date:20080517

Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?

Regards
Bilal

Bilal,

I linked a store and product for you in the thread
already. A simple
google search will turn up hundred if not thousands of
suppliers.

Just google Sata to Molex power turns up a quarter
million hits.
Find a supplier you like, and purchase.

http://www.google.com/search?q=sata+to+molex+power+adaptorie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a

Thanks,
Steve Totaro

On Mon, May 5, 2008 at 7:59 AM, bilal ghayyad
[EMAIL PROTECTED]
wrote:
Dears;

Till that monment, I did not get my answer. What
shall
I do when the power supply in the PC does not have
a
NORMA power connector to use it for the DIGIUM
card?
The only available connectors are SATA power
connector.

So is there a convertor to convert from SATA to
NORMAL? Or what should I do?

Maybe I understood that I have to use extenal power
supply to supply electrical for the card? Am
correct?

Regards
Bilal
--

On May 4, 2008 08:40:10 pm Jay R. Ashworth wrote:
Customer's insistence. We didn't have a
choice,
really.
Nothing wrong with that, it just adds more
billable hours. :-)
As long as it does.

I don't know about you, but whenever a customer
wants
me to do work and
does
not want to follow my recommendations, I have the
paper trail copied
out in
triplicate just to cover my ass.

Sometimes they're right, but generally when they
ask
me to do something
they
are asking me because they are unable to do it
themselves, so I am
extra-cautious when they won't follow my advice.

-A.

Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now.

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Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?

Regards
Bilal

Bilal,

I linked a store and product for you in the thread
already. A simple
google search will turn up hundred if not thousands of
suppliers.

Just google Sata to Molex power turns up a quarter
million hits.
Find a supplier you like, and purchase.

http://www.google.com/search?q=sata+to+molex+power+adaptorie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a

Thanks,
Steve Totaro

On Mon, May 5, 2008 at 7:59 AM, bilal ghayyad
[EMAIL PROTECTED]
wrote:
Dears;

So is there a convertor to convert from SATA to
NORMAL? Or what should I do?

Maybe I understood that I have to use extenal power
supply to supply electrical for the card? Am
correct?

Regards
Bilal
--

On May 4, 2008 08:40:10 pm Jay R. Ashworth wrote:
Customer's insistence. We didn't have a
choice,
really.
Nothing wrong with that, it just adds more
billable hours. :-)
As long as it does.

I don't know about you, but whenever a customer
wants
me to do work and
does
not want to follow my recommendations, I have the
paper trail copied
out in
triplicate just to cover my ass.

Sometimes they're right, but generally when they
ask
me to do something
they
are asking me because they are unable to do it
themselves, so I am
extra-cautious when they won't follow my advice.

-A.

Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now.

http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

2008-05-17 Thread Gordon Henderson

On Sat, 17 May 2008, bilal ghayyad wrote:

 Well, why Digium is still using this kind of power
 connector while all new machines does not come with
 these types?

The new machines that I buy come with legacy power connectors. The flash 
IDE drives I buy need legacy power connectors, and since convertors are 
trivially avalable why is it an issue?

Gordon

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[asterisk-users] One way sound when Using Dial cmd without t option (SOLVED) Need explanation

2008-05-17 Thread Mohammad A. Navid

I'm implementing a simple calling card feature for testing purpose. I have a
DID number, when I called my DID number and enter the phone number to call,
Asterisk would dial the number for me but the sound was only one way.
After hours of struggling with the problem, I found out that I need to add
t to my dial options, this is the correct way of dialing out:

 - Dial(SIP/carrier/310555|20|t)

Now I need to know what was going on? Why with option t both parties can
hear each other, but without option t in dial cmd only one party could
hear?

Another interesting issue is, if I use Answer() command at the begining the
sound becomes one way even if I use t in options.


One more interesting thing, my carrier for calling out only accepts G7.29
where as my DID provider passes calls as ulaw. However, when using
voipjet.com (as secondary carrier) which carries out the calls as ulaw,
having the Dial cmd without t option works fine.

I'd appreciate if someone explain to me the following questions:
 1) Why when there is codec difference, Dial cmd needs the t option
 2) Why using Answer cmd causes problem in this case (all the cases, when
using same codec and also different codecs)
 3) Why with same codecs, Dial cmd does not need t option?

Thanks

Moe
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Re: [asterisk-users] Discover connected Zap lines

So no way to discover the status of FXO if a cable
pluged or not?

Regards
Bilal
-

2008/5/2 Tzafrir Cohen [EMAIL PROTECTED]:
 On Fri, May 02, 2008 at 09:06:01AM +0200, Vinz486
wrote:
   2008/4/30 Tzafrir Cohen
[EMAIL PROTECTED]:
On Wed, Apr 30, 2008 at 09:07:48PM +0200,
Vinz486
 wrote:
  
  


-
   [May  2 08:51:00] WARNING[5119]: chan_zap.c:6685
 handle_init_event:
   Detected alarm on channel 3: No Alarm
   [May  2 08:51:03] NOTICE[5119]: chan_zap.c:6678
 handle_init_event:
   Alarm cleared on channel 4
  


-

  This means that you should be able to see it in the
InAlarm:
 field
 in
  'zap show channel 3'


Ok. Made some experiments.

InAlarm field show 1 if cable unplugged *BUT* only if
in previouos
time cable was plugged.

In few words, at boot, InAlarm is 0, Cable plugged: 0,
Cable unplugged:
 1

If i use this field, after a boot without cable, my
software will
think that the cable is plugged.

I found another useful field: Hookstate (FXS only).

It tell me if the cable is plugged ever after a boot
without cable.

Hookstate (FXS only): Offhook  --Cable plugged

Hookstate (FXS only): Onhook  --Cable unplugged


I hope this can help other people (and make to think
at Zaptel
developing to insert a field exactly for this purpose,
eg: Cable:
plugged or Cable: unplugged).

Bye.

-- 
PicoStreamer - the real WEB live streaming software
vinz486.com





  

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Re: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 9:18 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
 On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote:
 On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
  It seems any constructive criticism offered, you take as an attack
  against Digium.  That is not a good attitude.

 I dunno, Steve; I wouldn't call Digium needs to 'man-up' constructive
 criticism, myself.  I'd call it an ad-hominem.

 Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
 wrong with that... unless you're an *employee*, and you're going out of
 your way to hide it.

 I'm been a member of this community far longer than I've worked for Digium,
 and even then, I form my own opinions and I call them as I see them.  If I
 can't say something because of insider knowledge, I know well enough to keep
 my mouth shut, but this is not one of those times.  And if there _is_
 something wrong with the way Digium is doing something, I also am more than
 happy to put up a big fuss until it's fixed.

 I'm probably a bit of a loose cannon, but they knew that when they hired
 me.  ;-)

 --
 Tilghman


Man up and post some common benchmarks.  It is easy to leave out
context and take one line to prove your point.  Politicians do it
every day as of late.

Let me contact Sangoma, I am sure they will do it.  In fact, they
wanted me to do head to head benchmarks against Digium products.  Is
Digium game because Sangoma is ready willing and able?

It is about the money, like it or not.  You are going to an Avaya type
licensing scheme, everything is charged per port.  The box is capable
of doing more but you turn it off until you get more money.  It's like
the Definity G3s I have worked with.  The box can do everything but
until you pony up, it is not activated.

Any SwitchVox sale I have tried to pitch dies quickly and this is even
involving Switchvox reps on a conference call.

How about if I don't want support and use my own hardware, then can do
I still have to pay to upgrade to SMB or whatever?  Follow the logic?
Anyways, the profit margin on appliances is way too low.  I might as
well sell 3Coms or NECs if I am selling boxes with per seat license
fees and have to hack the box to do any customization.

They are not being conservative, when all you do is put a CC and then
a button shows up to upgrade, this is the same hardware mind you

Guess I will stick to my DL 380s and (if a GUI is required) FreePBX or
Druid (if it tests out ok).

Thanks,
Steve Totaro

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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 8:51 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
 On Fri, May 16, 2008 at 9:18 PM, Tilghman Lesher
 [EMAIL PROTECTED] wrote:
 On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote:
 On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
  It seems any constructive criticism offered, you take as an attack
  against Digium.  That is not a good attitude.

 I dunno, Steve; I wouldn't call Digium needs to 'man-up' constructive
 criticism, myself.  I'd call it an ad-hominem.

 Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
 wrong with that... unless you're an *employee*, and you're going out of
 your way to hide it.

 I'm been a member of this community far longer than I've worked for Digium,
 and even then, I form my own opinions and I call them as I see them.  If I
 can't say something because of insider knowledge, I know well enough to keep
 my mouth shut, but this is not one of those times.  And if there _is_
 something wrong with the way Digium is doing something, I also am more than
 happy to put up a big fuss until it's fixed.

 I'm probably a bit of a loose cannon, but they knew that when they hired
 me.  ;-)

 --
 Tilghman


 Man up and post some common benchmarks.  It is easy to leave out
 context and take one line to prove your point.  Politicians do it
 every day as of late.

 Let me contact Sangoma, I am sure they will do it.  In fact, they
 wanted me to do head to head benchmarks against Digium products.  Is
 Digium game because Sangoma is ready willing and able?

 It is about the money, like it or not.  You are going to an Avaya type
 licensing scheme, everything is charged per port.  The box is capable
 of doing more but you turn it off until you get more money.  It's like
 the Definity G3s I have worked with.  The box can do everything but
 until you pony up, it is not activated.

 Any SwitchVox sale I have tried to pitch dies quickly and this is even
 involving Switchvox reps on a conference call.

 How about if I don't want support and use my own hardware, then can do
 I still have to pay to upgrade to SMB or whatever?  Follow the logic?
 Anyways, the profit margin on appliances is way too low.  I might as
 well sell 3Coms or NECs if I am selling boxes with per seat license
 fees and have to hack the box to do any customization.

 They are not being conservative, when all you do is put a CC and then
 a button shows up to upgrade, this is the same hardware mind you

 Guess I will stick to my DL 380s and (if a GUI is required) FreePBX or
 Druid (if it tests out ok).

 Thanks,
 Steve Totaro


Anyways, isn't Asterisk 1.2.x and FC6 EOL?

Thanks,
Steve Totaro

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Re: [asterisk-users] trixbox, sangoma a200, dell poweredge

2008-05-17 Thread David Cook

  Is there any reason why I should be experiencing such bad line
 quality on inbound calls from PSTN? Call quality is perfect when
 plugging in a regular analogue phone.

 Do you have other phone lines you can try the A200 with?  Have you
 asked Sangoma support?

Ditto on Sangoma support - they are excellent.

Do you have hardware echo cancellation on this board? (Is there a D at
the end of the model number?). Sangoma's hw echo cancellation is
outstanding, if you are hearing training then I assume you aren't
using it or the board doesn't have it.

Assuming the PC is doing other only nominal things, yes, this PC is
capable for what you are doing. (PC is NOT the VPN endpoint, don't have
Tomcat, SQL or a spam filtering on a mailserver running on this box,
etc.). Check that you are using IO-APIC and that everybody is getting
their own interrupts.

Do ifconfig w1g1.
w1g1  Link encap:Point-to-Point Protocol
  UP POINTOPOINT RUNNING NOARP  MTU:8  Metric:1
  RX packets:1243390644 errors:0 dropped:0 overruns:0 frame:0
  TX packets:1243390644 errors:0 dropped:0 overruns:0 carrier:0
  collisions:0 txqueuelen:100
  RX bytes:9947125152 (9.2 GiB)  TX bytes:9947125152 (9.2 GiB)
  Interrupt:233 Memory:c206-61fff
Check that errors/dropped/overruns are low if not 0. If they are not 0,
re-run ifconfig and check that the numbers don't increase (they can
have errors when the drivers first load and get synced up, but then
stabalize. Be concerned if the number is not 0 and is over 1000.

Based on your description of 2 fxo  1 fxs board, I think you actually
have an A400 (as the A200 only accepts 2 modules total however it may
still be reported as an A200 family - don't know, haven't used a 400
yet).

Do wanrouter hwprobe to find out your info - note HWEC=32 means HW echo
caneller, IRQ=233 (higher than 16) means you have IO-APIC activated.
---
| Wanpipe Hardware Probe Info |
---
1 . AFT-A200-SH : SLOT=2 : BUS=5 : IRQ=233 : CPU=A : PORT=PRI : HWEC=32
: V=11
Card Cnt: S508=0  S514X=0  S518=0  A101-2=0  A104=0  A300=0  A200=1 
A108=0

Armed with more info, Sangoma support (or us on the list) can help out
more.

dbc.

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Re: [asterisk-users] Discover connected Zap lines

On Sat, May 17, 2008 at 05:00:43AM -0700, bilal ghayyad wrote:
 So no way to discover the status of FXO if a cable
 pluged or not?

What specific card do you use?

What version of Zaptel?

Did you actually read my message you were responding to?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Anonymous statistics collection tool for Asterisk servers?

On Sat, May 03, 2008 at 12:19:03PM -0400, Dean Collins wrote:
 I think it would be great for someone to write a small 'anonymous
 collection module' that an Asterisk sys-admin could download and install
 on their asterisk server which uploaded the stats to a community website
 where the data was anonymous but still valuable for the community.
 
  
 
 Even if it just collected number of new installations globally this
 would be a huge help to people selling asterisk to their customers who
 continually ask I've heard about this Asterisk open source stuff but
 how many are there installed globally anyway?
 
 So this survey is to gauge the community's reaction to the development
 of an analytics tool like this. Please answer with your honest thoughts
 so we can gauge demand within the community for this tool.

It's not that there's no demand. It's just that I won't trust anybody
with those data. For instance, think of it as in what places can I find
servers volnurable to advisory AST-200x-xxx? 

One parallel to look at is Debian's popcon:

  http://popcon.debian.org/


It is also used by Ubuntu:

  
  http://popcon.ubuntu.com/

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root

On Fri, May 16, 2008 at 06:32:30PM -0500, James Sneeringer wrote:
 On Fri, May 16, 2008 at 3:04 AM, Lee, John (Sydney)
 [EMAIL PROTECTED] wrote:
  First of all, thanks Philipp, Alan, Tzafrir and James for your valuable
  comments.  I have listed below the exact list of commands to run for
  reinstalling asterisk 1.4.* as non-root on a Redhat / Fedora distro.
  Hope others can benefit.
 
  I have the following comments/questions though:
  1) #What is safe_asterisk used for actually?  I did not touch it in
  my modification because I don't know when is it triggered?
 
 The safe_asterisk script monitors the actual asterisk process, and if
 it dies for some reason, 

Not for some reason. For instyance, if asterisk decides to die the
script should not restart it. And if it got a SIGTERM? (e.g.: from init
on shutdown?)

 it restarts it and optionally notifies you.
 It's just a precaution. MySQL is often run under a script called
 mysqld_safe for the same reason.
 
  2) #I do not actually know whether we really need to modify
  /etc/asterisk/asterisk.conf?  Is this file read by asterisk at all?
  Seems like an important file name - asterisk.conf?
 
 It is read by asterisk, but whether you need to change any of the
 defaults really depends on your environment. Most of the options in it
 have equivalent command-line options, so you might want to use
 asterisk.conf instead of modifying the startup script (which could be
 overwritten the next time you upgrade).

Also note that asterisk.conf options override command-line options (and
not the other way around, as you might have learned to expect from most
other applications).

 
  4) There is an additional chmod to run for letting voicemail.conf to be
  written by group asterisk.
 
 What I found was that /etc/asterisk also needs to be writable by the
 asterisk user, because asterisk will unlink and recreate the file, so
 it needs to be able to write to the directory, not just the file. You
 can protect yourself a little bit by setting the sticky bit on
 /etc/asterisk, so even if asterisk goes nuts, it can't whack files it
 doesn't actually have write permissions on.
 
 chmod g+w /etc/asterisk/voicemail.conf
 chmod g+w,+t /etc/asterisk

Question: what does it take to move the voicemail file from
/etc/asterisk/voicemail.conf to /etc/asterisk/writble/voicemail.conf ?

Patch voicemail.conf and leave a compatibility symlink for the others?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Implementation of Video Conferencing using Asterisk

On Sat, May 17, 2008 at 6:08 AM, Kashif Naeem [EMAIL PROTECTED] wrote:
 Hello All

 Is it possible to implement and deploy Video Conferencing using Asterisk ?
 Has anyone done it before ?

 Regards,

 --
 Kashif Naeem
 Business Development Manager
 Hadi Telecom
 www.haditelecom.com

 Cell: +92 (0)345 4226006
 Office: +92 (0)42 5692766

 Email: [EMAIL PROTECTED]
 MSN: [EMAIL PROTECTED]
 Gmail: [EMAIL PROTECTED]
 Skype: kashif.naeem

 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.

Looks buggy http://www.voip-info.org/wiki/view/Asterisk+video

This looks more promising but looks can be deceiving...

http://www.voip-info.org/wiki/index.php?page=1videoConference+-+Open+source+web2.0+video+conferencing+software+for+Asterisk

Thanks,
Steve Totaro

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Re: [asterisk-users] Not hearing first prompts

2008-05-17 Thread Sherwood McGowan

Alan Lord wrote:
 Sherwood McGowan wrote:
 snip /
   
   
   
 Hrm...I have encountered this before and sometimes doing an explicit 
 Answer() then a Wait(2), then calling the service can help.

 Hope this is helpful

 Sherwood McGowan

 

 Bingo!

 Thanks a bunch. That sorted it.

 Al

   
Fantastic! Very glad I could help.

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[asterisk-users] sipbroker CLI

2008-05-17 Thread Adrian Marsh

Hi,
 
Can anyone confirm if calls placed via sipbroker have their NUM CLI
changed by sipbroker??
I'm testing between two asterisk servers in seperate locations. When I
place a call directly, the CLI is fine. When the call is placed via
sipbroker lookup, the NAME stays the same, but the NUM is recieved as
sipbroker.  I'm trying to figure out if its being set by the sending
Asterisk server SIP account, or if siproker themselves would mess with
the CLI.
 
Thanks,
 
Adrian
 
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[asterisk-users] Googles 411 services

2008-05-17 Thread Adrian Marsh

All,
 
Does anyone know of a SIP URI direct to googles  800-GOOG-411 service?
 
When I put calls via sipbroker, half the time the calls fail.  An enum
lookup shows 3 URIs listed, none of them seem to be google directly, and
I think 1 of them fails 100%, and the remaining one fails at other
random times.
 
Thanks,
 
Adrian
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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-17 Thread Mike Trest - On Travel

Al, Randy, (and others):

What Al calls one very weak area for Asterisk is IMHO a difference 
in market perceptions.
Asterisk is positioned for CPE - PBX - Appliance market which needs 
feature-rich appeal
and mass-market focus.

Using asterisk for large scale does not mean that I have used it as 
a large scale PBX.
Indeed, the FARM approach that we will be discussing on Friday 23rd 
is for very-large
scale deployments with a reduced-feature-set focus.

Simply put, this is not on Digium'a program for broad market 
push.  Rightfully, Digium is
expecting it's distribution channels to push it  CPE-PBX and mass 
market solutions.

So it is up to the thousands of Asterisk consultants to be aware of 
these techniques
and to serve the much smaller number of clients (mostly VoIP network 
operators)
who need to deploy very large scale networks.

Indeed,  I am now working on a design now that supports 100,000+ 
simultaneous participants
in an application specific deployment.   In this scale of telephony 
application, the issues of
IP bandwidth and PSTN carrier access points are much more difficult 
to manage than anything
related to the Asterisk platform.

If this is your interest, then drop in 
http://voipusersconference.org The context of the discussion
is NON-COMMERCIAL. I have no product or service for sale.  I am just 
discussing a different approach
to using Asterisk.

..mike..




At 09:42 AM 5/16/2008, randulo wrote:
http://voipusersconference.org

On Fri, May 16, 2008 at 1:59 PM, Al Baker [EMAIL PROTECTED] wrote:
  this is one very weak area for *. There is NO ANSWER.

Hi,

There have been a couple of threads on this subject this week, so I'd
remind everyone that next Friday's VoIP Users Conference is about
*large scale* asterisk:

After many requests, we finally have someone to talk on large scale
implementation of VoIP systems with asterisk. Using a farm of Asterisk
and Digium cards, tens Of Thousands of simultaneous calls can be made
and Mike Trest has offered to take it all apart for us to look inside.

More about Mike Trest: http://www.mike.trest.com/

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Re: [asterisk-users] asterisk virtualization on VMWARE SX infrastructure

2008-05-17 Thread Michiel van Baak

On 14:42, Sat 17 May 08, Steve Totaro wrote:
 On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote:
  Hi
 
  what about asterisk virtualization on VMWARE XS infrastructure?
 
  The system installed will manage a call center with 50 operator,
  queues, CDR logging on external database.
 
  the protocol used is SIP, probably with G711 codec.
 
  Virtualization of Asterisk i a risk regarding performance?
 
  Thanks to all
 
 I wouldn't do it.  Maybe in a lab but certainly not for a 50 seat call center.

I would ;)
We run asterisk under vmware in production and have no problem with it.
This is in a pure voip setup.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer aficionados are both called users?


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[asterisk-users] Discover connected Zap lines

Hi Cohen;

I am using TDM22 (2 fxo and 2 fxs) digium card.
I am using zaptel 1.4.10.1

I readed, but not sure if readed all, as alot of
messages were going and coming.

Can u help?

Regadrs
Bilal


--
On Sat, May 17, 2008 at 05:00:43AM -0700, bilal
ghayyad wrote:
 So no way to discover the status of FXO if a cable
 pluged or not?

What specific card do you use?

What version of Zaptel?

Did you actually read my message you were responding
to?

-- 
   Tzafrir Cohen
icq#16849755 
jabber:[EMAIL PROTECTED]
+972-50-7952406  
mailto:[EMAIL PROTECTED]
http://www.xorcom.com 
iax:[EMAIL PROTECTED]/tzafrir




  

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[asterisk-users] zaptel 1.4.10 doesn't build on debian etch epia itx system

2008-05-17 Thread Erik de Wild: Tripple-o

Today I have been  messing around  with updating my residential  
phonesystem (it was running a 1.0 version from years ago). I have  
downloaded the last source packages for zaptel-1.4.10.1and   
asterisk-1.4.19.2. Zaptel doesn't want to build. After a long time of  
making this is the output that stops it suddenly. Does it makes sense  
to try another lower version of Zaptel, do I miss a package or should  
I change a line in the Makefile like I had to do to build Asterisk  
(Proc=i586 instead of Proc=uname -m which result in i686. The  
updated box is now running without zaptel and it seems to work ok but  
I would like to add ztdummy for conferences. Any suggestion to solve  
this problem is very welcome.

Friendly regards,


Erik de Wild

output uname -a
Linux debian 2.6.18-6-486 #1 Sun Feb 10 22:06:33 UTC 2008 i686 GNU/Linux



#
gcc -g -O2 -I.  -g -fPIC -Wall -DBUILDING_TONEZONE- 
DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool  
zttool.o  -lnewt
#
Can't locate Config_heavy.pl in @INC (@INC contains: /etc/perl /usr/ 
local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/ 
share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/ 
site_perl .) at /usr/lib/perl/5.8/Config.pm line 65.
#
make[2]: Entering directory `/usr/src/asterisk/zaptel-1.4.10.1/kernel/ 
xpp/utils'
#
cc -I../.. -o print_modes -g -Wall  print_modes.c
#
./print_modes init_fxo_modes
#
for i in zt_registration xpp_sync lszaptel xpp_blink zapconf  
zaptel_hardware; do perl -I./zconf -c $i || exit 1; done
#
Can't locate File/Basename.pm in @INC (@INC contains: ./zconf /etc/ 
perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/ 
perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/ 
local/lib/site_perl .) at zt_registration line 11.
#
BEGIN failed--compilation aborted at zt_registration line 11.
#
make[2]: *** [perlcheck] Error 1
#
make[2]: Leaving directory `/usr/src/asterisk/zaptel-1.4.10.1/kernel/ 
xpp/utils'
#
make[1]: *** [utils-subdirs] Error 2
#
make[1]: Leaving directory `/usr/src/asterisk/zaptel-1.4.10.1'
#
make: *** [all] Error 2 

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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-17 Thread Mike Trest - On Travel

At 11:44 AM 5/16/2008, you wrote:
Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls
that a particular machine could handle, but from a support perspective, it
doesn't matter how many the machine could theoretically handle, it matters
how many it could handle in the particular installation in a supportable
configuration (those are all those pesky variables we've been talking about).

Absolutely!   Right On!   Tell it like it is!And many other 
cryptic encouragements.

With very large scale deployments, I have a set of numbers available 
in my head
that work well to predict how many machines will be needed for a 
particular application
but I wind up being surprised by non-predictable rate of arrival  issues.

Since most of my deployments are tied with Television and other promotional
support, a single reference by the on-screen (or on-radio) commentator, and the
phones are instantly flooded with thousands of new call setup 
requests.  Indeed,
one such incident in a NASCAR race with 13M viewers, produced 18,000 new calls
within two minutes.   The rate of arrival of new calls was dispersed 
to a farm of 60
Asterisk in three widely separated regions of the 
US.   However,  approximately
15,000 calls were actually dropped on the PSTN / SS7 network before 
ever reaching
three dispersed Asterisk farms.

Those farms were being fed inbound calls by a network of 
250+  Nortel switches with
millions of subscribers.   However, the Los Angeles area PSTN network 
access facility
had only 900 spare channels available in that two minute 
period.   Meanwhile, every asterisk
answered every call and joint the callers into appropriate conference 
groups until every
single available port was fully occupied. This illustrates that 
such issues of call capacity
exist completely apart from the Asterisk or whatever machine is used 
for implementation.

So everyone should not be surprised by it depends kinds of answers 
to the question
of concurrent call counts.  This application was so far off the 
typical product specifications
that nothing published by Digium or anyone else could anticipate 
those surprises that
come when you least expect.

..mike..




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[asterisk-users] Local loopback vs SIP/IAX2

2008-05-17 Thread Steve Finkelstein

Hi all,

Would anyone be able to point me in the right direction as far as the
pros/cons of using a local loopback with a T1 provider, or just
peering with a company using SIP/IAX2 or my small office asterisk
setup?  I've seen setups in both scenarios.  The only potential pro of
the T1 that I can think of is quality of voice and having an SLA with
a provider.  Otherwise, are the costs justified?

Thanks for your input.

/sf

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Re: [asterisk-users] Googles 411 services

2008-05-17 Thread Brian J. Murrell

On Sat, 2008-05-17 at 18:38 +0100, Adrian Marsh wrote:
 All,
  
 Does anyone know of a SIP URI direct to googles  800-GOOG-411 service?

Yeah, I suppose a direct SIP connection would be nice.
 

 An enum lookup shows 3 URIs listed, none of them seem to be google
 directly,

No, they are SIP-PSTN termination services.  I use them via an ENUM
lookup for all of my toll-free calling since my ITSP doesn't terminate
toll-free for me at no charge.

 and I think 1 of them fails 100%, and the remaining one fails at other
 random times.

Yeah, they do have a random failure rate, which is why my enum macro
returns all three and rolls over to alternate values if any fail.  Check
the archives (within the last few weeks) for more details.

b,




signature.asc
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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 3:11 PM, Mike Trest - On Travel [EMAIL PROTECTED] 
wrote:
 At 11:44 AM 5/16/2008, you wrote:
Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls
that a particular machine could handle, but from a support perspective, it
doesn't matter how many the machine could theoretically handle, it matters
how many it could handle in the particular installation in a supportable
configuration (those are all those pesky variables we've been talking about).

 Absolutely!   Right On!   Tell it like it is!And many other
 cryptic encouragements.

 With very large scale deployments, I have a set of numbers available
 in my head
 that work well to predict how many machines will be needed for a
 particular application
 but I wind up being surprised by non-predictable rate of arrival  issues.

 Since most of my deployments are tied with Television and other promotional
 support, a single reference by the on-screen (or on-radio) commentator, and 
 the
 phones are instantly flooded with thousands of new call setup
 requests.  Indeed,
 one such incident in a NASCAR race with 13M viewers, produced 18,000 new calls
 within two minutes.   The rate of arrival of new calls was dispersed
 to a farm of 60
 Asterisk in three widely separated regions of the
 US.   However,  approximately
 15,000 calls were actually dropped on the PSTN / SS7 network before
 ever reaching
 three dispersed Asterisk farms.

 Those farms were being fed inbound calls by a network of
 250+  Nortel switches with
 millions of subscribers.   However, the Los Angeles area PSTN network
 access facility
 had only 900 spare channels available in that two minute
 period.   Meanwhile, every asterisk
 answered every call and joint the callers into appropriate conference
 groups until every
 single available port was fully occupied. This illustrates that
 such issues of call capacity
 exist completely apart from the Asterisk or whatever machine is used
 for implementation.

 So everyone should not be surprised by it depends kinds of answers
 to the question
 of concurrent call counts.  This application was so far off the
 typical product specifications
 that nothing published by Digium or anyone else could anticipate
 those surprises that
 come when you least expect.

 ..mike..


I don't think anyone is expecting any rough numbers from Digium about
the telco's ingress/egress.

Thanks,
Steve Totaro

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[asterisk-users] Hangup issue

2008-05-17 Thread Cyril SCETBON

Hi guys,

My asterisk server is connected to a pstn gateway using SIP. When I 
receive a call and use the Hangup command the pstn seems to not 
correctly see the request and the caller gets a 'number unknown message.

Below are the debug message printed on the CLI :


 -- Executing [EMAIL PROTECTED]:3] 
Hangup(SIP/192.168.19.1-0818f100, ) in new stack
   == Spawn extension (accueil, 483062608, 3) exited non-zero on 
'SIP/192.168.19.1-0818f100'
Scheduling destruction of SIP dialog 
'[EMAIL PROTECTED]' in 384 ms (Method: ACK)
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for 
address/port to send to
set_destination: set destination to 192.168.19.1, port 5060
Reliably Transmitting (NAT) to 192.168.19.1:53728:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0

SIP/2.0 200 OK

-
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 
'[EMAIL PROTECTED]' Method: ACK

SIP/2.0 200 OK

Any idea about what's happening and how to resolve it ?

Regards
-- 
Cyril SCETBON


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Re: [asterisk-users] asterisk virtualization on VMWARE SX infrastructure

On Sat, May 17, 2008 at 2:56 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
 On 14:42, Sat 17 May 08, Steve Totaro wrote:
 On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote:
  Hi
 
  what about asterisk virtualization on VMWARE XS infrastructure?
 
  The system installed will manage a call center with 50 operator,
  queues, CDR logging on external database.
 
  the protocol used is SIP, probably with G711 codec.
 
  Virtualization of Asterisk i a risk regarding performance?
 
  Thanks to all

 I wouldn't do it.  Maybe in a lab but certainly not for a 50 seat call 
 center.

 I would ;)
 We run asterisk under vmware in production and have no problem with it.
 This is in a pure voip setup.
 --

 Michiel van Baak
 [EMAIL PROTECTED]
 http://michiel.vanbaak.eu
 GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

 Why is it drug addicts and computer aficionados are both called users?


Production = 50 seat call center?  What would an hour or two of
downtime cost you in your production setup?

Thanks,
Steve Totaro

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Re: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 8:37 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
 On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
 It seems any constructive criticism offered, you take as an attack
 against Digium.  That is not a good attitude.

 I dunno, Steve; I wouldn't call Digium needs to 'man-up' constructive
 criticism, myself.  I'd call it an ad-hominem.

 Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
 wrong with that... unless you're an *employee*, and you're going out of
 your way to hide it.

 Cheers,
 -- jra
 --
 Jay R. Ashworth   Baylink  [EMAIL 
 PROTECTED]
 Designer The Things I Think   RFC 2100
 Ashworth  Associates http://baylink.pitas.com '87 e24
 St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)


He is an employee and he does not post from a Digium account or
include that fact in his signature.  Not that it is to hide the fact,
but it certainly is obfuscated.

Thanks,
Steve Totaro

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Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

2008-05-17 Thread Matt Watson

On May 17, 2008 06:59:43 am Gordon Henderson wrote:
 On Sat, 17 May 2008, bilal ghayyad wrote:
  Well, why Digium is still using this kind of power
  connector while all new machines does not come with
  these types?

 The new machines that I buy come with legacy power connectors. The flash
 IDE drives I buy need legacy power connectors, and since convertors are
 trivially avalable why is it an issue?


molex power connectors are not legacy or old style except when used in 
reference to SATA devices.  Seeing as how Digium interface cards are not SATA 
devices, why would you expect them to be using a SATA connector?  Its not 
Digiums fault that the PSU you bought doesn't include molex connectors.

SATA uses a different power connector for a few reasons, but the biggest is 
that SATA supports hot-plugging (assuming your controller, drive, and OS 
support it), in order for hot-plug to work the drive needs a 3.3V voltage as 
well as  5V and 12V, molex only gives 5V and 12V.  The actual physical 
connector that molex uses also does not lend itself very well to 
hot-plugging.

SATA power connectors, while there is no real reason that non-SATA devices 
couldn't use them, they simply were designed for SATA specifically.  

Personally, I prefer molex connectors for most things simply because they are 
far more secure than SATA connectors (at least the ones i've used).


-- 
Matt
http://www.mattgwatson.ca

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Re: [asterisk-users] asterisk virtualization on VMWARE SX infrastructure

2008-05-17 Thread Michiel van Baak

On 16:18, Sat 17 May 08, Steve Totaro wrote:
 On Sat, May 17, 2008 at 2:56 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
  On 14:42, Sat 17 May 08, Steve Totaro wrote:
  On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote:
   Hi
  
   what about asterisk virtualization on VMWARE XS infrastructure?
  
   The system installed will manage a call center with 50 operator,
   queues, CDR logging on external database.
  
   the protocol used is SIP, probably with G711 codec.
  
   Virtualization of Asterisk i a risk regarding performance?
  
   Thanks to all
 
  I wouldn't do it.  Maybe in a lab but certainly not for a 50 seat call 
  center.
 
  I would ;)
  We run asterisk under vmware in production and have no problem with it.
  This is in a pure voip setup.
 
 Production = 50 seat call center?  What would an hour or two of
 downtime cost you in your production setup?

That would be: one of the many customers we host.
We have a hosted setup with over 100 companies, so an hour or two will
be a massive claim I'm sure.

At the moment we have 4 asterisk servers under vmware that act like one
freaking big and stable machine to the outside world.
I dont think this is because of vmware, could have done the same setup
with asterisk dedicated hardware but why bother when it works this way
as well ?

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer aficionados are both called users?


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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-17 Thread Michiel van Baak

On 16:20, Sat 17 May 08, Steve Totaro wrote:
 On Fri, May 16, 2008 at 8:37 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
  On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:
  It seems any constructive criticism offered, you take as an attack
  against Digium.  That is not a good attitude.
 
  I dunno, Steve; I wouldn't call Digium needs to 'man-up' constructive
  criticism, myself.  I'd call it an ad-hominem.
 
  Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
  wrong with that... unless you're an *employee*, and you're going out of
  your way to hide it.
 
  Cheers,
  -- jra
  --
  Jay R. Ashworth   Baylink  [EMAIL 
  PROTECTED]
  Designer The Things I Think   RFC 
  2100
  Ashworth  Associates http://baylink.pitas.com '87 
  e24
  St Petersburg FL USA  http://photo.imageinc.us +1 727 647 
  1274
 
  Those who cast the vote decide nothing.
  Those who count the vote decide everything.
-- (Joseph Stalin)
 
 
 He is an employee and he does not post from a Digium account or
 include that fact in his signature.  Not that it is to hide the fact,
 but it certainly is obfuscated.

I think it just shows that his opinions are his, and in no way are
linked to the 'digium opinion'


-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer aficionados are both called users?


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Re: [asterisk-users] asterisk virtualization on VMWARE SX infrastructure

On Sat, May 17, 2008 at 4:54 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
 On 16:18, Sat 17 May 08, Steve Totaro wrote:
 On Sat, May 17, 2008 at 2:56 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
  On 14:42, Sat 17 May 08, Steve Totaro wrote:
  On Sat, May 17, 2008 at 2:18 PM, nik600 [EMAIL PROTECTED] wrote:
   Hi
  
   what about asterisk virtualization on VMWARE XS infrastructure?
  
   The system installed will manage a call center with 50 operator,
   queues, CDR logging on external database.
  
   the protocol used is SIP, probably with G711 codec.
  
   Virtualization of Asterisk i a risk regarding performance?
  
   Thanks to all
 
  I wouldn't do it.  Maybe in a lab but certainly not for a 50 seat call 
  center.
 
  I would ;)
  We run asterisk under vmware in production and have no problem with it.
  This is in a pure voip setup.

 Production = 50 seat call center?  What would an hour or two of
 downtime cost you in your production setup?

 That would be: one of the many customers we host.
 We have a hosted setup with over 100 companies, so an hour or two will
 be a massive claim I'm sure.

 At the moment we have 4 asterisk servers under vmware that act like one
 freaking big and stable machine to the outside world.
 I dont think this is because of vmware, could have done the same setup
 with asterisk dedicated hardware but why bother when it works this way
 as well ?

 --

 Michiel van Baak
 [EMAIL PROTECTED]
 http://michiel.vanbaak.eu
 GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

 Why is it drug addicts and computer aficionados are both called users?


I suppose if it works, it works and that is all that matters.

But I would say why bother when it works this way as well? too about
the dedicated hardware.  ;-)  It just seems like another layer to
break.

Thanks,
Steve Totaro

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Re: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 08:18:46PM -0500, Tilghman Lesher wrote:
  Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
  wrong with that... unless you're an *employee*, and you're going out of
  your way to hide it.
 
 I'm been a member of this community far longer than I've worked for Digium,
 and even then, I form my own opinions and I call them as I see them.  If I
 can't say something because of insider knowledge, I know well enough to keep
 my mouth shut, but this is not one of those times.  And if there _is_
 something wrong with the way Digium is doing something, I also am more than
 happy to put up a big fuss until it's fixed.
 
 I'm probably a bit of a loose cannon, but they knew that when they hired
 me.  ;-)

That's about what I thought, and -- speaking for myself -- I'm fine
with that.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 08:51:04AM -0400, Steve Totaro wrote:
 It is about the money, like it or not.  You are going to an Avaya type
 licensing scheme, everything is charged per port.  The box is capable
 of doing more but you turn it off until you get more money.  It's like
 the Definity G3s I have worked with.  The box can do everything but
 until you pony up, it is not activated.

Yeah, and if you cut a jumper on a VAX11/780, it went twice as fast.

So *what*, Steve?  Are they not allowed to make money?

 Any SwitchVox sale I have tried to pitch dies quickly and this is even
 involving Switchvox reps on a conference call.

Ah.  Then if you don't like the way they do business, vote with your
wallet; don't sell their junk.

 How about if I don't want support and use my own hardware, then can do
 I still have to pay to upgrade to SMB or whatever?  Follow the logic?
 Anyways, the profit margin on appliances is way too low.  I might as
 well sell 3Coms or NECs if I am selling boxes with per seat license
 fees and have to hack the box to do any customization.

Yup.  :-)

 They are not being conservative, when all you do is put a CC and then
 a button shows up to upgrade, this is the same hardware mind you
 
 Guess I will stick to my DL 380s and (if a GUI is required) FreePBX or
 Druid (if it tests out ok).

Sounds like the best answer to me, for you.  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 09:06:15AM -0400, Steve Totaro wrote:
 Anyways, isn't Asterisk 1.2.x and FC6 EOL?

1.2 better not be EOL.  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 10:56:21PM +0200, Michiel van Baak wrote:
  He is an employee and he does not post from a Digium account or
  include that fact in his signature.  Not that it is to hide the fact,
  but it certainly is obfuscated.
 
 I think it just shows that his opinions are his, and in no way are
 linked to the 'digium opinion'

That's the common approach, yes.  If he says something that seems...
off, in that context, I'm sure we'll call him on it.

Cheers,
-- jr 'or his bosses will' a
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] zaptel 1.4.10 doesn't build on debian etch epia itx system

On Sat, May 17, 2008 at 09:01:26PM +0200, Erik de Wild: Tripple-o wrote:
 Today I have been  messing around  with updating my residential  
 phonesystem (it was running a 1.0 version from years ago). I have  
 downloaded the last source packages for zaptel-1.4.10.1and   
 asterisk-1.4.19.2. Zaptel doesn't want to build. After a long time of  
 making this is the output that stops it suddenly. Does it makes sense  
 to try another lower version of Zaptel, do I miss a package or should  
 I change a line in the Makefile like I had to do to build Asterisk  
 (Proc=i586 instead of Proc=uname -m which result in i686. The  
 updated box is now running without zaptel and it seems to work ok but  
 I would like to add ztdummy for conferences. Any suggestion to solve  
 this problem is very welcome.
 
 Friendly regards,
 
 
 Erik de Wild
 
 output uname -a
 Linux debian 2.6.18-6-486 #1 Sun Feb 10 22:06:33 UTC 2008 i686 GNU/Linux
 
 
 
 #
 gcc -g -O2 -I.  -g -fPIC -Wall -DBUILDING_TONEZONE- 
 DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool  
 zttool.o  -lnewt
 #
 Can't locate Config_heavy.pl in @INC (@INC contains: /etc/perl /usr/ 
 local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/ 
 share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/ 
 site_perl .) at /usr/lib/perl/5.8/Config.pm line 65.

That's strange.

 #
 make[2]: Entering directory `/usr/src/asterisk/zaptel-1.4.10.1/kernel/ 
 xpp/utils'
 #
 cc -I../.. -o print_modes -g -Wall  print_modes.c
 #
 ./print_modes init_fxo_modes
 #
 for i in zt_registration xpp_sync lszaptel xpp_blink zapconf  
 zaptel_hardware; do perl -I./zconf -c $i || exit 1; done
 #
 Can't locate File/Basename.pm in @INC (@INC contains: ./zconf /etc/ 
 perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/ 
 perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/ 
 local/lib/site_perl .) at zt_registration line 11.
 #
 BEGIN failed--compilation aborted at zt_registration line 11.

File::Basename is included in the package perl-modules .

 #
 make[2]: *** [perlcheck] Error 1

(perl-check is a target that runs a test compilation (perl -C) of a
number of perl scripts to make sure that they can run. In this case
zt_registration fails due to the lack of File::Basename

You could avoid xpp/utils altogether if you want to save a few MB-s .
But I think you'll e missing some nice utilities.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

On Sat, May 17, 2008 at 04:45:00PM -0400, Matt Watson wrote:
 well as  5V and 12V, molex only gives 5V and 12V.  The actual physical 
 connector that molex uses also does not lend itself very well to 
 hot-plugging.

IME, it doesn't lend itself very well to *cold*-plugging, either.

:-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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[asterisk-users] Using a Loopback Plug for an RJ-45 Ethernet Interface for testing a Digium Card

2008-05-17 Thread Jose P. Espinal

Hello,

Someone told me about using a Loopback plug for RJ-45 for testing if a 
Digium Card gave him 'green' Alarm (for testing if the card had been 
damaged by  a strange voltage surge); would this have some bad side effect?

Thanks in advice,

--
Jose P. Espinal

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Re: [asterisk-users] Using a Loopback Plug for an RJ-45 Ethernet Interface for testing a Digium Card

On Sat, May 17, 2008 at 05:50:46PM -0400, Jose P. Espinal wrote:
 Someone told me about using a Loopback plug for RJ-45 for testing if a 
 Digium Card gave him 'green' Alarm (for testing if the card had been 
 damaged by  a strange voltage surge); would this have some bad side effect?

Well, it's actually probably an RJ-48X interface, but who's counting.

If your question, though, is would plugging a loopback-wired 8p8c modular 
plug into the jack on the back of a T-1 card cause it any damage in
itself?, I can't imagine the answer would be yes, no.

If you loop it and it stays in Red alarm, it's probably broken.

I can't speak to exactly what the alarm status stuff does if the port
you're looping expects to have a PRI plugged into it: I would expect
Green, but no actual traffic, but I could be wrong, I'm a bit new on
that front.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Using a Loopback Plug for an RJ-45 EthernetInterface for testing a Digium Card

2008-05-17 Thread Alexander Lopez

It will go Green if a PROPER loopback plug is inserted.

Pins 1 and 2 shorted to 4 ad 5

Pin 1 to 4
Pin 2 to 5

Leave the others open...


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
 Sent: Saturday, May 17, 2008 6:02 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Using a Loopback Plug for an RJ-45
 EthernetInterface for testing a Digium Card
 
 On Sat, May 17, 2008 at 05:50:46PM -0400, Jose P. Espinal wrote:
  Someone told me about using a Loopback plug for RJ-45 for testing if
a
  Digium Card gave him 'green' Alarm (for testing if the card had been
  damaged by  a strange voltage surge); would this have some bad side
 effect?
 
 Well, it's actually probably an RJ-48X interface, but who's counting.
 
 If your question, though, is would plugging a loopback-wired 8p8c
modular
 plug into the jack on the back of a T-1 card cause it any damage in
 itself?, I can't imagine the answer would be yes, no.
 
 If you loop it and it stays in Red alarm, it's probably broken.
 
 I can't speak to exactly what the alarm status stuff does if the port
 you're looping expects to have a PRI plugged into it: I would expect
 Green, but no actual traffic, but I could be wrong, I'm a bit new on
 that front.
 
 Cheers,
 -- jra
 --
 Jay R. Ashworth   Baylink
 [EMAIL PROTECTED]
 Designer The Things I Think
RFC
 2100
 Ashworth  Associates http://baylink.pitas.com
'87
 e24
 St Petersburg FL USA  http://photo.imageinc.us +1 727
647
 1274
 
Those who cast the vote decide nothing.
Those who count the vote decide everything.
  -- (Joseph Stalin)
 
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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 5:36 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
 On Sat, May 17, 2008 at 09:06:15AM -0400, Steve Totaro wrote:
 Anyways, isn't Asterisk 1.2.x and FC6 EOL?

 1.2 better not be EOL.  :-)

 Cheers,
 -- jra
 --
 Jay R. Ashworth   Baylink  [EMAIL 
 PROTECTED]
 Designer The Things I Think   RFC 2100
 Ashworth  Associates http://baylink.pitas.com '87 e24
 St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)


End of life date for Asterisk 1.2 was August 1, 2007.

Thanks,
Steve Totaro

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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 5:36 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
 On Sat, May 17, 2008 at 08:51:04AM -0400, Steve Totaro wrote:
 It is about the money, like it or not.  You are going to an Avaya type
 licensing scheme, everything is charged per port.  The box is capable
 of doing more but you turn it off until you get more money.  It's like
 the Definity G3s I have worked with.  The box can do everything but
 until you pony up, it is not activated.

 Yeah, and if you cut a jumper on a VAX11/780, it went twice as fast.

 So *what*, Steve?  Are they not allowed to make money?

 Any SwitchVox sale I have tried to pitch dies quickly and this is even
 involving Switchvox reps on a conference call.

 Ah.  Then if you don't like the way they do business, vote with your
 wallet; don't sell their junk.

 How about if I don't want support and use my own hardware, then can do
 I still have to pay to upgrade to SMB or whatever?  Follow the logic?
 Anyways, the profit margin on appliances is way too low.  I might as
 well sell 3Coms or NECs if I am selling boxes with per seat license
 fees and have to hack the box to do any customization.

 Yup.  :-)

 They are not being conservative, when all you do is put a CC and then
 a button shows up to upgrade, this is the same hardware mind you

 Guess I will stick to my DL 380s and (if a GUI is required) FreePBX or
 Druid (if it tests out ok).

 Sounds like the best answer to me, for you.  :-)

 Cheers,
 -- jra
 --
 Jay R. Ashworth   Baylink  [EMAIL 
 PROTECTED]
 Designer The Things I Think   RFC 2100
 Ashworth  Associates http://baylink.pitas.com '87 e24
 St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)



Sure they are allowed to make money but don't lie and say it is not
about the money, It is about support.  Yeah supporting the company's
cash flow.

I can express my opinion and I did.  Maybe Digium will take notice,
maybe they won't.

Maybe next they will charge $250 for conference bridge capabilities.
 It's a joke to cripple things that can be enabled by flicking a
switch.  Your system comes with eight ports of VM but for another $250
we can give you 12..

Thanks,
Steve Totaro

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Re: [asterisk-users] Discover connected Zap lines

On Sat, May 17, 2008 at 11:56:47AM -0700, bilal ghayyad wrote:
 Hi Cohen;
 
 I am using TDM22 (2 fxo and 2 fxs) digium card.
 I am using zaptel 1.4.10.1
 
 I readed, but not sure if readed all, as alot of
 messages were going and coming.
 
 Can u help?

You should see (RED) in /proc/zaptel/1 for the channel if it is
disconnected. Not to mention that the channel will be in alarm (InAlarm
in zap show channel NNN).

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-17 Thread Tilghman Lesher

On Saturday 17 May 2008 17:43:51 Steve Totaro wrote:
 On Sat, May 17, 2008 at 5:36 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
  On Sat, May 17, 2008 at 08:51:04AM -0400, Steve Totaro wrote:
  It is about the money, like it or not.  You are going to an Avaya type
  licensing scheme, everything is charged per port.  The box is capable
  of doing more but you turn it off until you get more money.  It's like
  the Definity G3s I have worked with.  The box can do everything but
  until you pony up, it is not activated.
 
  Yeah, and if you cut a jumper on a VAX11/780, it went twice as fast.
 
  So *what*, Steve?  Are they not allowed to make money?

 Sure they are allowed to make money but don't lie and say it is not
 about the money, It is about support.  Yeah supporting the company's
 cash flow.

If that were all it was about, then you could call sales and get an infinite
number of licenses for a particular machine.  Go ahead and try it.  Call them.
There is a hard upper limit on the number of licenses they will sell for a
single machine, because any more is not supportable.

They may say, Let us call you back on that, because the next thing they're
going to do is consult with Engineering and Support and find out if that is
doable.  The maximum number of calls that you can buy for a single machine
is really about what is supportable (because if we sell more, and it doesn't
work, it's going to cost us in support time, on the phone, and possibly ending
up with a customer refund, because what (hypothetically) was sold was not
supportable).

Yes, the various tiers below that absolute limit is about money; it's about
charging based upon what we think it will cost us to support that number
of users, should something go wrong, and the customer needs to call in.  And
yes, there's a bit of profit margin in there.  I don't completely understand
the formula, and I don't pretend to.  However, to say that the maximum number
of supportable users on a platform is about making money is just completely
wrong.  The maximum number is about avoiding a situation where we would
lose money.

 I can express my opinion and I did.  Maybe Digium will take notice,
 maybe they won't.

 Maybe next they will charge $250 for conference bridge capabilities.
  It's a joke to cripple things that can be enabled by flicking a
 switch.  Your system comes with eight ports of VM but for another $250
 we can give you 12..

I wasn't aware that you were a customer of either Switchvox or Business
Edition.  Last I checked, the open source version that you use is not
constrained in that way (and it isn't likely to be constrained in the future,
either).  The whole reason that Business Edition exists is because some
customers demand professional support for Asterisk, and paying for that
support costs money.  That's all.  Business Edition does not significantly
differ from the open source version -- the only reason we put a license
code on it is to ensure that when people call in for support, they have
essentially already prepaid for that support.

-- 
Tilghman

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Re: [asterisk-users] Asterisk concurrent calls count

2008-05-17 Thread Michael Graves

On Sat, 17 May 2008 18:32:57 -0500, Tilghman Lesher wrote:

 Maybe next they will charge $250 for conference bridge capabilities.
  It's a joke to cripple things that can be enabled by flicking a
 switch.  Your system comes with eight ports of VM but for another $250
 we can give you 12..

Forgive me for jumping in on this, but that's terribly naive.

I work in the broadcast and TV production business. Some time ago a
major company called Quantel created a hardware system called Edit
Box. It was wickedly fast and could do things with multiple streams of
uncompressed video in real-time. It was way beyond PCs, Macs or *nix
boxes of the day. It started at $500,000 USD for four streams.

I was at a Chicago post production facility the day that Quantel
delivered an upgrade that allowed the system to manipulate 8
simultaneous video streams. The field service tech walk in with a disk
and installed a software patch. Voila, twice as many layers. 

That upgrade cost another $250,000.

This kind of thing goes on all the time. The hardware has the core
capabilities but licensing controls your access to it. You get what you
pay for.

Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
[EMAIL PROTECTED]



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[asterisk-users] Paging

Hi List;

In the below link:

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page

I saw this line and did not find for it explaination,
anyone can explain it?

exten =
7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
PROTECTED]/nLocal/interal
[EMAIL PROTECTED]|)

What local means? and why to use @page?
What Local/[EMAIL PROTECTED]/n means?
What Local/interal [EMAIL PROTECTED]| means?

Any help?
Regards
Bilal


  

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Re: [asterisk-users] Using a Loopback Plug for an RJ-45 EthernetInterface for testing a Digium Card

2008-05-17 Thread Jose P. Espinal

Thank you very much for your replies!,

And thanks Alexander for the T1 loopback pins schema :)


Alexander Lopez wrote:
 It will go Green if a PROPER loopback plug is inserted.

 Pins 1 and 2 shorted to 4 ad 5

 Pin 1 to 4
 Pin 2 to 5

 Leave the others open...


   
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
 Sent: Saturday, May 17, 2008 6:02 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Using a Loopback Plug for an RJ-45
 EthernetInterface for testing a Digium Card

 On Sat, May 17, 2008 at 05:50:46PM -0400, Jose P. Espinal wrote:
 
 Someone told me about using a Loopback plug for RJ-45 for testing if
   
 a
   
 Digium Card gave him 'green' Alarm (for testing if the card had been
 damaged by  a strange voltage surge); would this have some bad side
   
 effect?

 Well, it's actually probably an RJ-48X interface, but who's counting.

 If your question, though, is would plugging a loopback-wired 8p8c
 
 modular
   
 plug into the jack on the back of a T-1 card cause it any damage in
 itself?, I can't imagine the answer would be yes, no.

 If you loop it and it stays in Red alarm, it's probably broken.

 I can't speak to exactly what the alarm status stuff does if the port
 you're looping expects to have a PRI plugged into it: I would expect
 Green, but no actual traffic, but I could be wrong, I'm a bit new on
 that front.

 Cheers,
 -- jra
 --
 Jay R. Ashworth   Baylink
 [EMAIL PROTECTED]
 Designer The Things I Think
 
 RFC
   
 2100
 Ashworth  Associates http://baylink.pitas.com
 
 '87
   
 e24
 St Petersburg FL USA  http://photo.imageinc.us +1 727
 
 647
   
 1274

   Those who cast the vote decide nothing.
   Those who count the vote decide everything.
 -- (Joseph Stalin)

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Re: [asterisk-users] Queue Stats

2008-05-17 Thread Mark Hamilton

Hi Nicolas,

Thank you so very much for this!
(Also on behalf of a large group of Asterisk queue users, I'm sure!)


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño
Sent: May 16, 2008 7:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queue Stats

Hello,

I have finally released the queue stats package to the public.. please go
to:

http://www.asternic.org/stats

To get it or see the online demo.

-- 
Nicolás Gudiño
Buenos Aires - Argentina

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Re: [asterisk-users] More dialplan visualization (neat graphs!)

2008-05-17 Thread Mark Hamilton

Martin,

That's a wicked visualization tool!
Thanks for this contribution!

Thanks,
Mark.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin B.
Smith
Sent: May 17, 2008 2:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] More dialplan visualization (neat graphs!)

Howdy all,

The Asterisk-Java project has included some rudimentary parsing related to
dialplans and extensions.conf. I've done a blog post at
http://asterisk-java.org/ related to it, and giving a demo of some dialplan
visualizations. It could eventually get fleshed out into an open-source
visual diaplan designer or visualizer. The web start demo requires Java 6.
I'd love your feedback.

Thanks,


Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 


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Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root

2008-05-17 Thread Mark Hamilton

Lee,

You should probably clean it up and put it up on the wiki. I don't think
anyone has put up a step-by-step like you did before.

There might be much easier additions/modifications done to it, and it will
be available to everybody. 

Thanks for this, btw.
Mark.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee, John
(Sydney)
Sent: May 16, 2008 4:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root

First of all, thanks Philipp, Alan, Tzafrir and James for your valuable
comments.  I have listed below the exact list of commands to run for
reinstalling asterisk 1.4.* as non-root on a Redhat / Fedora distro.
Hope others can benefit.

I have the following comments/questions though:
1) #What is safe_asterisk used for actually?  I did not touch it in
my modification because I don't know when is it triggered?
2) #I do not actually know whether we really need to modify
/etc/asterisk/asterisk.conf?  Is this file read by asterisk at all?
Seems like an important file name - asterisk.conf?


3) It is safer to define a user called asterisk in group asterisk unless
you want to make more changes to 2 files i.e. zaptel.rules and
/etc/init.d/asterisk
4) There is an additional chmod to run for letting voicemail.conf to be
written by group asterisk.



# /etc/init.d/asterisk stop
Shutting down asterisk:[  OK  ]

# /usr/sbin/groupadd asterisk

# /usr/sbin/useradd -d /var/lib/asterisk -g asterisk asterisk
useradd: warning: the home directory already exists.
Not copying any file from skel directory into it.

# cp Makefile Makefile.org

*** 
*** Change the following line from:
***
*** ASTVARRUNDIR=$(localstatedir)/run
***
*** to
***
*** ASTVARRUNDIR=$(localstatedir)/run/asterisk
***
# vi Makefile
[...]
ifeq ($(OSARCH),SunOS)
  ASTETCDIR=/var/etc/asterisk
  ASTLIBDIR=/opt/asterisk/lib
  ASTVARLIBDIR=/var/opt/asterisk
  ASTSPOOLDIR=/var/spool/asterisk
  ASTLOGDIR=/var/log/asterisk
  ASTHEADERDIR=/opt/asterisk/include
  ASTBINDIR=/opt/asterisk/bin
  ASTSBINDIR=/opt/asterisk/sbin
  ASTVARRUNDIR=/var/run/asterisk
  ASTMANDIR=/opt/asterisk/man
else
  ASTETCDIR=$(sysconfdir)/asterisk
  ASTLIBDIR=$(libdir)/asterisk
  ASTHEADERDIR=$(includedir)/asterisk
  ASTBINDIR=$(bindir)
  ASTSBINDIR=$(sbindir)
  ASTSPOOLDIR=$(localstatedir)/spool/asterisk
  ASTLOGDIR=$(localstatedir)/log/asterisk
  ASTVARRUNDIR=$(localstatedir)/run/asterisk
  ASTMANDIR=$(mandir)
[...]

cd /usr/src/asterisk-1.4
make clean
./configure
make
make install

***
*** Don't panic!  
*** /var/run/asterisk should just be an empty directory but should just
exist.
***

# chown --recursive asterisk:asterisk /var/lib/asterisk
# chown --recursive asterisk:asterisk /var/log/asterisk
# chown --recursive asterisk:asterisk /var/run/asterisk
# chown --recursive asterisk:asterisk /var/spool/asterisk
# chown --recursive asterisk:asterisk /usr/lib/asterisk
# chown --recursive asterisk:asterisk /dev/zap

# chmod --recursive u=rwX,g=rX,o= /var/lib/asterisk
# chmod --recursive u=rwX,g=rX,o= /var/log/asterisk
# chmod --recursive u=rwX,g=rX,o= /var/run/asterisk
# chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk
# chmod --recursive u=rwX,g=rX,o= /usr/lib/asterisk
# chmod --recursive u=rwX,g=rX,o= /dev/zap

# chown --recursive root:asterisk /etc/asterisk
# chmod --recursive u=rwX,g=rX,o= /etc/asterisk

# cp /etc/asterisk/asterisk.conf /etc/asterisk/asterisk.conf.org
# vi /etc/asterisk/asterisk.conf
*** 
*** Change the following line from:
***
*** astrundir = /var/run
***
*** to
***
*** astrundir = /var/run/asterisk
***

# cp /etc/init.d/asterisk /etc/init.d/asterisk.org

# vi /etc/init.d/asterisk
*** 
*** Uncomment the following line from:
***
*** #AST_USER=asterisk
*** #AST_GROUP=asterisk
***
*** to
***
*** AST_USER=asterisk
*** AST_GROUP=asterisk
***

*** 
*** Asterisk needs to write to voicemail.conf for password change.
***
# chmod g+w /etc/asterisk/voicemail.conf

*** 
*** Restart Asterisk by either of below:
***
# /etc/init.d/asterisk restart

# asterisk -U asterisk -G asterisk

Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 06:34:12PM -0400, Steve Totaro wrote:
 End of life date for Asterisk 1.2 was August 1, 2007.

Well, my app won't *run* on 1.4 reliably yet, so I hope they get it
fixed soon...

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 06:43:51PM -0400, Steve Totaro wrote:
 Maybe next they will charge $250 for conference bridge capabilities.
  It's a joke to cripple things that can be enabled by flicking a
 switch.  Your system comes with eight ports of VM but for another $250
 we can give you 12..

Feature pricing (also called value pricing) is a time honored
tradition -- especially in the telecom business.

Ever wonder why your telco charged you $2.50 a month for call waiting
when it involved exactly, let me see, right: *no hardware at all*?

Because they could. 

And more to the point: because Nortel charged *them* $20k a year[1] to
enable the feature in the generic, and they were damn sure gonna get 
that money back from someone.

In this case, *they give away the entire source package.  For free*.

I like you a lot, Steve, generally, but it feels a bit like you're
whining, on this one, to me...

Cheers,
-- jra
[1] These features were indeed charged for, though usually in packages;
I don't have exact numbers -- though I *do* have a DMS 100 Feature
Portfolio on my bookshelf, so I could give you a feature number, if you
like.
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 07:07:51PM -0500, Michael Graves wrote:
 I work in the broadcast and TV production business. Some time ago a
 major company called Quantel created a hardware system called Edit
 Box. It was wickedly fast and could do things with multiple streams of
 uncompressed video in real-time. It was way beyond PCs, Macs or *nix
 boxes of the day. It started at $500,000 USD for four streams.

... and now I can do that on a Mac laptop.

Aren't you glad *you* didn't have to write that check?  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Discover connected Zap lines

On Sun, May 18, 2008 at 01:52:03AM +0300, Tzafrir Cohen wrote:
 You should see (RED) in /proc/zaptel/1 for the channel if it is
 disconnected. Not to mention that the channel will be in alarm (InAlarm
 in zap show channel NNN).

Is that true for *all* makes of card?  I know the Sangomas put it in
ifconfig, but for pick-it-off-with-SNMP-for-Nagios purposes, that will
make my life a lot easier...

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Discover connected Zap lines