from:"Christopher Dobbs"





I have an embedded Linux Distro that is specifically designed to fit on
as small as a 32MB CF card.

Includes:
  HTTP Server
  SSH Client and server
  DHCPCD 
  DHCPD
  bash
  and more

Contact me off list if you are interested.

--
Christopher Dobbs

Michael Graves wrote:

  On Wed, 22 Dec 2004 12:39:46 -0600, Kristian Kielhofner wrote:

  
  
Michael Graves wrote:



  They look cute, but not enough RAM for *. Someone already has * ported
to the Soekris 4801. Have a look at www.soekris.com.

Michael
  

I didn't have to "port" Asterisk, the Soekris boards have 586's on them, 
I just compiled Asterisk accordingly and copied the binaries.  Easy.  As 
far as RAM, I am using Asterisk on a PC Engines WRAP with "only" 64mb of 
RAM, and it works fine.  You can remove modules, tune some Makefile vars 
to make * run smaller (I didn't have to for 64mb), and that's without 
tweaking the code.  I would love to see someone get * to run on a 
gumstix.  Why I don't know, but how does "coolness" sound?


  
  
My appologies. I thought the 64 MB was too small for actual production
use.

I'm hoping to change my * server to booting from an IDE flash module
over the holidays. I agree that the Soekris and Gumstix platforms are
cool, but I've already got a working server. I just want it to behave
more like an appliance.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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Re: [Asterisk-Users] Problem ringing simultaneous channels

FXS ports are always answered immediately.
Do you have to dial out over PTSN?
If so you are going to have an interesting time breaking this problem.
--
Christopher Dobbs
Russell Horn wrote:
Alexander,
I'm afraid it's POTS (X101P) and from what I have seen since I posted
this is my problem.
I wouldn't mind it hanging up the IAX2 channel and then calling it
again, but it seems that everytime the new call via Zap/2 means no
other calls are possible.
There is ISDN in the office, but I don't have any access until April
:/ If what I'm trying is impossible it will just have to wait
Russell.
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Re: [Asterisk-Users] polycom and cdp

Here is a list of the libs I am using
ld-2.2.5.so*
ld-linux.so.2@
libc-2.2.5.so*
libc.so.6@
libcom_err.so.2@
libcom_err.so.2.0*
libcrypt-2.2.5.so*
libcrypt.so.1@
libdl-2.2.5.so*
libdl.so.2@
libe2p.so.2@
libe2p.so.2.3*
libext2fs.so.2@
libext2fs.so.2.4*
libm-2.2.5.so*
libm.so.6@
libncurses.so.5@
libncurses.so.5.2*
libnsl-2.2.5.so*
libnsl.so.1@
libnss_dns-2.2.5.so*
libnss_dns.so.2@
libnss_files-2.2.5.so*
libnss_files.so.2@
libproc.so.2.0.7*
libresolv-2.2.5.so*
libresolv.so.2@
libutil-2.2.5.so*
libutil.so.1@
[EMAIL PROTECTED]
libcrypto.so@
libcrypto.so.0@
libcrypto.so.0.9.6*
libss.so@
libssl.a
libssl.so@
libssl.so.0@
libssl.so.0.9.6*
libwrap.a
libwrap.so.0@
libwrap.so.0.7.6*
libz.so@
libz.so.1@
libz.so.1.1.4*
libcrypto.a
libcrypto.so@
libcrypto.so.0@
libcrypto.so.0.9.6*
libss.so@
libssl.a
libssl.so@
libssl.so.0@
libssl.so.0.9.6*
libwrap.a
libwrap.so.0@
libwrap.so.0.7.6*
libz.so@
libz.so.1@
libz.so.1.1.4*
--
Christopher Dobbs
Richard wrote:
Hi,
Has anyone tried to use cdp to push the voice vlan tag to polycom phones?
The document says that it is supported, but I can't make it work.
Thanks,
Richard
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Re: [Asterisk-Users] polycom and cdp

Sorry, Replied to wrong message:)
--
Christopher Dobbs
Christopher Dobbs wrote:
Here is a list of the libs I am using
ld-2.2.5.so*
ld-linux.so.2@
libc-2.2.5.so*
libc.so.6@
libcom_err.so.2@
libcom_err.so.2.0*
libcrypt-2.2.5.so*
libcrypt.so.1@
libdl-2.2.5.so*
libdl.so.2@
libe2p.so.2@
libe2p.so.2.3*
libext2fs.so.2@
libext2fs.so.2.4*
libm-2.2.5.so*
libm.so.6@
libncurses.so.5@
libncurses.so.5.2*
libnsl-2.2.5.so*
libnsl.so.1@
libnss_dns-2.2.5.so*
libnss_dns.so.2@
libnss_files-2.2.5.so*
libnss_files.so.2@
libproc.so.2.0.7*
libresolv-2.2.5.so*
libresolv.so.2@
libutil-2.2.5.so*
libutil.so.1@
[EMAIL PROTECTED]
libcrypto.so@
libcrypto.so.0@
libcrypto.so.0.9.6*
libss.so@
libssl.a
libssl.so@
libssl.so.0@
libssl.so.0.9.6*
libwrap.a
libwrap.so.0@
libwrap.so.0.7.6*
libz.so@
libz.so.1@
libz.so.1.1.4*
libcrypto.a
libcrypto.so@
libcrypto.so.0@
libcrypto.so.0.9.6*
libss.so@
libssl.a
libssl.so@
libssl.so.0@
libssl.so.0.9.6*
libwrap.a
libwrap.so.0@
libwrap.so.0.7.6*
libz.so@
libz.so.1@
libz.so.1.1.4*
--
Christopher Dobbs
Richard wrote:
Hi,
Has anyone tried to use cdp to push the voice vlan tag to polycom 
phones?
The document says that it is supported, but I can't make it work.

Thanks,
Richard
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Re: [Asterisk-Users] Why does * only work with an ancient mpg123?

2004-12-20 Thread Christopher Dobbs

I have:
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59s-mh4 (2000/Oct/27). Written and copyrights by Michael Hipp.
It works verry Well.
--
Christopher Dobbs
Remco Barende wrote:
On Mon, 20 Dec 2004, Eric Wieling aka ManxPower wrote:
Remco Barende wrote:
Hi list!
Just wondering, why is * sticking with an mpg123 version from the 
stoneage?

Gentoo comes with 0.59s-r8 and this version doesn't even start.
Ik know I could forcibly unmerge mpg123 and install the old version 
but I guess some day newer versions will have to be supported?

Asterisk sets the following mpg123 options:
mpg123 -q -s --mono -r 8000 -b 2048 -f 4096
-q, --quiet  Quiet.  Suppress diagnostic messages.
-s, --stdout  The decoded  audio  samples  are  written  to  standard 
output, instead  of  playing them through the audio device.  This option
must be used if your audio hardware is not supported by  mpg123.
The  output format is raw (headerless) linear PCM audio data, 16
bit, stereo, host byte order.

-r rate, --rate rate  Set sample rate (default: automatic).  You may 
want  to change this  if  you  need  a  constant  bitrate independed 
of the mpeg stream rate. mpg123 automagically converts the rate. You 
should then combine this with --stereo or --mono.

-b size, --buffer size  Use an audio output buffer of size Kbytes.  
This  is useful  to bypass  short periods of heavy system activity, 
which would normally cause the audio output  to  be  interrupted.   
You should specify  a buffer size of at least 1024 (i.e. 1 Mb, which 
equals about 6 seconds of audio data) or more; less than about 300 
does not make  much  sense.  The default is 0, which turns buffering 
off.

-f factor, --scale factor  Change scale factor (default: 32768).
Pretty much any program that accepts these options to generate raw 
(headerless) linear PCM audio data, 16 bit, mono, host byte order, at 
8khz to stdout will work.  At this time the only one that does this 
that I know is mpg123 0.59r

Thanks! But when I look at the output of mpg123 0.59s-r8 all these 
commandline switches are still supported, why it it only the old 
version that is supported, not the newer ones?

asterisk # mpg123
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59s-r8 (2000/Oct/27). Written and copyrights by Michael Hipp.
Uses code from various people. See 'README' for more!
THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!
usage: mpg123 [option(s)] [file(s) | URL(s) | -]
supported options [defaults in brackets]:
   -vincrease verbosity level   -qquiet (don't print title)
   -ttestmode (no output)   -swrite to stdout
   -w filename write Output as WAV file
   -k n  skip first n frames [0]-n n  decode only n frames [all]
   -ccheck range violations -yDISABLE resync on errors
   -b n  output buffer: n Kbytes [0]-f n  change scalefactor [32768]
   -r n  set/force samplerate [auto]-g n  set audio hardware 
output gain
   -os,-ol,-oh  output to built-in speaker,line-out connector,headphones
-a d  set audio device
   -2downsample 1:2 (22 kHz)-4downsample 1:4 (11 kHz)
   -d n  play every n'th frame only -h n  play every frame n times
   -0decode channel 0 (left) only   -1decode channel 1 (right) 
only
   -mmix both channels (mono)   -p p  use HTTP proxy p 
[$HTTP_PROXY]
   -@ f  read filenames/URLs from f
   -zshuffle play (with wildcards)  -Zrandom play
   -u a  HTTP authentication string -E f  Equalizer, data from file
See the manpage mpg123(1) or call mpg123 with --longhelp for more 
information.

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[Asterisk-Users] Asterisk Startup Scripts (My Bad)

2004-12-20 Thread Christopher Dobbs

Sorry to all.
I sent the wrong shell scripts.
Here are the correct ones.
Again, sorry to all!
--
Christopher Dobbs
Shahed wrote:
Hi Christopher,
The files you posted for inittab have a reference
to astmain.
I cant find this file anywhere (including WiKi / google etc).
Is this a custom script that you have written ?
Regards
Shahed

#! /bin/bash

chvt 24
asterisk 
-vvc
  /dev/tty24  /dev/tty24
sleep 5

#! /bin/bash

asterisk -r

ast:12345:respawn:/sbin/astmain

pbx:x:0:0::/root:/sbin/astrun
#! /bin/bash

exec ssh [EMAIL PROTECTED]
#! /bin/bash

exec ssh [EMAIL PROTECTED]
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Re: [Asterisk-Users] A few simple (I hope) questions from a first-timer

2004-12-20 Thread Christopher Dobbs

There is a better way, have you looked at RxFAX and TxFAX applications.
I my,self have not used them, but there is plenty of info in the WiKi.
Asterisk will branch to a fax extension after an answer() if you have it 
set up that way.

--
Christopher Dobbs
Joel Moore wrote:
We have the following situation:
We're ordering two PSTN lines for our new office (no broadband at all --
it's not even available).  The first line is going to be our primary # and
it will also serve as our fax line using distinctive ring.
The second line is going to be for a dial-up ISP (*sigh*) but now we'd like
it to do double duty as the second line of a hunt group (we can't even get
voicemail service for these lines from Verizon).  We'd be willing to hamper
our Internet access a little to make this work.
We'll be ordering a 2 FXO, 2 FXS TDM400P card for this setup.
Now the questions:
1) I want to make sure I understand Asterisk's support for distinctive ring
properly.  I assume I can branch the first line into our fax machine before
I plug it into the * server and that * can be configured to ignore the fax
line's distinctive ring.  Correct?
Is there a better way to do this?  Would we be better off letting * handle
faxes?  Is there anyway to pair up * with some fax software and let the * PC
be the fax machine (thereby enjoying some additional benefits such as
routing faxes to an email server)?
2) Is there any way to use * with the data line (i.e. does it have some sort
of PPP module?) or do I have to branch that line into a modem before
plugging it into the TDM400P card?
3) Does call waiting work with *?  I see some messages saying that call
waiting is a wasted feature but the Asterisk web site mentions call waiting
is supported.  Despite have 2 lines on the hunt group we'd like to also have
call waiting so we can answer incoming calls when boths lines are tied up.
Also, An ideal situation would be where the data line gets put on hold
(something similar to the ICN feature found on USR modems) whenever the
second line is needed for voice (incoming or outgoing).  This may be too
complicated, though.
Joel Moore
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Re: [Asterisk-Users] Disabling ! command

I have asterisk auto-magicly start from inittab.
I have yest to try something that can do anything to the server (other 
than stop asterisk) from the * CLI.

If anyone would like a copy of the scripts i use, contact me off-list.
BTW: I had already removed the ! command from * before using it this way.
--
Christopher Dobbs
Roy Sigurd Karlsbakk wrote:
since I run asterisk as root with a CLI open on TTY12 I was wondering
if the ! (shell) command can be disabled from the config, for safety
reasons it seems me usefully.

well. IMHO if someone can get access to your asterisk console, they 
can always ctrl+z or shutdown now or something. secure your server. 
don't trust asterisk to do it

roy
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Re: [Asterisk-Users] Disabling ! command

I decided, since there was interest, to send them to the who list.
Here they are.
BTW: I am also sending an excerpt from my passwd and inittab files.
the passwd entry is so that i can ssh into [EMAIL PROTECTED] and get an 
asterisk CLI.
This Feature is why I have ! disabled.

I hope that this helps people out.
I would strongly suggest this only if:
   1) The ! command is disabled int the CLI.
   2) Telnet is COMPLETELY disabled!!!
   3) SSH access is only by the use of RSA keys.
This is the setup that we use.
It works very good for us.
--
Christopher Dobbs
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Re: [Asterisk-Users] Disabling ! command

-- Sorry, Forgot to attach the files!! :)
I decided, since there was interest, to send them to the who list.
Here they are.
BTW: I am also sending an excerpt from my passwd and inittab files.
the passwd entry is so that i can ssh into [EMAIL PROTECTED] and get an
asterisk CLI.
This Feature is why I have ! disabled.
I hope that this helps people out.
I would strongly suggest this only if:
   1) The ! command is disabled int the CLI.
   2) Telnet is COMPLETELY disabled!!!
   3) SSH access is only by the use of RSA keys.
This is the setup that we use.
It works very good for us.
--
Christopher Dobbs

ast:12345:respawn:/sbin/astmain

pbx:x:0:0::/root:/sbin/astrun
#! /bin/bash

exec ssh [EMAIL PROTECTED]
#! /bin/bash

exec ssh [EMAIL PROTECTED]
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Re: [Asterisk-Users] Q about IAX (and IAXy)

As Per The WiKi:
IAX sends both command information and voice data over the same connection.
This allows it th transverse a NAT seamlessly.
As for Double NAT, My setup is:
Home PBX [Wireless] ISP WiFi NAT [Ethernet] Primary 
NAT [Ethernet] Work PBX

So, Yes it will work over double NAT, I can send and receive calls at home.
In fact, I call my wife all the time from work.
--
Christopher Dobbs
Nabeel Jafferali wrote:
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is NAT-transperant. I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port 5060) with whatever command. The SIP
server respends to your device's apparent IP and port (this is decided
depending on how that NAT is set up, STUN, etc.). The voice is then sent
to the apparent RTP port on your device (deciding what that is, again,
would depend on the NAT set up).
How does IAX eliminate this problem of ports being mapped by your NAT
router and external IPs? Does it use one port for both commands and
voice packets? Does the remote server just use the received from IP
address and port to respond?
Finally, would an IAXy work seamlessly in a configuration where it is
plugged into a NAT router which is plugged into another NAT router  -
double NATted? The * server is on a public IP.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
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Re: [Asterisk-Users] voicemail without prompt





The way I have it set up, is that the mailbox is the same as the exten.
I then wrote a macro that does it for me.

[macro-stdiax]
; ARG1 = User
; ARG2 = Voice Mail Number
exten = s,1,Dial(IAX2/${ARG1}/[EMAIL PROTECTED]||Ttr)
;exten = s,2,Voicemail(u${ARG2})
;exten = s,3,Hangup
;exten = s,102,Voicemail(b${ARG2})
;exten = s,103,Hangup

macro-eracewcustomer]
; ARG1 = User
exten = s,1,Dial(IAX2/${ARG1}/[EMAIL PROTECTED]|20|Ttr)
exten = s,2,Voicemail(u${ARG1})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG1})
exten = s,103,Hangup

[macro-stdtrunk]
; ARG1 = Zap Port
; ARG2 = Voice Mail Number
exten = s,1,Dial(Zap/${ARG1}|20|Ttr)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup

[macro-dialtrunk]
exten = s,1,Dial(${TRUNK}/${ARG1})
exten = s,2,Congestion

[macro-dialiax]
exten = s,1,Dial(IAX/${ARG1}/${ARG2})
exten = s,2,Congestion



[dialing_context]

exten = 2050,1,Macro(stdtrunk,1,${EXTEN})
exten = 0205,1,Macro(stdiax,${EXTEN},${EXTEN})


--
Christopher Dobbs
Antony Stone wrote:

  On Friday 17 December 2004 21:25, Ross Kevlin wrote:

  
  
this would still only work if the mailbox number was the same as the caller
id. I need some way to get the actual mailbox number of the caller.

  
  
Where / how are your mailbox numbers stored?

It shouldn't be too difficult to create a script or DB request to provide the 
CID and get the mailbox number in response?

Just out of interest, why don't you make the mailbox ID = caller ID?

Antony.

  




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Re: [Asterisk-Users] Call on hold disconnects...





Are you having the phone place the person on hold, or are you having *
place them on hold?
I dial #700 and it puts them on hold and they stay there,
it also reads off to me the number I dial to get them off hold.

REF: /etc/asterisk/features.conf

--
Christopher Dobbs

Shoval Tomer wrote:

  That's both true and false.

We have a legacy PBX here. Panasonic make.
Analog extensions connected to it (a.k.a "stupid" extensions) behace exactly like the grandstream - you can put a call on hold, but if you put the handset back on the cradle it's bye bye Mary.

Digital extensions (a.k.a "smart" extensions) can hold a call indefinitely.
They can do other neat stuff too...

 

  
  
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTECTED]]
Sent: Friday, December 17, 2004 11:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Call on hold disconnects...

Antony,
Thanks. It seems that the GS will not keep the call on hold.
In the real world though, when you place a call on hold, it is held until
further action.
The caller will hear messages, music, anything while you are gone to look
for a file, etc.

Technically, if you place the call on hold and put the handset back on the
cradle, you DID NOT HANG UP to end the call.
If you want to hang up the call you will first have to take the call off
hold... No.

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]] On Behalf Of Antony Stone
Sent: Friday, December 17, 2004 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call on hold disconnects...


On Friday 17 December 2004 20:43, Ferguson, Michael wrote:



  OK. I guess I was not clear. Sorry.

The phone rings.
The person picks up the handset and speaks to the caller.
He then puts the call on hold by pressing the "HOLD" button on the GS
100 phone. The caller hears music on hold.
  

So far, so good.



  The hand set is placed back on the cradle (as is done on a regular
phone with a hold button)
  

I'm not sure I agree with this.   Some phones may allow you to hang up and
not
disconnect the call, but I don't think it's universal.   Some phones
interpret this to mean "oh, you want to hang up? Okay - I'll hang up the
call
then."



  The call is disconnected.
  

Well, yes, because you hung up.

What happens if you do something else, like dial another extension, or
press
the hold button again (perhaps to retreive the original caller)?

I repeat one of my original questions - if this is not what you expected
to
happen when you hang up the phone, how would you expect to hang up the
call
when you wanted to?

Antony.

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Re: [Asterisk-Users] Asterisk Hardware

What codec is your soft phone using?
Some of the codecs stink, also is the link to the * server heavily used?
--
Christopher Dobbs
Nihal wrote:
Does some hardware just not work very well with Asterisk?
I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram.
While listening to the demo over a softphone (over the LAN) I get a number of 
crackles and skips.
IS THIS NORMAL FOR ASTERISK?
Or is it hardware related?
Thanks,
Nihal
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Re: [Asterisk-Users] New Asterisk Prompts





Thank you!!

--
Christopher Dobbs

Brian Wilkins wrote:

  All,  
   Enjoy these free prompts as an addition to your sounds collection. I hope 
you find them useful. You can find them attached to this message.


  
  

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Re: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOCNear You!)

2004-12-15 Thread Christopher Dobbs





We have discussed T1/E1 Modules for our channel banks, but, our main
focus at this time is FXO/FXS interfaces.

A T1 card is about $440 US. An Ethernet card is about $15 US.
That is why we are doing this.

Further more, we believe that we can sell these units cheaper than a
T1/E1 channel bank,
increasing the savings to the end user.

(P.S. If Mr Spencer would like to comment on what Digium would think of
these units, I
 Would appreciate it.)

--
Christopher Dobbs

Matthew Boehm wrote:

  And T1s too.  If you can supply a seperate piece of hardware that can handle
all the T1 crap then pass calls to asterisk as SIP/IAX, that would be
awesome in our situation.

Right now our only solution is 8 T1s into a 5300, then SIP to asterisk.

-Matthew
- Original Message - 
From: "Marc Storck" [EMAIL PROTECTED]
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 8:09 PM
Subject: Re: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a
NOCNear You!)


  
  
Will you have a Channel Bank for E1, many E1s instead of the FXS/FXO

  
  ports??
  
  
    Marc

Christopher Dobbs wrote:


  My company has started development on a Ethernet based channel bank.

Here are the (current) spec's
   - 10/100 Ethernet Port
   - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired)
   - Serial Console
   - TDMoE
   - IAX2
   - EETP (A protocol that we have designed for IP Telephony)

We have just started prototyping this device, so...

-- 
Christopher Dobbs

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[Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC Near You!)

2004-12-13 Thread Christopher Dobbs

My company has started development on a Ethernet based channel bank.
Here are the (current) spec's
   - 10/100 Ethernet Port
   - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired)
   - Serial Console
   - TDMoE
   - IAX2
   - EETP (A protocol that we have designed for IP Telephony)
We have just started prototyping this device, so...
--
Christopher Dobbs
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Re: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC Near You!)

2004-12-13 Thread Christopher Dobbs

Had not thought of that.
We are mostly interested in providing POTS lines inside of a company.
Think of it as a a massive CPE device running on the VPN of a 
corporation with offices in many towns.

who knows?
We saw all of the traffic about Ethernet channel banks and started 
talking, and well, we are going to try this one.

--
Christopher Dobbs
Marc Storck wrote:
Will you have a Channel Bank for E1, many E1s instead of the FXS/FXO 
ports??

Marc
Christopher Dobbs wrote:
My company has started development on a Ethernet based channel bank.
Here are the (current) spec's
   - 10/100 Ethernet Port
   - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired)
   - Serial Console
   - TDMoE
   - IAX2
   - EETP (A protocol that we have designed for IP Telephony)
We have just started prototyping this device, so...
--
Christopher Dobbs
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Re: [Asterisk-Users] Voice Prompt Info

2004-12-11 Thread Christopher Dobbs

Your previous messages came through, but had [Asterisk-Users] Re: 
Asterisk-Users Digest, Vol 5, Issue 158 as the subject.

I for one usually skip messages where the person did not think to change 
the digest subject to something more meaningfully.

To help others help you could those of you who get digest form please 
fix the subject before replying?

Thank you in advance.
--
Christopher Dobbs
[EMAIL PROTECTED] wrote:
I have sent this twice now but, I think, for some reason, it has been 
sent as HTML which is causing it to be drooped (and rightly so).  I 
apologize in advance if, suddenly, those two make it though along with 
this one.
Anyway, I should have been more clear in my original message. I am 
looking for departments that fit - into - those strings. Pretty much, 
if a person could replace DEPT with what they are thinking,  they are 
on track.  I mention the strings them selves only as a way to show 
context.  When I first posted that message I had a handful of examples 
that did not fit into that 'mold' but, for the life of me, I can not 
think of one now.

Thanks;
James

Date:
Fri, 10 Dec 2004 16:24:00 -0800
You should not put the press or the number in the prompt.
Have them as separate sounds, that way, they are more generic.
[EMAIL PROTECTED] wrote:
I am looking for titles that fit into the string:
press 1 for the DEPT department or  press 1 for DEPT
but if you have other suggestions, let me know.
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Re: [Asterisk-Users] pc





I know that this has been replied to, but, where I work, We use the
Carrier Access Group Adit 600 Channel banks.
It supports two T1's and my boss tells me that we can get the for about
400-800 dollars.

They use 8 port c\expansion cards, so you can three groups of ports,
say 16 FXS (2x FXS) and 8 FXO ports (1x FXO).

These units are programmed using a crossover cable and a serial port.

They have a lot of other features that I don't know how to use, but
they work grate for us.

I hope this helps.

--
Christopher Dobbs

Shoval Tomer wrote:

  Can you recommend a channel bank make and model that will support all
(or most) of Asterisk's features and can be installed by a newbie?

I'll search for it on ebay


  
  
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, December 08, 2004 7:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] pc

On Wed, 2004-12-08 at 19:09 +0200, Shoval Tomer wrote:


  
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, December 08, 2004 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] pc

On Wed, 2004-12-08 at 18:07 +0200, Shoval Tomer wrote:


  I'm going to install asterisk with four digium cards.
Can anyone mention a brand that carries boards with 4 compatible
  

  

  
  pci
  
  

  

  slots?
  

What 4 cards are you thinking of installing? Most people seem to

  

  
  run
  
  

  
into trouble after the second concurrent card.

  
  
I was thinking one 4 port FXO card
Two 4 port FXS card
And the fourth with one FXO and three FXs
For a total of 11 FXSs and 5 FXOs (for five POTS lines and 11 analog
extensions).
  

You would be far better off going with a channel bank and T1 card.
Immediately you get room for expansion as you won't be limited to just
16 ports. As for cost, 4x $305(I think that is the current price) =
$1220, $1220 - $500(t100P) = $720, or enough left over to get taken on
parts from ebay a couple of times before you reach level costs.



  What trouble should I run into?
Define trouble? Hard to configure, or impossible because of IRQ
  

  
  sharing
  
  

  issues or whatever?
  

Trouble being 4 x 1000 interupts per second on the machine. Lack of
expansion, interupt sharing, any number of annoyances.



  
Anyways, almost all motherboards put 4 PCI slots on the board as

  

  
  it is
  
  

  
not much extra and expected on anything non entry level.

  
  I thought so. Apparently most of them only come with three.
I was hoping to locate a motherboard that has five so I'll have room
  

  
  to
  
  

  expand, but four will be great.

Anybody knows if PCI express technology is compatible with
  

  
  Wildcards?
  
  
--
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] four wildcards in a single pc





I have done multi-TDM4xx cards in a box.
DO NOT TRY THIS!!

Sorry for yelling but,
Our system worked, and I use that term loosely, for about a week.
The hardware simply cannot handle the onslaught of IRQ's that this
causes.
We got popping and beeping, and dead lines.

We put each card in a separate machine, and used IAX to link them.
All is working perfectly now.

I am going to try TDMOE next.
If you are interested in the results of that, contact me off list. (DO
NOT just send me a message saying just "Yes I am interested" I have
received several of those and the just get nuked 'cause I don't know
what the are interested in)

--
Christopher Dobbs

Shoval Tomer wrote:

  Hi.
Please excuse me asking this again. But it really puzzles me.

We're installing asterisk at a branch office at NJ (HQ is at
Petach-Tikva)
It'll need to support 5 POTS lines, 11 analog extensions and four VOIP
phones.

I wanted to go with a T1 card from digium and a channel bank, but we
have a dead line. It has to be up and running by January 1st.
I don't have the time to start shopping at ebay, where you don't know
what you'll get, and you need to install, under time pressure something
you not familiar with.

So I thought of installing a combination of four pci cards in the
machine, and everybody on the list just keeps telling me it won't work.

I have installed successfully more then four cards in a machine before.
I had a firewall with eight network interfaces (one quad card, one duo
and two singles)
I have machines with two dialogic boards, a pci display card, and a
network interface.
And I know I've had machines at home that had a display adaptor, modem,
network, scsi, and soundblaster all together.

Yet, people claim it won't work because of lack of IRQs, and that it's
not related to Digium.

What am I missing?


Shoval Tomer,
IT Manager,
SofTov Advanced Systems, Ltd.
Office: +972-3-9230686 ext. 179
Fax: +972-3-9216642
Mobile: +972-54-8000200


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Re: [Asterisk-Users] Ripping CD audio for MOH

Jean-Michel Hiver wrote:

What applications (osx or linux) are best?  Optimal settings?
 

linux 'grip' is very nice.
As is RipperX (http://ripperx.sourceforge.net/)
--
Christopher Dobbs
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Re: [Asterisk-Users] Voice Prompt Info

You should not put the press or the number in the prompt.
Have them as separate sounds, that way, they are more generic.
EG:
Background(press-1-for)
Background(sales)
Background(and)
Background(service)
Background(department)
Background(press-2-for)
Background(Tech)
Background(support)
--
Christopher Dobbs
[EMAIL PROTECTED] wrote:
I am trying to put together a list of 'departments' to request as 
voice prompts.  I have the biggies (sales, accounting, shipping, 
etc...) but I want to make sure I do not miss any. If anyone anyone 
has some suggestions (Ha... that is like going to an NRA meeting ans 
asking if anybody has a gun  :-)  ) please forward them to me (and / 
or post here although, with the volume of this list I do not always 
have time to read every digest so the 'and' option may be best.) so 
that I can compile a single list, verify that they are not already 
available, group them, and send them on.  Please put 'voice prompt' in 
the subject line of anything you forward me so that I am less likely 
to miss it.
I am looking for titles that fit into the string:
press 1 for the DEPT department or  press 1 for DEPT
but if you have other suggestions, let me know.
I will be collecting these for about a week so please try to get them 
to me in that time frame.
I am hopeful that, with these prompts, it will be possible to make a 
complete (albeit fairly generic) tree, all with the same voice.

Thanks;
James
alspachfam at charter dot net
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Re: [Asterisk-Users] Kind of off-topic: VoIP services and multiple callers

2004-12-07 Thread Christopher Dobbs

I disagree completely,
We provide VOIP services (in the final testing phases), and there is 
more interest in unlimited than in per minute rates.
We have run the stats and our unlimited plans are prices so that we get 
more than what the end user uses. (on average, that is)

--
Christopher Dobbs
nik martin wrote:
Andrew Kohlsmith wrote:
On December 6, 2004 10:12 pm, Michael Giagnocavo wrote:
Except the providers who offer unlimited -- in that case, they 
want you
to use as little as possible, so they can make their money.

They're the ones that are on the way to bankruptcy.
EXACTLY ;)
Aint no free lunch, my friends.
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Re: [Asterisk-Users] gsm codec, very poor quality.

2004-12-07 Thread Christopher Dobbs





I use sox to do wav to gsm conversion and have not had a problem with
sound quality.
Another trick i use, i have a menu that all it is for is recording the
menu prompts for the rest 
of the system. It was complicated to write, but, well worth it in
terms of creating the menu prompts I need.

Jon Radon wrote:

  Sorry this doesn't answer your question.  Any reason to not leave them as wav's?


On Tue, 7 Dec 2004 10:42:58 +0100, Matthew Oulton
[EMAIL PROTECTED] wrote:
  
  
Currently I am creating .wav files and then converting them via SOX to .au
file format, then running them through a gsm codec convertor which all works
fine except that it sounds like the recording was made with a sock in my
mouth !!
 
Could someone in * land help me to get a good sound quality with gsm format.
 
Thanks in advance.
 
 


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Re: [Asterisk-Users] Faxing..not 100%

2004-12-07 Thread Christopher Dobbs





I am trying to do exactly what you have done, can you send some
examples to me.
please reply to [EMAIL PROTECTED] so as I do not miss any reply you may
send.

--
Christopher Dobbs

Andrew Kohlsmith wrote:

  On December 7, 2004 01:06 pm, Matthew Boehm wrote:
  
  
POTS - PRI - Asterisk - ATA (Fax)

  
  
My setup:
POTS - PRI - Asterisk - Asterisk - TDM430P (Fax)

The asterisk-asterisk link is a dedicated 1024kbps SDSL link on dedicated 
ethernet cards (i.e. all other network traffic works is on the other network 
interfaces)

Our faxing is pretty solid.  Not *perfect* but certainly better than 50%.

Anything unusual about your PRI?  Do you hear crackles or "zapping" noises on 
it on voice calls?  Is your * box overloaded?  Anything funny with 
interrupts?  Using a RedHat kernel?

-A.
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[Asterisk-Users] Help with music over intercom.

2004-12-03 Thread Christopher Dobbs

I am using Console/DSP for an intercom.  I want to play my MP3 
collection over it when no one is using it, like when they do in the 
supermarket.

Can anyone help me with this.
Any suggestions will be appreciated.
--
Christopher Dobbs
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Re: [Asterisk-Users] Changing Asterisk Voicemail Storage Location

2004-11-25 Thread Christopher Dobbs

If a patch is developed that will acomplish this division, I am
interested in it.
My company is planning on deplying a massive * network with a central
server providing VM.

This would make the VM server easyer to admin.

--
Christopher Dobbs

Adam Goryachev wrote:

On Thu, 2004-11-25 at 16:22, Java Rockx wrote:

Can anyone tell me how difficult it would be to change the way asterisk stores/retrieves user
messages as follows?

Currently mailboxes are in
/var/spool/asterisk/voicemail/{context}

But I need to store messages in a hash to limit the number of directories per context. All mailbox
extensions are the user's 10-digit phone number (aka, DID). The parts of a DID are as follows
So my hashing would look like this

/var/spool/asterisk/voicemail/{context}/{npa}/{nxx}/{line}

And in the {line} directory we would have the usual Asterisk files/directories for inbox, etc.

We're looking at a large number of mailboxes and this would give us a maximum of 1 mailboxes
per directory - which plays nice with the Linux file system.

You might look at alternative filesystem formats. "Linux file system" is
not any file system I've heard of. Most likely you are referring to the
filesystem that you get by default when you do an install and just click
next without understanding the option each step of the way.
Specifically, look at reiserfs, it is very good at handling directories
with large number of files, as frequantly seen in mail servers using
maildir format etc...

I'm not sure I understand all the details, but reiserfs should be
equivalent in speed to a DB at least, I've frequantly seen it
referred to in that way back when I used to subscribe to their mailing
list.

I suppose you might ask the question, is it faster to parse the mailbox
name in userspace and then look up the correct file, or let the kernel
parse the name, and find the file for you

Hope this helps you...

Regards,
Adam

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Re: [Asterisk-Users] Fw: Gift for Mark Spencer

2004-11-23 Thread Christopher Dobbs





You, bloody moron. Is not most email unsolicited.

I never asked you to send an email,
Your message is off topic,
Your getting rude,
therefor YOU are a spammer.

Now, in the name of all that is decent, drop this thread. Lets get
back to what we are all here for,
discussing Asterisk.

--
Christopher Dobbs
Systems Manager
Eracew Computer Services

Joe Greco wrote:

  
On Tue, 2004-11-23 at 19:06 -0600, Joe Greco wrote:
[..snip..]


  On the flip side, senders of spam should not expect recipients to go 
to much (or any) trouble on their behalf, especially given the current
spam environment on the 'net.  They - not hackerwaCker - blew the 
surprise by sending the message to recipients unknown.
  

I'll just summarise all you said into one conclusion which remains the
same as to what Steven said: hackerwanker is a moron.

:-)

  
  
No, that's not what I said.  If you want the short, brutal summary, it'd
be:  

The spammer who sent the message is the moron.

Really, there are all sorts of bizarre phishing schemes and other scams out
on the 'net.  If you go asking random people for donations, and cannot put
the request in the context of solid well-knowns, such as an organization or
individual who is clearly legitimate, then it looks quite possibly like a
scam of some sort, and posting it to the list isn't exactly unreasonable -
it's more like a "watch out for this scam" community service.

However, we also have to remember that even being a well-known wouldn't make
it right to send unsolicited bulk e-mail.

So.

It's unfortunate (for the people trying to organize the gift) that
hackerwacker sent an alert to the list.  It's not unusual, though.  As
service providers, many of us actively encourage customers to put a stop 
to abuses of the mail system such as chain letters and other scams by 
asking people to take active countermeasures.  I'd consider this to be 
an example of just such a countermeasure.

It seems fitting that spammers should not have their goals furthered by the
act of spamming.  It would seem that this is precisely what happened in
this case.

I'll further note that I did receive a copy of the spam in question.  While
I did not choose to complain to the relevant sites about it, or to post a
message to the mailing list, the Boulder Pledge is certainly applicable -
I will not be contributing towards a gift effort that spammed.

... JG
  




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[Asterisk-Users] Secure IAX Communications

2004-11-19 Thread Christopher Dobbs





I have been moitoring your discutions about Secure IAX.

I have been thinking about writing a wrapper for IAX that uses OppenSSL
to encrypt the IAX stream.
If enough prople are interested in this I will do it.

If you are interested, email me offlist at [EMAIL PROTECTED].
if I do this I will be submitting it to Digium for inclusion in
mainstream Asterisk.

But I will only spend the time to do this if there is enough interest.

(While I am at it, If some one knows anything about ADSI, I would like
to add support for ADSI to IAX.)

Christopher Dobbs
Software Engineer
Eracew Computer Services
http://www.eracew.net/
[EMAIL PROTECTED]



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[Asterisk-Users] The Apperiant Death of IAXtel

Given that IAXtel has not been responding for some time, I am willing to 
setup accounts for thoes who want to have that kind of functionallity.  
If you are interested, send me an email with your requested username and 
password, and i will send you your account information.

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Re: [Asterisk-Users] The Apperiant Death of IAXtel

Let me clerify, Send a username and password for use on my new IAX relay 
system.
It wont use real phone numbers, but it will work to link the /free 
world/ of IAX users.

Kevin Walsh wrote:
Christopher Dobbs [EMAIL PROTECTED] wrote:
 

Given that IAXtel has not been responding for some time, I am willing to
setup accounts for thoes who want to have that kind of functionallity.
If you are interested, send me an email with your requested username and
password, and i will send you your account information.
   

Given that IAXtel has not been responding for some time, what use is
a username and password?
 

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Re: [Asterisk-Users] The Apperiant Death of IAXtel

Send account request information to [EMAIL PROTECTED]
Christopher Dobbs wrote:
Let me clerify, Send a username and password for use on my new IAX 
relay system.
It wont use real phone numbers, but it will work to link the /free 
world/ of IAX users.

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Re: [Asterisk-Users] The Apperiant Death of IAXtel

What ever, Just trying to be a help to the system.
Duane wrote:
Christopher Dobbs wrote:
Let me clerify, Send a username and password for use on my new IAX 
relay system.
It wont use real phone numbers, but it will work to link the /free 
world/ of IAX users.

Why not just use www.e164.org via enum lookups then, it does let you 
use both real phone numbers (after verification) and non-real numbers...

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Re: [Asterisk-Users] The Apperiant Death of IAXtel