when
> manual downloading is attempted.  The other 3 produce this:        >
> Westell technical support is offered to Internet Service Providers, >
> independent phone companies, and distributors.  If you are an end >
> user, please contact your Internet Service Provider for technical >
> support.    or this:         > This product was specifically designed
> for Bell South.  You will need > to contact Bell South for any support
> issues.  Their support number is > 888-321-2375 (888-321-ADSL).  I
> downloaded the 2 available manuals, (A90-327W10-06 and D90-327W11-06).
>  At least superficially, they differ from each other only with respect
> to the presence or absence of the USB port.  How they differ from the
> B90-327W15-06  is anyone's guess. 
> 
> Given that the two available manuals are 187 and 157 pages
> respectively, I suspect that they contain many more pearls of wisdom
> than can be conveyed in a lifetime verbally by a Vz "tech support"
> person and would like to get the correct manual for this modem.   Has
> anyone succeeded in getting the correct PDF for the B90-327W15-06?
> 
> Thanks
> Steph


From stfkerman at jps.net  Sun Mar  9 12:21:06 2008
From: stfkerman at jps.net (Steph Kerman)
Date: Sun, 09 Mar 2008 13:21:06 -0400
Subject: [VoIP] More of the usual Verizon lameness
In-Reply-To: <47D3E0F5.6395.12D6836@localhost>
References: <47D3E0F5.6395.12D6836@localhost>
Message-ID: <47D41C82.2000308@jps.net>

No specific question at this time.  Just want to have the doc in 
anticipation of future needs and to peruse to see what's there.  I doubt 
Vz ahem.... "tech support" (probably neither) would want to spend a few 
hours telling me about it in lieu of my being able to browse it.  And I 
wouldn't find that productive either.

Thanks,
Steph

Chad Perkins wrote:
> I have a VZ supplied A90-327W15-06 (among others) and am 
> familiar with their products.  Did you have question about something 
> the product, or just lamenting the lack of provided documentation?
>
> Chad
> +1 955-9924
>
> On 8 Mar 2008 at 13:11, Steph Kerman wrote:
>   
>> Does anyone have a user guide for the Westell B90-327W15-06 ADSL modem
>> & wireless gateway?  This is a Verizon version.  The Westell website
>> says: > This product was specifically designed for Verizon. You will
>> need to > contact Verizon for any support issues. Their support number
>> is > 800-567-6789.  As one would expect, even when told that the
>> Westell website instructs you to contact Verizon, Verizon insists that
>> Westell must be contacted for the manual, that Vz does not provide
>> manuals or PDFs, but that they will support the modem live in real
>> time.   Frankly, given the quality of their tech support help, it
>> would probably be a lot more productive to just "do it myself" with
>> the manual.
>>
>> Of the nine 327W series modems listed on the Westell site, 3 of the 9
>> have the above cop-out.  A 4th produces a "not found" error when
>> manual downloading is attempted.  The other 3 produce this:        >
>> Westell technical support is offered to Internet Service Providers, >
>> independent phone companies, and distributors.  If you are an end >
>> user, please contact your Internet Service Provider for technical >
>> support.    or this:         > This product was specifically designed
>> for Bell South.  You will need > to contact Bell South for any support
>> issues.  Their support number is > 888-321-2375 (888-321-ADSL).  I
>> downloaded the 2 available manuals, (A90-327W10-06 and D90-327W11-06).
>>  At least superficially, they differ from each other only with respect
>> to the presence or absence of the USB port.  How they differ from the
>> B90-327W15-06  is anyone's guess. 
>>
>> Given that the two available manuals are 187 and 157 pages
>> respectively, I suspect that they contain many more pearls of wisdom
>> than can be conveyed in a lifetime verbally by a Vz "tech support"
>> person and would like to get the correct manual for this modem.   Has
>> anyone succeeded in getting the correct PDF for the B90-327W15-06?
>>
>> Thanks
>> Steph
>>     
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From chad at maine.edu  Mon Mar 10 19:13:10 2008
From: chad at maine.edu (Chad Perkins)
Date: Mon, 10 Mar 2008 20:13:10 -0400
Subject: [VoIP] InnoMedia MTA6328-2Re ATA working on CNET
Message-ID: <47D59656.4800.57DD14@localhost>

The InnoMedia MTA6328-2Re is a in-line (Ethernet) two port FXS ATA that supports 
dial pulse.  This unit is normally sold as a re-branded product and may have been 
locked by the provider (i.e. the now defunct SunRocket).  I got it for a penny plus 
shipping on eBay.

I had to modify the digit plan to allow calls to non-standard North American CNET 
numbers by adding |1[2-9]XXXXXX.T|1[2-9]XXXXXX.# to the end of the ATA's digit 
plan.  Here is the rest of my set up:

http://maine.maine.edu/~chad/mta6328.html

Chad

From ian at uax.org.uk  Tue Mar 11 03:54:48 2008
From: ian at uax.org.uk (Ian Jolly)
Date: Tue, 11 Mar 2008 08:54:48 -0000
Subject: [VoIP] Rheinigidale 1
Message-ID: <C83EFA9883B94409BECDC5765CD44A9C@IanPC>


The UK's last single digit telephone number lives again!

Not many will realise that this number was in service in the UK's public telephone network until c1991.  Only once it was automated, was British Telecom able to say that their network was fully automatic.  

The line was only ever accessible via the Operator.  

Dial the Operator on 100 (if you've set it up on your Asterisk) or +44100 from Overseas and when 'Ethel' asks 'Number Please'   say  "Rheinigidale 1"  and hopefully you'll be put through.  It seems to be about 95% accurate.  If she doesn't recognise you first time, you should get a second chance.

Not only has the number been 'preserved' but the person who answers will be the person whose house it was in !   Can't get better preservation than that?   She is available 24/7 to answer your calls.

This the location of Rheinigidale http://www.flashearth.com/?lat=57.922278&lon=-6.682643&z=10.6&r=0&src=msl  and it is only in recent years that a proper road has been built to the village - previously it was only accessible by a track over the mountains.  The original telephone worked over a VHF radio link to the mainland and the line was then extended to the auto-manual switchboard at Inverness some 100+ miles away.

The audio comes from a BBC Radio programme made in 2004 called "The Secret Life of Telephone Numbers".  It was to have been a 5 minute long 'filler' but after talking to the the producer, I convinced him that there was a little more to telephone numbers and it became a half hour long programme!  It also included the sound track from the video I took during the change-over of the UK's public networks last Electro-Mechanical Exchange on the UK's most remote inhabited island - http://www.flashearth.com/?lat=60.227104&lon=-2.175248&z=7&r=0&src=ol  (The tiny island way to the west of the Shetland Islands)  I hope before too long to have the exchange connected to CNET).

Ian Jolly


From ian at uax.org.uk  Tue Mar 11 07:25:37 2008
From: ian at uax.org.uk (Ian Jolly)
Date: Tue, 11 Mar 2008 12:25:37 -0000
Subject: [VoIP] Foula - Britain's Most Remote Inhabited Island and Britain's
	Last Public EM Exchange
Message-ID: <C6F44B1341C14FF981B5C9B5343F5AB5@IanPC>

Another line now connected  0393 3 32 27  /  +44 393 332 27    "Mrs Elizabeth Houlborn" will tell you a little about the telephones on Foula!

Other numbers on Foula 0393 3 32 XX have the original British Telecom 'Changed Code and Number Announcement' on them informing callers than Foula had lost its own STD/Area code and the numbers had changed to 6 digit numbers.  Foula had the STD Code 0393 3  and the numbers were shown as four digit numbers in the Telephone Directory - all being shown as 32XX.  However only the last two digits were used on the exchange to dial from one line to another.  Anyone on the island dialling the four digit Foula 'directory' number always reached  Foula 32 !!

I hope to have the actual exchange connected to CNET in a year or so.  

Ian Jolly

From dfroula at sbcglobal.net  Tue Mar 11 07:33:50 2008
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 11 Mar 2008 05:33:50 -0700 (PDT)
Subject: [VoIP] Foula - Britain's Most Remote Inhabited Island and
	Britain's Last Public EM Exchange
In-Reply-To: <C6F44B1341C14FF981B5C9B5343F5AB5@IanPC>
Message-ID: <908653.57789.qm@web83204.mail.mud.yahoo.com>

Hmm...wonder if this is my ancestral home.... :-)

Don Froula
1-762

--- Ian Jolly <ian at uax.org.uk> wrote:

> Another line now connected  0393 3 32 27  /  +44 393
> 332 27    "Mrs Elizabeth Houlborn" will tell you a
> little about the telephones on Foula!
> 
> Other numbers on Foula 0393 3 32 XX have the
> original British Telecom 'Changed Code and Number
> Announcement' on them informing callers than Foula
> had lost its own STD/Area code and the numbers had
> changed to 6 digit numbers.  Foula had the STD Code
> 0393 3  and the numbers were shown as four digit
> numbers in the Telephone Directory - all being shown
> as 32XX.  However only the last two digits were used
> on the exchange to dial from one line to another. 
> Anyone on the island dialling the four digit Foula
> 'directory' number always reached  Foula 32 !!
> 
> I hope to have the actual exchange connected to CNET
> in a year or so.  
> 
> Ian Jolly
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From spenadel at gmail.com  Tue Mar 11 07:56:30 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Tue, 11 Mar 2008 08:56:30 -0400
Subject: [VoIP] Rheinigidale 1
In-Reply-To: <C83EFA9883B94409BECDC5765CD44A9C@IanPC>
References: <C83EFA9883B94409BECDC5765CD44A9C@IanPC>
Message-ID: <028c01c88377$5432ee70$fc98cb50$@com>

Very cool Ian.  What do you use for the voice recognition?

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Ian Jolly
Sent: Tuesday, March 11, 2008 4:55 AM
To: CNET VoIP
Subject: [VoIP] Rheinigidale 1


The UK's last single digit telephone number lives again!

Not many will realise that this number was in service in the UK's public
telephone network until c1991.  Only once it was automated, was British
Telecom able to say that their network was fully automatic.  

The line was only ever accessible via the Operator.  

Dial the Operator on 100 (if you've set it up on your Asterisk) or +44100
from Overseas and when 'Ethel' asks 'Number Please'   say  "Rheinigidale 1"
and hopefully you'll be put through.  It seems to be about 95% accurate.  If
she doesn't recognise you first time, you should get a second chance.

Not only has the number been 'preserved' but the person who answers will be
the person whose house it was in !   Can't get better preservation than
that?   She is available 24/7 to answer your calls.

This the location of Rheinigidale
http://www.flashearth.com/?lat=57.922278&lon=-6.682643&z=10.6&r=0&src=msl
and it is only in recent years that a proper road has been built to the
village - previously it was only accessible by a track over the mountains.
The original telephone worked over a VHF radio link to the mainland and the
line was then extended to the auto-manual switchboard at Inverness some 100+
miles away.

The audio comes from a BBC Radio programme made in 2004 called "The Secret
Life of Telephone Numbers".  It was to have been a 5 minute long 'filler'
but after talking to the the producer, I convinced him that there was a
little more to telephone numbers and it became a half hour long programme!
It also included the sound track from the video I took during the
change-over of the UK's public networks last Electro-Mechanical Exchange on
the UK's most remote inhabited island -
http://www.flashearth.com/?lat=60.227104&lon=-2.175248&z=7&r=0&src=ol  (The
tiny island way to the west of the Shetland Islands)  I hope before too long
to have the exchange connected to CNET).

Ian Jolly

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


From spenadel at gmail.com  Tue Mar 11 08:05:33 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Tue, 11 Mar 2008 09:05:33 -0400
Subject: [VoIP] Telephone Reminders now on line
Message-ID: <028d01c88378$993d6170$cbb82450$@com>

Hi all,

 

I've adapted a program from Nerd Vittles to standard Asterisk that allows
users to schedule telephone reminders.  Here's how it works:

 

You dial 888-8888 and get greeted by Allison.

Record the message that you want played back to you.

Enter your phone number for the reminder to be sent

Enter delivery date (pressing # defaults to today).

Enter delivery time

Enter whether it's a one time or recurring reminder.

Hang up.

 

Recurring messages can be cancelled through the system.

 

Have fun!

Lee

 

 


From voiptandem at shaneyoung.com  Tue Mar 11 08:23:43 2008
From: voiptandem at shaneyoung.com (Shane Young)
Date: Tue, 11 Mar 2008 08:23:43 -0500
Subject: [VoIP] Telephone Reminders now on line
In-Reply-To: <028d01c88378$993d6170$cbb82450$@com>
References: <028d01c88378$993d6170$cbb82450$@com>
Message-ID: <20080311082343.68d1hmfkwg8gg444@secure.shaneyoung.com>

When I was a kid, the town next to us had a Stromberg XY board.  To  
call a party on your party line, you'd dial 8+XXXX where XXXX was the  
other party's phone, then hang up.  The other party's phone would  
ring, and I assume yours would too, but I don't recall the details.

All I really remember was the "bug" in the system, where you could  
dial 8 and Anyone's number (in that exchange) and their phone would  
ring.

So, I could call into this system and enter somone elses number and  
deliver a message at 03:30 AM ?

Sounds like fun!!!! :)



Quoting Lee Spenadel <spenadel at gmail.com>:

> Hi all,
>
>
>
> I've adapted a program from Nerd Vittles to standard Asterisk that allows
> users to schedule telephone reminders.  Here's how it works:
>
>
>
> You dial 888-8888 and get greeted by Allison.
>
> Record the message that you want played back to you.
>
> Enter your phone number for the reminder to be sent
>
> Enter delivery date (pressing # defaults to today).
>
> Enter delivery time
>
> Enter whether it's a one time or recurring reminder.
>
> Hang up.
>
>
>
> Recurring messages can be cancelled through the system.
>
>
>
> Have fun!
>
> Lee
>
>
>
>
>
> _______________________________________________
> 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 madmanmarkau at hotmail.com  Tue Mar 11 08:38:10 2008
From: madmanmarkau at hotmail.com (Mad Mark)
Date: Tue, 11 Mar 2008 13:38:10 +0000
Subject: [VoIP] Telephone Reminders now on line
In-Reply-To: <20080311082343.68d1hmfkwg8gg444@secure.shaneyoung.com>
References: <028d01c88378$993d6170$cbb82450$@com>
	<20080311082343.68d1hmfkwg8gg444@secure.shaneyoung.com>
Message-ID: <BAY130-W22900056892831CA820300C40F0@phx.gbl>


I believe the same thing used to happen a lot in the American phone system. The Evan Doorbell tapes on http://www.phonetrips.com demonstrate this a few times. Also, you could use this circuit to call someone else's line with you connected to it. I remember one instance where Evan Doorbell used the system to dial the Operator on 0, then by dialing three other digits corresponding to a vacant code, he superimposed a reorder on the line. He jokingly asks the operator what's going on and why there is a reorder on the line, before hanging up.

Some systems didn't require you to dial the last four digits. For instance, in his fiends' home exchange in the Long Island area of New York, all you had to dial was 660. This circuit was also used as a party line. However, on his own home exchange, he had to dial 660 and his last four digits.

His friends' 660 could be used as a party line, albeit with significant attenuation on the line. However, his own home exchange could not. Evan Doorbell's home exchange was a Crossbar 1 exchange, and his friends' exchange was a Crossbar 5.

I wonder if your 8+XXXX callback circuit could have been used as a "party line"...

> Date: Tue, 11 Mar 2008 08:23:43 -0500
> From: voiptandem at shaneyoung.com
> To: voip at ckts.info
> Subject: Re: [VoIP] Telephone Reminders now on line
> 
> When I was a kid, the town next to us had a Stromberg XY board.  To  
> call a party on your party line, you'd dial 8+XXXX where XXXX was the  
> other party's phone, then hang up.  The other party's phone would  
> ring, and I assume yours would too, but I don't recall the details.
> 
> All I really remember was the "bug" in the system, where you could  
> dial 8 and Anyone's number (in that exchange) and their phone would  
> ring.
> 
> So, I could call into this system and enter somone elses number and  
> deliver a message at 03:30 AM ?
> 
> Sounds like fun!!!! :)
> 
> 
> 
> Quoting Lee Spenadel <spenadel at gmail.com>:
> 
> > Hi all,
> >
> >
> >
> > I've adapted a program from Nerd Vittles to standard Asterisk that allows
> > users to schedule telephone reminders.  Here's how it works:
> >
> >
> >
> > You dial 888-8888 and get greeted by Allison.
> >
> > Record the message that you want played back to you.
> >
> > Enter your phone number for the reminder to be sent
> >
> > Enter delivery date (pressing # defaults to today).
> >
> > Enter delivery time
> >
> > Enter whether it's a one time or recurring reminder.
> >
> > Hang up.
> >
> >
> >
> > Recurring messages can be cancelled through the system.
> >
> >
> >
> > Have fun!
> >
> > Lee
> >
> >
> >
> >
> >
> > _______________________________________________
> > 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/

_________________________________________________________________
It's simple! Sell your car for just $30 at CarPoint.com.au
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From spenadel at gmail.com  Tue Mar 11 09:42:38 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Tue, 11 Mar 2008 10:42:38 -0400
Subject: [VoIP] Telephone Reminders now on line
In-Reply-To: <20080311082343.68d1hmfkwg8gg444@secure.shaneyoung.com>
References: <028d01c88378$993d6170$cbb82450$@com>
	<20080311082343.68d1hmfkwg8gg444@secure.shaneyoung.com>
Message-ID: <02a701c88386$28fee970$7afcbc50$@com>

True, but abuse will have me move access to the internal context.

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Shane Young
Sent: Tuesday, March 11, 2008 9:24 AM
To: voip at ckts.info
Subject: Re: [VoIP] Telephone Reminders now on line

When I was a kid, the town next to us had a Stromberg XY board.  To  
call a party on your party line, you'd dial 8+XXXX where XXXX was the  
other party's phone, then hang up.  The other party's phone would  
ring, and I assume yours would too, but I don't recall the details.

All I really remember was the "bug" in the system, where you could  
dial 8 and Anyone's number (in that exchange) and their phone would  
ring.

So, I could call into this system and enter somone elses number and  
deliver a message at 03:30 AM ?

Sounds like fun!!!! :)



Quoting Lee Spenadel <spenadel at gmail.com>:

> Hi all,
>
>
>
> I've adapted a program from Nerd Vittles to standard Asterisk that allows
> users to schedule telephone reminders.  Here's how it works:
>
>
>
> You dial 888-8888 and get greeted by Allison.
>
> Record the message that you want played back to you.
>
> Enter your phone number for the reminder to be sent
>
> Enter delivery date (pressing # defaults to today).
>
> Enter delivery time
>
> Enter whether it's a one time or recurring reminder.
>
> Hang up.
>
>
>
> Recurring messages can be cancelled through the system.
>
>
>
> Have fun!
>
> Lee
>
>
>
>
>
> _______________________________________________
> 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/


From spenadel at gmail.com  Wed Mar 12 21:23:00 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Wed, 12 Mar 2008 22:23:00 -0400
Subject: [VoIP] Cepstral Help
Message-ID: <04f301c884b1$2a236660$7e6a3320$@com>

I'm trying to change the default voice from David to Allison.  Issue the
command:

 

 

swift --order -name Allison  at the command prompt, yet when I issue a swift
-voices command, David is still the default.  A sample TTS still has David
reading back the test.  

 

I know that I could force Allison, but a program that I'm using utilizes the
system default and there is no provision in that program to change it.  Am I
missing something?

 

Thanks

Lee

 


From voiptandem at shaneyoung.com  Wed Mar 12 22:43:54 2008
From: voiptandem at shaneyoung.com (Shane Young)
Date: Wed, 12 Mar 2008 22:43:54 -0500
Subject: [VoIP] Cepstral Help
In-Reply-To: <04f301c884b1$2a236660$7e6a3320$@com>
References: <04f301c884b1$2a236660$7e6a3320$@com>
Message-ID: <20080312224354.bw3293hnlcok8kc4@secure.shaneyoung.com>

 From their web site:

  * How do I select a new default voice for command-line swift?

To select the default voice for Swift in Linux, set the environment  
variable "SWIFT_DEFAULT_VOICE" to the name of the desired voice. For  
example:

export SWIFT_DEFAULT_VOICE=David

To save this environment variable between terminal sessions, add the  
above line to your home directory's .bash_profile file.



Now, keep in mind that this is all relative to the user running  
"swift".  If it runs as the webserver, then you have some more  
fiddling around to do.

I seem to recall that the default voice is either the first or the  
most-recent voice installed.  You could try un-installing and  
re-installing the voices in a different order as well.

It seems odd that you can't tell your program what voice to use.  Is  
this a pre-compiled binary that you can't edit?




Quoting Lee Spenadel <spenadel at gmail.com>:

> I'm trying to change the default voice from David to Allison.  Issue the
> command:
>
>
>
>
>
> swift --order -name Allison  at the command prompt, yet when I issue a swift
> -voices command, David is still the default.  A sample TTS still has David
> reading back the test.
>
>
>
> I know that I could force Allison, but a program that I'm using utilizes the
> system default and there is no provision in that program to change it.  Am I
> missing something?
>
>
>
> Thanks
>
> Lee
>
>
>
> _______________________________________________
> 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 spenadel at gmail.com  Wed Mar 12 22:54:00 2008
From: spenadel at gmail.com (Lee Spenadel)
Date: Wed, 12 Mar 2008 23:54:00 -0400
Subject: [VoIP] Cepstral Help
In-Reply-To: <20080312224354.bw3293hnlcok8kc4@secure.shaneyoung.com>
References: <04f301c884b1$2a236660$7e6a3320$@com>
	<20080312224354.bw3293hnlcok8kc4@secure.shaneyoung.com>
Message-ID: <050901c884bd$df848af0$9e8da0d0$@com>

I scoured their website and didn't see this.  It's not precompiled code,
I've tweaked it to run on Asterisk as opposed to PBX in a Flash.  I've gone
through the scripts and saw no references to the swift application and no
references to voices.

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Shane Young
Sent: Wednesday, March 12, 2008 11:44 PM
To: voip at ckts.info
Subject: Re: [VoIP] Cepstral Help

 From their web site:

  * How do I select a new default voice for command-line swift?

To select the default voice for Swift in Linux, set the environment  
variable "SWIFT_DEFAULT_VOICE" to the name of the desired voice. For  
example:

export SWIFT_DEFAULT_VOICE=David

To save this environment variable between terminal sessions, add the  
above line to your home directory's .bash_profile file.



Now, keep in mind that this is all relative to the user running  
"swift".  If it runs as the webserver, then you have some more  
fiddling around to do.

I seem to recall that the default voice is either the first or the  
most-recent voice installed.  You could try un-installing and  
re-installing the voices in a different order as well.

It seems odd that you can't tell your program what voice to use.  Is  
this a pre-compiled binary that you can't edit?




Quoting Lee Spenadel <spenadel at gmail.com>:

> I'm trying to change the default voice from David to Allison.  Issue the
> command:
>
>
>
>
>
> swift --order -name Allison  at the command prompt, yet when I issue a
swift
> -voices command, David is still the default.  A sample TTS still has David
> reading back the test.
>
>
>
> I know that I could force Allison, but a program that I'm using utilizes
the
> system default and there is no provision in that program to change it.  Am
I
> missing something?
>
>
>
> Thanks
>
> Lee
>
>
>
> _______________________________________________
> 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/


From dfroula at sbcglobal.net  Thu Mar 13 20:59:15 2008
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 13 Mar 2008 18:59:15 -0700 (PDT)
Subject: [VoIP] 12F683 PIC for Tone Generation
Message-ID: <816912.49576.qm@web83205.mail.mud.yahoo.com>

I've been fooling around with the 12F683 PIC
microcontroller for a few months. I am using the
PicBasic Pro compiler, which makes development quick
and easy, compared to assembly language.

I coded up and built a blue box circuit with 12
non-volatile memories. It uses the PicBasic Pro
"FREQOUT" command that can generate single or dual
tones using Pulse Width Modulation. With proper
filtering of the PWM, it could have all sorts of
interesting telephony applications.

Schematic, source code, and programmer hex files are
at the Binrev board:

http://www.binrev.com/forums/index.php?showtopic=36307

Don 

From t2600 at sbcglobal.net  Thu Mar 13 22:32:44 2008
From: t2600 at sbcglobal.net (Jerry Petrizze)
Date: Thu, 13 Mar 2008 19:32:44 -0800
Subject: [VoIP] 12F683 PIC for Tone Generation
References: <816912.49576.qm@web83205.mail.mud.yahoo.com>
Message-ID: <002e01c88584$1170c6c0$0501a8c0@pentium4>

Yo Don:  How does one access this stuff???
I got an error message when trying.
Jerry Petrizze


----- Original Message ----- 
From: "Donald Froula" <dfroula at sbcglobal.net>
To: <voip at ckts.info>
Sent: Thursday, March 13, 2008 5:59 PM
Subject: [VoIP] 12F683 PIC for Tone Generation


> I've been fooling around with the 12F683 PIC
> microcontroller for a few months. I am using the
> PicBasic Pro compiler, which makes development quick
> and easy, compared to assembly language.
> 
> I coded up and built a blue box circuit with 12
> non-volatile memories. It uses the PicBasic Pro
> "FREQOUT" command that can generate single or dual
> tones using Pulse Width Modulation. With proper
> filtering of the PWM, it could have all sorts of
> interesting telephony applications.
> 
> Schematic, source code, and programmer hex files are
> at the Binrev board:
> 
> http://www.binrev.com/forums/index.php?showtopic=36307
> 
> 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  Thu Mar 13 21:38:55 2008
From: stfkerman at jps.net (Steph Kerman)
Date: Thu, 13 Mar 2008 22:38:55 -0400
Subject: [VoIP] 12F683 PIC for Tone Generation
In-Reply-To: <002e01c88584$1170c6c0$0501a8c0@pentium4>
References: <816912.49576.qm@web83205.mail.mud.yahoo.com>
	<002e01c88584$1170c6c0$0501a8c0@pentium4>
Message-ID: <47D9E53F.6000007@jps.net>

Worked for me!

=SK=

Jerry Petrizze wrote:
> Yo Don:  How does one access this stuff???
> I got an error message when trying.
> Jerry Petrizze
>
>
> ----- Original Message ----- 
> From: "Donald Froula" <dfroula at sbcglobal.net>
> To: <voip at ckts.info>
> Sent: Thursday, March 13, 2008 5:59 PM
> Subject: [VoIP] 12F683 PIC for Tone Generation
>
>
>   
>> I've been fooling around with the 12F683 PIC
>> microcontroller for a few months. I am using the
>> PicBasic Pro compiler, which makes development quick
>> and easy, compared to assembly language.
>>
>> I coded up and built a blue box circuit with 12
>> non-volatile memories. It uses the PicBasic Pro
>> "FREQOUT" command that can generate single or dual
>> tones using Pulse Width Modulation. With proper
>> filtering of the PWM, it could have all sorts of
>> interesting telephony applications.
>>
>> Schematic, source code, and programmer hex files are
>> at the Binrev board:
>>
>> http://www.binrev.com/forums/index.php?showtopic=36307
>>
>> 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 dfroula at sbcglobal.net  Thu Mar 13 23:06:53 2008
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 13 Mar 2008 21:06:53 -0700 (PDT)
Subject: [VoIP] 12F683 PIC for Tone Generation
In-Reply-To: <002e01c88584$1170c6c0$0501a8c0@pentium4>
Message-ID: <627694.61188.qm@web83205.mail.mud.yahoo.com>

Sorry, Jerry. I guess you have to be logged in to
access the files.

Here are the direct links:

schematic:
http://www.geocities.com/donfroula/pwm_bb_schematic.gif

prototype pic:
http://www.geocities.com/donfroula/IMG_3313.JPG

programmer pic:
http://www.geocities.com/donfroula/IMG_3317.JPG

source code:
http://www.geocities.com/donfroula/ADC_blue_box_09.txt

hex file for programmer:
http://www.geocities.com/donfroula/ADC_blue_box_09.hex.txt

Don

--- Jerry Petrizze <t2600 at sbcglobal.net> wrote:

> Yo Don:  How does one access this stuff???
> I got an error message when trying.
> Jerry Petrizze
> 
> 
> ----- Original Message ----- 
> From: "Donald Froula" <dfroula at sbcglobal.net>
> To: <voip at ckts.info>
> Sent: Thursday, March 13, 2008 5:59 PM
> Subject: [VoIP] 12F683 PIC for Tone Generation
> 
> 
> > I've been fooling around with the 12F683 PIC
> > microcontroller for a few months. I am using the
> > PicBasic Pro compiler, which makes development
> quick
> > and easy, compared to assembly language.
> > 
> > I coded up and built a blue box circuit with 12
> > non-volatile memories. It uses the PicBasic Pro
> > "FREQOUT" command that can generate single or dual
> > tones using Pulse Width Modulation. With proper
> > filtering of the PWM, it could have all sorts of
> > interesting telephony applications.
> > 
> > Schematic, source code, and programmer hex files
> are
> > at the Binrev board:
> > 
> >
>
http://www.binrev.com/forums/index.php?showtopic=36307
> > 
> > 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 dfroula at sbcglobal.net  Thu Mar 13 23:13:08 2008
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 13 Mar 2008 21:13:08 -0700 (PDT)
Subject: [VoIP] 12F683 PIC for Tone Generation
In-Reply-To: <627694.61188.qm@web83205.mail.mud.yahoo.com>
Message-ID: <639906.21469.qm@web83201.mail.mud.yahoo.com>

Other versions of code using the same circuit
schematic at:

http://www.geocities.com/donfroula/

in the ADC_blue_box... files.

Note, the circuit uses a single pin and resistor
voltage divider to interface all 13 keys of the
keypad.

Other stuff from my older 68HC705C8 design as well.

Don

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

> Sorry, Jerry. I guess you have to be logged in to
> access the files.
> 
> Here are the direct links:
> 
> schematic:
>
http://www.geocities.com/donfroula/pwm_bb_schematic.gif
> 
> prototype pic:
> http://www.geocities.com/donfroula/IMG_3313.JPG
> 
> programmer pic:
> http://www.geocities.com/donfroula/IMG_3317.JPG
> 
> source code:
>
http://www.geocities.com/donfroula/ADC_blue_box_09.txt
> 
> hex file for programmer:
>
http://www.geocities.com/donfroula/ADC_blue_box_09.hex.txt
> 
> Don
> 
> --- Jerry Petrizze <t2600 at sbcglobal.net> wrote:
> 
> > Yo Don:  How does one access this stuff???
> > I got an error message when trying.
> > Jerry Petrizze
> > 
> > 
> > ----- Original Message ----- 
> > From: "Donald Froula" <dfroula at sbcglobal.net>
> > To: <voip at ckts.info>
> > Sent: Thursday, March 13, 2008 5:59 PM
> > Subject: [VoIP] 12F683 PIC for Tone Generation
> > 
> > 
> > > I've been fooling around with the 12F683 PIC
> > > microcontroller for a few months. I am using the
> > > PicBasic Pro compiler, which makes development
> > quick
> > > and easy, compared to assembly language.
> > > 
> > > I coded up and built a blue box circuit with 12
> > > non-volatile memories. It uses the PicBasic Pro
> > > "FREQOUT" command that can generate single or
> dual
> > > tones using Pulse Width Modulation. With proper
> > > filtering of the PWM, it could have all sorts of
> > > interesting telephony applications.
> > > 
> > > Schematic, source code, and programmer hex files
> > are
> > > at the Binrev board:
> > > 
> > >
> >
>
http://www.binrev.com/forums/index.php?showtopic=36307
> > > 
> > > 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 ian at uax.org.uk  Fri Mar 14 04:22:51 2008
From: ian at uax.org.uk (Ian Jolly)
Date: Fri, 14 Mar 2008 09:22:51 -0000
Subject: [VoIP] New exchange  connects   +44 295 76 XXX
Message-ID: <7D257557EAE0478B892E026020610862@IanPC>

Welcome to Kierien deep in rural Herefordshire - another UAX13 just about connected to CNET.  On 0295 76 XXX  as Sulgrave - with the Asterisk being set up as its GSC - Banbury (0295 XXXX).

We spent yesterday playing with a new asterisk box and connecting it up to a new broadband connection.  

On the initial attempt to load Linux/asterisk a week or two ago, we had no success.  So I brought the PC home and Peter Duffield offered to have a go with it.  It ended up being a problem with the hard disk if I remember correctly but Peter cracked it getting the programs loaded and set up the TDM card - a ZapMicro version of the TDM400 with one FXS and one FXO - working.

I then wrote the configs and had the Asterisk working no problem at home -  

But it wouldn't dial out when connected at Kierien's.  Whilst me laptop with its softphone of Mold (0352) worked OK and we had all the relevant ports open/forwarded still no lick.  It turned out to be a problem which Mr Asterisk came up with the answer for - Tnx Jon - I'd have never found it !!  I'll remember next time.  

We had a number of other minor problems as well but after a pub lunch (at about 3pm to 4pm!) all was eventually sorted out except for one problem!  One problem was traced down to the old mechanical "Regenerator No1" misdialling. That was cured by replacing them with No 5 electronic regenerators.  The outgoing calls go out via Level 1/0 relay sets which re-insert the previously dialled digits. This makes writing the configs a bit easier!  Another was a lack of 48 volts on the FXS - that turned out to be a faulty plug on the patch lead which we were using to connect it to the UAX.

Incoming calls are pulsed from the FXO port into the UAX on an incoming Group Selector (with Level '0' barred - as it should be!) .  The calls to 290 rang the telephone but on 'Called Sub Answer' most of the calls drop out immediately.  A similar thing happens on 299 - inverted ring tone was received from the UAX or the call drops out. On several occasions, a call was setup to 290 but suddenly part way through the call it would drop out.  

However time ran out and I had an over two hour drive home and managed to get home just before my coach turned into a pumpkin!

The Asterisk may not be up and running as Kierien has quite a bit of tidying up of the cabling - we just had things lashed up to try it out.  

The UAX will be moving later in the year so but it is another welcome addition to CNET. The U13 was originally at Sulgrave in Oxfordshire parented on Banbury (0295) off Level 76.  It currently consisted of a C unit, two A units and a B unit fitted with the junction hunters and Level 1/0 relaysets plus a Test Selector.  It is all housed in the nearest replica of an old GPO brick 'A' type building that I've ever seen!  It is slightly longer than an 'A' type building as it has a 'power room' with Lister diesel generator fitted!  

Hopefully we'll have cracked the remaining problem before too long.  Meanwhile the Asterisk may be down whilst improvements are made.

Ian

From hockd at dteenergy.com  Fri Mar 14 04:45:27 2008
From: hockd at dteenergy.com (Dennis D Hock)
Date: Fri, 14 Mar 2008 05:45:27 -0400
Subject: [VoIP] New exchange  connects   +44 295 76 XXX
In-Reply-To: <7D257557EAE0478B892E026020610862@IanPC>
References: <7D257557EAE0478B892E026020610862@IanPC>
Message-ID: <OF0FD0D7BC.253ADE43-ON8525740C.003599AD-8525740C.003599B0@dteenergy.com>

Congrats Ian I am sure you will get the bugs worked out in short order.
 
Dennis Hock

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


To: "CNET VoIP" <voip at ckts.info>, <CNET-UK-I at yahoogroups.com>
From: "Ian Jolly" <ian at uax.org.uk>
Sent by: voip-bounces at ckts.info
Date: 03/14/2008 05:22AM
Subject: [VoIP] New exchange connects +44 295 76 XXX

Welcome to Kierien deep in rural Herefordshire - another UAX13 just about connected to CNET.  On 0295 76 XXX  as Sulgrave - with the Asterisk being set up as its GSC - Banbury (0295 XXXX).

We spent yesterday playing with a new asterisk box and connecting it up to a new broadband connection.  

On the initial attempt to load Linux/asterisk a week or two ago, we had no success.  So I brought the PC home and Peter Duffield offered to have a go with it.  It ended up being a problem with the hard disk if I remember correctly but Peter cracked it getting the programs loaded and set up the TDM card - a ZapMicro version of the TDM400 with one FXS and one FXO - working.

I then wrote the configs and had the Asterisk working no problem at home -  

But it wouldn't dial out when connected at Kierien's.  Whilst me laptop with its softphone of Mold (0352) worked OK and we had all the relevant ports open/forwarded still no lick.  It turned out to be a problem which Mr Asterisk came up with the answer for - Tnx Jon - I'd have never found it !!  I'll remember next time.  

We had a number of other minor problems as well but after a pub lunch (at about 3pm to 4pm!) all was eventually sorted out except for one problem!  One problem was traced down to the old mechanical "Regenerator No1" misdialling. That was cured by replacing them with No 5 electronic regenerators.  The outgoing calls go out via Level 1/0 relay sets which re-insert the previously dialled digits. This makes writing the configs a bit easier!  Another was a lack of 48 volts on the FXS - that turned out to be a faulty plug on the patch lead which we were using to connect it to the UAX.

Incoming calls are pulsed from the FXO port into the UAX on an incoming Group Selector (with Level '0' barred - as it should be!) .  The calls to 290 rang the telephone but on 'Called Sub Answer' most of the calls drop out immediately.  A similar thing happens on 299 - inverted ring tone was received from the UAX or the call drops out. On several occasions, a call was setup to 290 but suddenly part way through the call it would drop out.  

However time ran out and I had an over two hour drive home and managed to get home just before my coach turned into a pumpkin!

The Asterisk may not be up and running as Kierien has quite a bit of tidying up of the cabling - we just had things lashed up to try it out.  

The UAX will be moving later in the year so but it is another welcome addition to CNET. The U13 was originally at Sulgrave in Oxfordshire parented on Banbury (0295) off Level 76.  It currently consisted of a C unit, two A units and a B unit fitted with the junction hunters and Level 1/0 relaysets plus a Test Selector.  It is all housed in the nearest replica of an old GPO brick 'A' type building that I've ever seen!  It is slightly longer than an 'A' type building as it has a 'power room' with Lister diesel generator fitted!  

Hopefully we'll have cracked the remaining problem before too long.  Meanwhile the Asterisk may be down whilst improvements are made.

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

From cbgraf at sbcglobal.net  Fri Mar 14 10:11:46 2008
From: cbgraf at sbcglobal.net (Charles Graf)
Date: Fri, 14 Mar 2008 08:11:46 -0700 (PDT)
Subject: [VoIP] Cisco MC3810-V3 Question(S)
In-Reply-To: <20080226170745.fte74b446ck0gkc4@secure.shaneyoung.com>
Message-ID: <80811.6772.qm@web80504.mail.mud.yahoo.com>

Hi Shane,
    Thank you for the cable. Is the Lyndale Ave a good address to send reimbursement for the postage?

    Do you know if I will still need a Linix box to use the Cisco 3810 on CNET?

Charles

Shane Young <voiptandem at shaneyoung.com> wrote: Charles

I assume by "terminal" you mean what "console" for the 3810.

The pin-outs are on Cisco's site.

If you'd like a Cisco console cable, send me your address and I'll  
mail one to you.  They come packaged with every router and switch, so  
we have boxes of them lying around here..

--Shane

Quoting Charles Graf :

> Hi All,
>     I have a Cisco MC 3810 V3 with 4 FXS and 2 FXO jacks, an   
> Ethernet, T1 and E1 jacks. Will I need an additional computer   
> running Asterisk to get this box on line? I am not really concerned   
> about having a fully functional system with voice mail, call   
> routing, etc. All I wish to do is connect it to a demo SxS switch   
> and maybe a plane Jane desk phone or two. I have DSL service.
>
>     Does any one have a wiring diagram/pin outs for the terminal cable?
>
> Thank You
> Charles
>
> PS, I have reserved the prefix 653-xxxx for my switch.
>
>
> _______________________________________________
> 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/


From voiptandem at shaneyoung.com  Fri Mar 14 10:47:52 2008
From: voiptandem at shaneyoung.com (Shane Young)
Date: Fri, 14 Mar 2008 10:47:52 -0500
Subject: [VoIP] Cisco MC3810-V3 Question(S)
In-Reply-To: <80811.6772.qm@web80504.mail.mud.yahoo.com>
References: <80811.6772.qm@web80504.mail.mud.yahoo.com>
Message-ID: <20080314104752.corly1o7k8cwo8kg@secure.shaneyoung.com>

Charles

Don't worry about the postage, sorry it took me so long to get it in the mail!

  :)

As I mentioned before, the technically less complicated route would be  
for you to get your own Asterisk box.  Without it, we'll have to get  
your 3810 to tandem from somone elses, which will require a bit of  
work in your firewall or router.

How comfortable are you with editing rules in your router/firewall?

Do you have a static IP address from your DSL provider or do you get  
one out of a pool which might change somewhat randomly.

Do you have any space or power concerns for something that could be as  
large as a standard PC?

It would be kind of an interesting experiment to see if we can get it  
to work off of someone elses tandem.  I think it would be a good way  
for more people to get involved.

Let me know about the questions above and we can go from there..  I  
have a tandem in a data center we could connect you to and try it out.



--Shane


Quoting Charles Graf <cbgraf at sbcglobal.net>:

> Hi Shane,
>     Thank you for the cable. Is the Lyndale Ave a good address to   
> send reimbursement for the postage?
>
>     Do you know if I will still need a Linix box to use the Cisco   
> 3810 on CNET?
>
> Charles
>
> Shane Young <voiptandem at shaneyoung.com> wrote: Charles
>
> I assume by "terminal" you mean what "console" for the 3810.
>
> The pin-outs are on Cisco's site.
>
> If you'd like a Cisco console cable, send me your address and I'll
> mail one to you.  They come packaged with every router and switch, so
> we have boxes of them lying around here..
>
> --Shane
>
> Quoting Charles Graf :
>
>> Hi All,
>>     I have a Cisco MC 3810 V3 with 4 FXS and 2 FXO jacks, an
>> Ethernet, T1 and E1 jacks. Will I need an additional computer
>> running Asterisk to get this box on line? I am not really concerned
>> about having a fully functional system with voice mail, call
>> routing, etc. All I wish to do is connect it to a demo SxS switch
>> and maybe a plane Jane desk phone or two. I have DSL service.
>>
>>     Does any one have a wiring diagram/pin outs for the terminal cable?
>>
>> Thank You
>> Charles
>>
>> PS, I have reserved the prefix 653-xxxx for my switch.
>>
>>
>> _______________________________________________
>> 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/
>
> _______________________________________________
> 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 cbgraf at sbcglobal.net  Fri Mar 14 11:52:39 2008
From: cbgraf at sbcglobal.net (Charles Graf)
Date: Fri, 14 Mar 2008 09:52:39 -0700 (PDT)
Subject: [VoIP] Cisco MC3810-V3 Question(S)
In-Reply-To: <20080314104752.corly1o7k8cwo8kg@secure.shaneyoung.com>
Message-ID: <381950.21544.qm@web80511.mail.mud.yahoo.com>

Shane,
    Space and power is not a problem for a standard computer of which I have several to choose from. Do I still need a PC interface board with a FXO/FXS function?

    I have no problems with editing the router for static routes and opening ports, etc. I am currently on dynamic IP address and will look into getting a static address, although my better half prefers a changing address... security issues with her work when she works from home.

    I have been unable to locate where to download Red Hat Linix from, but did find and got the Fedora version. I already have the Asterisk software.

    Forgive my lack of knowledge on telephony terminology, how is the term 'tandem' used in this case? Is it used as in a remote PC controls the function of my 3810 via the Internet? If that is the case, then my own PC and the 3810 with a hub between them and a connection to my router would be all I need, correct? Other then loading the software and configuration files and the usual software debugging I see mentioned on the list.

Charles

Charles

Shane Young <voiptandem at shaneyoung.com> wrote: Charles

Don't worry about the postage, sorry it took me so long to get it in the mail!

  :)

As I mentioned before, the technically less complicated route would be  
for you to get your own Asterisk box.  Without it, we'll have to get  
your 3810 to tandem from somone elses, which will require a bit of  
work in your firewall or router.

How comfortable are you with editing rules in your router/firewall?

Do you have a static IP address from your DSL provider or do you get  
one out of a pool which might change somewhat randomly.

Do you have any space or power concerns for something that could be as  
large as a standard PC?

It would be kind of an interesting experiment to see if we can get it  
to work off of someone elses tandem.  I think it would be a good way  
for more people to get involved.

Let me know about the questions above and we can go from there..  I  
have a tandem in a data center we could connect you to and try it out.



--Shane


Quoting Charles Graf :

> Hi Shane,
>     Thank you for the cable. Is the Lyndale Ave a good address to   
> send reimbursement for the postage?
>
>     Do you know if I will still need a Linix box to use the Cisco   
> 3810 on CNET?
>
> Charles
>
> Shane Young  wrote: Charles
>
> I assume by "terminal" you mean what "console" for the 3810.
>
> The pin-outs are on Cisco's site.
>
> If you'd like a Cisco console cable, send me your address and I'll
> mail one to you.  They come packaged with every router and switch, so
> we have boxes of them lying around here..
>
> --Shane
>
> Quoting Charles Graf :
>
>> Hi All,
>>     I have a Cisco MC 3810 V3 with 4 FXS and 2 FXO jacks, an
>> Ethernet, T1 and E1 jacks. Will I need an additional computer
>> running Asterisk to get this box on line? I am not really concerned
>> about having a fully functional system with voice mail, call
>> routing, etc. All I wish to do is connect it to a demo SxS switch
>> and maybe a plane Jane desk phone or two. I have DSL service.
>>
>>     Does any one have a wiring diagram/pin outs for the terminal cable?
>>
>> Thank You
>> Charles
>>
>> PS, I have reserved the prefix 653-xxxx for my switch.
>>
>>
>> _______________________________________________
>> 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/
>
> _______________________________________________
> 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

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

_______________________________________________
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Project Web Page: http://www.ckts.info/


From jnovack at stromberg-carlson.org  Sun Mar 16 08:39:54 2008
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 16 Mar 2008 09:39:54 -0400
Subject: [VoIP] Cisco MC3810-V3 Question(S)
In-Reply-To: <381950.21544.qm@web80511.mail.mud.yahoo.com>
References: <381950.21544.qm@web80511.mail.mud.yahoo.com>
Message-ID: <47DD232A.1060002@stromberg-carlson.org>



Charles Graf wrote:
>     I have been unable to locate where to download Red Hat Linix from, but did find and got the Fedora version. I already have the Asterisk software.
>   
You want CentOS - which is the Free version of RHEL

Fewer issues than the various Fedoras

I use CentOS version 3 for the 1.2 version of Asterisk virtually trouble 
free.
CentOS 4 would be required for the 1.4 version of Asterisk.
In either case, I suggest you install everything, then turn off services 
you don't need or want after the install.
When I didn't do that, later I got complaints of missing files for this 
or that, and had to go back and figure out what to put where.
Others more knowledgeable in Linux don't need to go that route

CentOS 5 is current, but I am not pleased with the install. There is no 
"install everything" and I had to go find some packages that Asterisk 
wanted after the install.

Just my opinion, worth what you paid for it!

John  Novack

-- 
Dog is my co-pilot


From chad at maine.edu  Tue Mar 18 18:57:32 2008
From: chad at maine.edu (Chad Perkins)
Date: Tue, 18 Mar 2008 19:57:32 -0400
Subject: [VoIP] Ouch, part reset, quickly restoring reality
Message-ID: <47E01EAC.23914.429357@localhost>

Message on Asterisk 1.2 verbose console:

'Ouch, part reset, quickly restoring reality'

Anyone had (and fixed) this problem?  I'm starting to think I've lost 
the battle with this Rev.F TDM400P.

Chad

From ian at uax.org.uk  Thu Mar 20 15:16:07 2008
From: ian at uax.org.uk (Ian Jolly)
Date: Thu, 20 Mar 2008 20:16:07 -0000
Subject: [VoIP] Drop out on answer on Strowger switch.
Message-ID: <73BD3699B6174DD98EF371DE756FE848@IanPC>

Hi All

We have an Asterisk with a Zapmicro (version of the Digium TDM400) with one FXS and an FXO connected to a former British Post Office UAX13 Strowger exchange. 

The FXO is connected direct to its own first group selector.  

On dialling a call into the Strowger switch, the call rings out but 90% of the time, as soon as the call is answered the call drops out.  At the point that the call is answered, the Strowger switch reverses the polarity of the line to the FXO.

On the other 10% of the calls, speech is possible for up to a minute after answering then the call drops out without warning.

We have the following in zapata.conf

answeronpolarityswitch=yes
cleardownonpolarityswitch=no

but to no effect   :-((

Anyone any ideas?

Ian J

 

From dfroula at sbcglobal.net  Fri Mar 21 12:19:38 2008
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 21 Mar 2008 10:19:38 -0700 (PDT)
Subject: [VoIP] 12F683 PIC for Tone Generation
Message-ID: <678325.39754.qm@web83201.mail.mud.yahoo.com>

I tinkered with this blue box circuit some more.

I added DTMF mode back, storage of tone mode along
with digits in dialing memories, and configurable
eeprom storage of default tone mode (MF/DTMF) and tone
duration/spacing (75 ms./120ms.). 2038 out of 2048
available program bytes were used. I had to shut off
the automatic compiler watchdog timer refresh and the
on-chip watchdog timer to get additional code space
for the enhancements.

Features:

- Supports a 12 digit keypad, plus an additonal 2600
button. All keys are SPST switch closures with a
common connection. (See schematic in this thread.)

- Both MF and DTMF are supported. When initially
programmed, the chip defaults to MF mode on powerup.
The powerup mode is stored in eeprom. This may be
temporarily toggled to the opposite mode by holding
down the 2600 key while powering up. The next powerup
reverts to the default tone mode. The default tone
mode may be persistently toggled by holding down the
"*" while powering up. This will toggle the tone mode,
but also write the new setting to eeprom so subsequent
powerups will default to the opposite tone mode. A
tone confirms the write to eeprom.

- Two tone durations/spacings are supported, 75 ms.
(default) and 120 ms. The tone duration/spacing is
stored in eeprom. The duration/spacing may be
permanently toggled by holding down the "#" key at
power up. This will toggle the duration/spacing and
write the new value to eeprom for subsequent powerups.
A tone confirms the write to eeprom. The
duration/spacing value is used in manual mode and also
in memory playback mode. The KP tone is always 120 ms.
duration, regardless of this setting. The 2600 tone
duration is always 1.5 seconds.

- There are two operating modes, normal and playback.
The chip always powers up in normal mode. Operating
modes are toggled by holding down the 2600 key for 1
second. Tones confirm the mode change. In normal mode,
dialing is manual using the current tone mode and
duration settings. In playback mode, each keypad key
will play back any stored dialing sequences.

- There are 12-32 digit dialing memories, one for each
keypad key. On power-up, a playback to normal mode
change, or a dialing memory write the chip stores the
next 32 key presses in a RAM buffer. At any point,
this buffer may be saved to one of the 12 dialing
memories by pressing and holding the corresponding key
for 2 seconds. (The digit played when initiating a
write to the memory will not be saved.) A tone
confirms the eeprom write. Memories are cleared by
storing to memory immediately after a power-up, mode
change from playback to normal, or after a previous
dialing memory save. The MF/DTMF tone mode is saved in
memory, so the dialing memories may contain a mix of
MF and DTMF sequences. If a 2600 tone is saved, a 1.5
second fixed delay will be added after the tone to
allow for a wink ack from the trunk. Then, any
additional tones will play.

Source
http://www.geocities.com/donfroula/ADC_blue_box_12.pbp.txt

.hex file for programmer
http://www.geocities.com/donfroula/ADC_blue_box_12.hex.txt

"schematic"  :-)
http://www.geocities.com/donfroula/pwm_bb_schematic.gif
--- Donald Froula <dfroula at sbcglobal.net> wrote:

> Other versions of code using the same circuit
> schematic at:
> 
> http://www.geocities.com/donfroula/
> 
> in the ADC_blue_box... files.
> 
> Note, the circuit uses a single pin and resistor
> voltage divider to interface all 13 keys of the
> keypad.
> 
> Other stuff from my older 68HC705C8 design as well.
> 
> Don
> 
> --- Donald Froula <dfroula at sbcglobal.net> wrote:
> 
> > Sorry, Jerry. I guess you have to be logged in to
> > access the files.
> > 
> > Here are the direct links:
> > 
> > schematic:
> >
>
http://www.geocities.com/donfroula/pwm_bb_schematic.gif
> > 
> > prototype pic:
> > http://www.geocities.com/donfroula/IMG_3313.JPG
> > 
> > programmer pic:
> > http://www.geocities.com/donfroula/IMG_3317.JPG
> > 
> > source code:
> >
>
http://www.geocities.com/donfroula/ADC_blue_box_09.txt
> > 
> > hex file for programmer:
> >
>
http://www.geocities.com/donfroula/ADC_blue_box_09.hex.txt
> > 
> > Don
> > 
> > --- Jerry Petrizze <t2600 at sbcglobal.net> wrote:
> > 
> > > Yo Don:  How does one access this stuff???
> > > I got an error message when trying.
> > > Jerry Petrizze
> > > 
> > > 
> > > ----- Original Message ----- 
> > > From: "Donald Froula" <dfroula at sbcglobal.net>
> > > To: <voip at ckts.info>
> > > Sent: Thursday, March 13, 2008 5:59 PM
> > > Subject: [VoIP] 12F683 PIC for Tone Generation
> > > 
> > > 
> > > > I've been fooling around with the 12F683 PIC
> > > > microcontroller for a few months. I am using
> the
> > > > PicBasic Pro compiler, which makes development
> > > quick
> > > > and easy, compared to assembly language.
> > > > 
> > > > I coded up and built a blue box circuit with
> 12
> > > > non-volatile memories. It uses the PicBasic
> Pro
> > > > "FREQOUT" command that can generate single or
> > dual
> > > > tones using Pulse Width Modulation. With
> proper
> > > > filtering of the PWM, it could have all sorts
> of
> > > > interesting telephony applications.
> > > > 
> > > > Schematic, source code, and programmer hex
> files
> > > are
> > > > at the Binrev board:
> > > > 
> > > >
> > >
> >
>
http://www.binrev.com/forums/index.php?showtopic=36307
> > > > 
> > > > 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 richardlane at exemail.com.au  Mon Mar 24 08:56:43 2008
From: richardlane at exemail.com.au (Richard Lane)
Date: Tue, 25 Mar 2008 00:56:43 +1100
Subject: [VoIP] STUN
Message-ID: <47E7B31B.6040605@exemail.com.au>

Hi

I was wondering if someone could explain stun in regards to how it works 
and how to implement it on asterisk so i can connect other members to my 
asterisk server via sip ata's

Richard

From martin at Princeton.EDU  Mon Mar 24 09:39