Re: [asterisk-users] Asterisk PBX causes mysql to take more CPU time

[Miguel Molina]

MURALI V escribió:

Hi All,
 
 
  I am using asterisk-1.4.22 with mysql as my storage medium for 
voice messages.Right now i am running 700+ extensions with this setup .
 
*System Configuration:*
 
asterisk-1.4.22.1

(using odbc storage for voice messages)
 
unixODBC-2.2.11-7.1

mysql-connector-odbc-3.51.12-2.2
mysql-server-5.0.45-7.el5
CentOS-5.2 (64 bit)
 
 
*The following are the issues i am facing:*
 
  1) CPU time always shows 30%-90% for mysql when asterisk is 
running . The CPU time goes normal for mysql  when asterisk is stopped.
Obviously the MySQL CPU time droppes to near zero if the only process 
who uses it stops! Check what kind of queries asterisk is making and if 
that can be optimized.
 
  2) calls are getting dropped suddenly.
 
  3) Some answered calls does not hear anything at the 
receiver/initiater end.
 
  4) asterisk itself gets crashed at times
 
  5) Core dumps are getting created.
Follow the guidelines for obtaining a backtrace with GDB. These can be 
found in the file "asterisk/doc/backtrace.txt"


Look in http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging

If you learn to read the backtrace, you can know with great detail what 
is causing asterisk to crash. You may run into known bugs, so you can 
check the bugtracker and/or the recent versions Changelog to see if it's 
solved. Maybe you'll have to decide to upgrade it or not, or upgrade in 
a separate machine and test throughoutly before putting the new version 
into production.


Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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[asterisk-users] Asterisk PBX causes mysql to take more CPU time

[MURALI V]

Hi All,


  I am using asterisk-1.4.22 with mysql as my storage medium for voice
messages.Right now i am running 700+ extensions with this setup .

*System Configuration:*

asterisk-1.4.22.1
(using odbc storage for voice messages)

unixODBC-2.2.11-7.1
mysql-connector-odbc-3.51.12-2.2
mysql-server-5.0.45-7.el5
CentOS-5.2 (64 bit)


*The following are the issues i am facing:*

  1) CPU time always shows 30%-90% for mysql when asterisk is running .
The CPU time goes normal for mysql  when asterisk is stopped.

  2) calls are getting dropped suddenly.

  3) Some answered calls does not hear anything at the
receiver/initiater end.

  4) asterisk itself gets crashed at times

  5) Core dumps are getting created.

Is there any optimizations or configurations needs to be done on my server
in order to drill down this issue?

Please help me out in troubleshooting this issue, This problems are getting
worsen day by day.

Thanks in advance.

-- 
always   withsmile
vimurli
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Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2

[Tzafrir Cohen]

On Sat, Jul 12, 2008 at 01:14:36AM +0100, Grey Man wrote:
> On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis <[EMAIL PROTECTED]> wrote:
> > I've just added a PREVIEW release of my upcoming how-to guide for Asterisk
> > PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2.
> > It addresses all kinks and showstoppers that many people have experienced
> > over the past year or so. Because this is a preview, it is not the final
> > version of this guide. It is subject to change (format, copy, layout, etc.)
> >
> > To view and download this guide, please visit
> > http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/
> >
> > Please take this opportunity to test the guide and provide any feedback. The
> > official release is set for Wednesday, July 16 and will be available on
> > CloudCrunch.
> >
> 
> There's already 4 public images on ec2 mentioning Asterisk in their
> names so wouldn't it be easier to try out one of those rather than
> install all the bits and pieces on a base Linux image?
> 
> An interesting paper on ec2 and Asterisk would be one that discusses
> what the call quality is like from both inside and outside the US.
> When I briefy ran up an instance at this time last year it actually
> seemed ok.
> 
> >From a provider's point of view running Asterisk on the ec2 cloud does
> pose some interesting questions. As a quick and dirty estimate if you
> assume one of the standard small ec2 instances could cope with 100
> simultaneous g711 calls (I don't know if that is the case just
> guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth
> both ways). Assuming that you'd then have 1MB/s average to account for
> quite and busy call times then it would be 3.6GB/hour or 86.4GB/day.
> At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The
> server instance will cost you $72/month so total cost for 100/calls
> per month is $332.
> 
> A typical dedicated server for $300/month is roughly equivalent to an
> ec2 small instance and comes with 500GB of bandwidth/month which is
> only a fifth of what's required but you could probably get the extra
> 2TB/month thrown in for $32/month making the dedicated server and ec2
> prices the same.

Another option, especially for lighter loads, is a virtual server
instance. There are many providers who will provide you a Xen guest for
e.g. 20$ or 30$ a month. For a lighter load this should be more than
enough. Those normally come with a dedicated IP address. Support plans
vary greatly from one provider to another.

-- 
   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 PBX How-to Guide for Amazon EC2

[Grey Man]

On Sat, Jul 12, 2008 at 2:16 AM, Grygoriy Dobrovolskyy
<[EMAIL PROTECTED]> wrote:
> Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited.

And I'm sure there will be someone somewhere who has a better deal as
well. Whenever hosting gets mentioned in tech forums you always end up
with lots of opinions (or pitches depending on how well they are
disguised).

My post was more in the vein of ballpark figures to compare the
technincal merits of a grid approach, in this case manifested by
Amazon's ec2, to a dedicated server approach for running a VoIP
provider service using Asterisk. It's probably already gone off the
OP's original topic enough now that they'll be people jumping up and
down that this should be on the Biz list...

Regards,

Greyman.

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Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2

[Grygoriy Dobrovolskyy]

Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited.

CPU
Intel  Xeon X5355
1x 4x 2.66 GHz
L2: 8Mo, FSB: 1333MHzQuadruple Coeur
Architecture64 bits RAM
8 Go FBDIMM DDR2 HDD
2x 750 Go Type HDD
SATA2 RAID HARD 1Interfaces
2 x 1 Gbps SPEED
2 Gbps Traffic
UNLIMITED
IP fixe2 adresses IP Fail-over +8
adressesVPS Ready [image:
Oui]IP Fail-over VPS64 IP (/26)IP enregistrées RIPE[image: Oui] Sauvegarde
FTP 750 Go

2008/7/12 Grey Man <[EMAIL PROTECTED]>:

> On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis <[EMAIL PROTECTED]>
> wrote:
> > I've just added a PREVIEW release of my upcoming how-to guide for
> Asterisk
> > PBX on EC2. It is based on months of testing and evaluating Asterisk on
> EC2.
> > It addresses all kinks and showstoppers that many people have experienced
> > over the past year or so. Because this is a preview, it is not the final
> > version of this guide. It is subject to change (format, copy, layout,
> etc.)
> >
> > To view and download this guide, please visit
> >
> http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/
> >
> > Please take this opportunity to test the guide and provide any feedback.
> The
> > official release is set for Wednesday, July 16 and will be available on
> > CloudCrunch.
> >
>
> There's already 4 public images on ec2 mentioning Asterisk in their
> names so wouldn't it be easier to try out one of those rather than
> install all the bits and pieces on a base Linux image?
>
> An interesting paper on ec2 and Asterisk would be one that discusses
> what the call quality is like from both inside and outside the US.
> When I briefy ran up an instance at this time last year it actually
> seemed ok.
>
> From a provider's point of view running Asterisk on the ec2 cloud does
> pose some interesting questions. As a quick and dirty estimate if you
> assume one of the standard small ec2 instances could cope with 100
> simultaneous g711 calls (I don't know if that is the case just
> guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth
> both ways). Assuming that you'd then have 1MB/s average to account for
> quite and busy call times then it would be 3.6GB/hour or 86.4GB/day.
> At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The
> server instance will cost you $72/month so total cost for 100/calls
> per month is $332.
>
> A typical dedicated server for $300/month is roughly equivalent to an
> ec2 small instance and comes with 500GB of bandwidth/month which is
> only a fifth of what's required but you could probably get the extra
> 2TB/month thrown in for $32/month making the dedicated server and ec2
> prices the same.
>
> There are serious pros and cons between these approaches. With the ec2
> you don't get a permanent static IP, with a dedicated server you do.
> With ec2 you could scale up and down between 1 server and 4 servers at
> the drop of a hat to save costs and cope with peak and quite times,
> with dedicated servers you're stuff with 12 or 24 month contracts for
> the number of servers you'd need under maximum load. And then of
> course the major factor for both is what the call quality will be
> like.
>
> Regards,
>
> Greyman.
>
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Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2

[Grey Man]

On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis <[EMAIL PROTECTED]> wrote:
> I've just added a PREVIEW release of my upcoming how-to guide for Asterisk
> PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2.
> It addresses all kinks and showstoppers that many people have experienced
> over the past year or so. Because this is a preview, it is not the final
> version of this guide. It is subject to change (format, copy, layout, etc.)
>
> To view and download this guide, please visit
> http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/
>
> Please take this opportunity to test the guide and provide any feedback. The
> official release is set for Wednesday, July 16 and will be available on
> CloudCrunch.
>

There's already 4 public images on ec2 mentioning Asterisk in their
names so wouldn't it be easier to try out one of those rather than
install all the bits and pieces on a base Linux image?

An interesting paper on ec2 and Asterisk would be one that discusses
what the call quality is like from both inside and outside the US.
When I briefy ran up an instance at this time last year it actually
seemed ok.

>From a provider's point of view running Asterisk on the ec2 cloud does
pose some interesting questions. As a quick and dirty estimate if you
assume one of the standard small ec2 instances could cope with 100
simultaneous g711 calls (I don't know if that is the case just
guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth
both ways). Assuming that you'd then have 1MB/s average to account for
quite and busy call times then it would be 3.6GB/hour or 86.4GB/day.
At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The
server instance will cost you $72/month so total cost for 100/calls
per month is $332.

A typical dedicated server for $300/month is roughly equivalent to an
ec2 small instance and comes with 500GB of bandwidth/month which is
only a fifth of what's required but you could probably get the extra
2TB/month thrown in for $32/month making the dedicated server and ec2
prices the same.

There are serious pros and cons between these approaches. With the ec2
you don't get a permanent static IP, with a dedicated server you do.
With ec2 you could scale up and down between 1 server and 4 servers at
the drop of a hat to save costs and cope with peak and quite times,
with dedicated servers you're stuff with 12 or 24 month contracts for
the number of servers you'd need under maximum load. And then of
course the major factor for both is what the call quality will be
like.

Regards,

Greyman.

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Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2

[MFH]

Very cool, you've piqued my interest.  Since I haven't launched an 
instance before, where's the best place to learn to do that?  What's the 
approximate monthly cost of hosting an Asterisk PBX on EC2?

Ronald Lewis wrote:
> I've just added a PREVIEW release of my upcoming how-to guide for 
> Asterisk PBX on EC2. It is based on months of testing and evaluating 
> Asterisk on EC2. It addresses all kinks and showstoppers that many 
> people have experienced over the past year or so. Because this is a 
> preview, it is not the final version of this guide. It is subject to 
> change (format, copy, layout, etc.)
>
> To view and download this guide, please visit 
> http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/
>
> Please take this opportunity to test the guide and provide any 
> feedback. The official release is set for Wednesday, July 16 and will 
> be available on CloudCrunch.
>
> Thanks!
>
> Ronald Lewis
> Denver, Colorado
> http://ronaldlewis.com
> 
>
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[asterisk-users] Asterisk PBX How-to Guide for Amazon EC2

[Ronald Lewis]

I've just added a PREVIEW release of my upcoming how-to guide for Asterisk
PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2.
It addresses all kinks and showstoppers that many people have experienced
over the past year or so. Because this is a preview, it is not the final
version of this guide. It is subject to change (format, copy, layout, etc.)

To view and download this guide, please visit
http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/

Please take this opportunity to test the guide and provide any feedback. The
official release is set for Wednesday, July 16 and will be available on
CloudCrunch.

Thanks!

Ronald Lewis
Denver, Colorado
http://ronaldlewis.com
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Re: [asterisk-users] Asterisk PBX using Outbound proxy

[Steve Davies]

On 18/04/2008, Rosa De Santis <[EMAIL PROTECTED]> wrote:
>
>  Hi all.
>
>  Please, how can I configure an Asterisk PBX using an outbound proxy (that
> resolve NAT Traversal)
>
>  I'm trying using the outboundproxy and outboundproxyport values in sip.conf
> but the PBX don't get registered on the outbound proxy side.
>
>  I'm using SER + Asterisk with Jasomi outbound proxy solution, and I want
> the PBX to have a SIP trunk, but in SER i see the pbx sip user registered as
> [EMAIL PROTECTED] , not the SER ip.
>
>  Any ideas please?
>  Thanks in advance, Rosa.

What "register =>" line are you using?

Take a look at:

  http://bugs.digium.com/view.php?id=12474

Perhaps it will help?

Regards,
Steve

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[asterisk-users] Asterisk PBX using Outbound proxy

[Rosa De Santis]

Hi all.
 
Please, how can I configure an Asterisk PBX using an outbound proxy (that 
resolve NAT Traversal)
 
I'm trying using the outboundproxy and outboundproxyport values in sip.conf but 
the PBX don't get registered on the outbound proxy side.
 
I'm using SER + Asterisk with Jasomi outbound proxy solution, and I want the 
PBX to have a SIP trunk, but in SER i see the pbx sip user registered as [EMAIL 
PROTECTED] , not the SER ip.
 
Any ideas please?
Thanks in advance, Rosa.
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Re: [asterisk-users] Asterisk PBX '&' '||' Grandstream GXP-2000 problem

[David Gomillion]

On 1/15/07, J. Espinal <[EMAIL PROTECTED]> wrote:


 Hi People,

We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk
PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHzBox... 
The issues that we are experiencing involves our Telephone
Operator's/Receptionist whom answer multiple incoming calls... As an
example.., when they answer line 1 and Line 2 starts to ring they would ask
the person on line 1 to "hold" and proceed to answer line 2 and forward line
2 to to the requested extension. The problem is when they attempt to pick
line 1 off the "hold" in order to handle that call, line 1 is either dropped
or the Grandstream Phone freezes and the user is forced to rest the phone.
The situation persist whenever there are multiple lines active with incoming
calls and upon answering one, placing the line on hold and attempting to
answer the other lines active calls will be dropped the the phone just
hangs/freezes. We know that the call is dropped because the people call back
complaining about being hung up on We have had our dedicated T1 (for
voice only) tested several times and it is good. We have had the Asterisk
PBX completely redone and gone over thoroughly and are at the point where we
are suspecting the configuration file for the Grandstream GXP-2000 Telephone
as the culprit. We would like to know what suggestions anyone out there
might have if any... Thanks,



Are you using G.729? Last I heard, grandstreams could only have one call via
G.729 at a time. It had something to do with the licensing that they used, I
think. Just a thought...
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[asterisk-users] Asterisk PBX '&' '||' Grandstream GXP-2000 problem

[J. Espinal]

Hi People,

   We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk 
PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz 
Box... The issues that we are experiencing involves our Telephone 
Operator's/Receptionist whom answer multiple incoming calls... As an 
example.., when they answer line 1 and Line 2 starts to ring they would 
ask the person on line 1 to "hold" and proceed to answer line 2 and 
forward line 2 to to the requested extension. The problem is when they 
attempt to pick line 1 off the "hold" in order to handle that call, line 
1 is either dropped or the Grandstream Phone freezes and the user is 
forced to rest the phone. The situation persist whenever there are 
multiple lines active with incoming calls and upon answering one, 
placing the line on hold and attempting to answer the other lines active 
calls will be dropped the the phone just hangs/freezes. We know that the 
call is dropped because the people call back complaining about being 
hung up on We have had our dedicated T1 (for voice only) tested 
several times and it is good. We have had the Asterisk PBX completely 
redone and gone over thoroughly and are at the point where we are 
suspecting the configuration file for the Grandstream GXP-2000 Telephone 
as the culprit. We would like to know what suggestions anyone out there 
might have if any... Thanks,



Jose P. Espinal
DomiNET

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Re: [asterisk-users] Asterisk PBX to a Nortel MICS PBX

[Anthony Rodgers]

Can you be more specific? What sort of linkages are available between  
the two offices?


CP

On 22-Oct-06, at 10:38 PM, dthurn wrote:


What's the best way to connect an Asterisk PBX to a Nortel MICS PBX.
I have two offices that I want to link together.


TTFN

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[asterisk-users] Asterisk PBX to a Nortel MICS PBX

[dthurn]

What's the best way to connect an Asterisk PBX to a Nortel MICS PBX.
I have two offices that I want to link together.


TTFN

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Re: [Asterisk-Users] Asterisk PBX in Debian

[Tzafrir Cohen]

Hi

Please use proper quoting...

See below

On Sat, Oct 08, 2005 at 12:23:21AM -0400, [EMAIL PROTECTED] wrote:
>  -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
> Cohen
> Sent: Friday, October 07, 2005 8:17 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Asterisk PBX in Debian
> 
> > On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote:
> 
> > > 
> > > Besides I found that using packages with asterisk on debian can do odd
> > 
> > > things to your custom sound files if you do a remove.
> > 
> > Regarding the sounds files: I don't think that the way Asterisk
> > installer handles them is very optimal either.
> > 
> > Your message got me thinking, though. I believe that Debian is right
> > installing all sounds to /usr/share/asterisk/sounds . But
> > /var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be
> > kept for custom sounds that are never touched by the package. 
> > 
> > I figure that file.c:build_filename could be changed to do the
> > following:
> > 
> >   if exists /var/lib/asterisk/sounds/
> > return /var/lib/asterisk/sounds/
> >   else if exists /usr/share/sounds/asterisk/
> > return /usr/share/sounds/asterisk/
> > 
> > What do you think? I figure I'll try to push this into Debian first.
> > (If this is indeed a good idea)
> 
> Using /var works, but setting it in asterisk could be a pain when it
> comes to voicemail prompts.  Plus, extensions.conf would need to grow
> and become a little cluttered.  Unless of course, one could do something
> to specify a new root voicemail path, and if the file is not found it
> plays from the default.

You missed the point: you still keep the same configuration. 
should first be looked for in the custom directory and only afterwards
in /usr/share/sounds/asterisk/sounds .

voicemail/ is currently (in Debian) a symlink from the sounds directory 
to /var/lib/asterisk/voicemail

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] Asterisk PBX in Debian

[gw]

 -Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 8:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX in Debian

On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote:
> Why bother with packages anyhow?  I just installed debian base and did

> a cvs get for head, and all good to go.

And if you have several systems?
>>>
I would make a custom package in that case, for easy updating.  Depends
of course if you are using head or not.

> 
> Besides I found that using packages with asterisk on debian can do odd

> things to your custom sound files if you do a remove.

Regarding the sounds files: I don't think that the way Asterisk
installer handles them is very optimal either.

Your message got me thinking, though. I believe that Debian is right
installing all sounds to /usr/share/asterisk/sounds . But
/var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be
kept for custom sounds that are never touched by the package. 

I figure that file.c:build_filename could be changed to do the
following:

  if exists /var/lib/asterisk/sounds/
return /var/lib/asterisk/sounds/
  else if exists /usr/share/sounds/asterisk/
return /usr/share/sounds/asterisk/

What do you think? I figure I'll try to push this into Debian first.
(If this is indeed a good idea)
>>>
Using /var works, but setting it in asterisk could be a pain when it
comes to voicemail prompts.  Plus, extensions.conf would need to grow
and become a little cluttered.  Unless of course, one could do something
to specify a new root voicemail path, and if the file is not found it
plays from the default.

Greg
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Re: [Asterisk-Users] Asterisk PBX in Debian

[Paul]

Michael Coburn wrote:


Or why not create all sound files under /usr/share/asterisk/sounds and
then subdirs from there for your own touched files i.e.
/usr/share/asterisk/sounds/custom ?
--
Michael Coburn
 


Short answer is because we live by rules.

Any good linux distro follows standards regarding the proper location of 
files. /usr/local /opt and /var/lib would be more appropriate for 
locally added sounds.


This is good reading for people new to linux. Maybe you always wondered 
why there is a /bin, /usr/bin and /usr/local/bin? This will enlighten you:


http://www.debian.org/doc/packaging-manuals/fhs/



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 8:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX in Debian

On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote:
 


Why bother with packages anyhow?  I just installed debian base and did
   


a
 


cvs get for head, and all good to go.
   



And if you have several systems?

 


Besides I found that using packages with asterisk on debian can do odd
things to your custom sound files if you do a remove.
   



Regarding the sounds files: I don't think that the way Asterisk
installer handles them is very optimal either.

Your message got me thinking, though. I believe that Debian is right
installing all sounds to /usr/share/asterisk/sounds . But
/var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be 
kept for custom sounds that are never touched by the package. 


I figure that file.c:build_filename could be changed to do the
following:

 if exists /var/lib/asterisk/sounds/
   return /var/lib/asterisk/sounds/
 else if exists /usr/share/sounds/asterisk/
   return /usr/share/sounds/asterisk/

What do you think? I figure I'll try to push this into Debian first.
(If this is indeed a good idea)

 



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RE: [Asterisk-Users] Asterisk PBX in Debian

[Michael Coburn]

Or why not create all sound files under /usr/share/asterisk/sounds and
then subdirs from there for your own touched files i.e.
/usr/share/asterisk/sounds/custom ?
--
Michael Coburn

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 8:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX in Debian

On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote:
> Why bother with packages anyhow?  I just installed debian base and did
a
> cvs get for head, and all good to go.

And if you have several systems?

> 
> Besides I found that using packages with asterisk on debian can do odd
> things to your custom sound files if you do a remove.

Regarding the sounds files: I don't think that the way Asterisk
installer handles them is very optimal either.

Your message got me thinking, though. I believe that Debian is right
installing all sounds to /usr/share/asterisk/sounds . But
/var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be 
kept for custom sounds that are never touched by the package. 

I figure that file.c:build_filename could be changed to do the
following:

  if exists /var/lib/asterisk/sounds/
return /var/lib/asterisk/sounds/
  else if exists /usr/share/sounds/asterisk/
return /usr/share/sounds/asterisk/

What do you think? I figure I'll try to push this into Debian first.
(If this is indeed a good idea)

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Asterisk PBX in Debian

[Tzafrir Cohen]

On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote:
> Why bother with packages anyhow?  I just installed debian base and did a
> cvs get for head, and all good to go.

And if you have several systems?

> 
> Besides I found that using packages with asterisk on debian can do odd
> things to your custom sound files if you do a remove.

Regarding the sounds files: I don't think that the way Asterisk
installer handles them is very optimal either.

Your message got me thinking, though. I believe that Debian is right
installing all sounds to /usr/share/asterisk/sounds . But
/var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be 
kept for custom sounds that are never touched by the package. 

I figure that file.c:build_filename could be changed to do the
following:

  if exists /var/lib/asterisk/sounds/
return /var/lib/asterisk/sounds/
  else if exists /usr/share/sounds/asterisk/
return /usr/share/sounds/asterisk/

What do you think? I figure I'll try to push this into Debian first.
(If this is indeed a good idea)

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] Asterisk PBX in Debian

[gw]

Why bother with packages anyhow?  I just installed debian base and did a
cvs get for head, and all good to go.

Besides I found that using packages with asterisk on debian can do odd
things to your custom sound files if you do a remove.

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 7:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX in Debian

On Fri, Oct 07, 2005 at 12:51:25AM -0500, Carlos Prieto wrote:
> Hi everyone.
> 
> I've installed Asterisk PBX using apt packages, but i don't have 
> actually any Digium card, so i want to use ztdummy.
> 
> I've tried to modify the Makefiles in the debian source package, i 
> don't get any error, but still the ztdummy module doesn't get
compiled.

What file exactly did you edit? What command did you run? I suspect the
file you have edited got overrun.

> 
> Does anybody has idea how to get the ztdummy module using the debian 
> package system?

I'm not sure you need. Packages from 

  deb http://rapid.dotsrc.org/rapid sarge main

already have ztdummy and ztdummy is on by default in the zaptel-source
package.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Asterisk PBX in Debian

[Tzafrir Cohen]

On Fri, Oct 07, 2005 at 12:51:25AM -0500, Carlos Prieto wrote:
> Hi everyone.
> 
> I've installed Asterisk PBX using apt packages, but i don't have actually
> any Digium card, so i want to use ztdummy.
> 
> I've tried to modify the Makefiles in the debian source package, i don't get
> any error, but still the ztdummy module doesn't get compiled.

What file exactly did you edit? What command did you run? I suspect the
file you have edited got overrun.

> 
> Does anybody has idea how to get the ztdummy module using the debian package
> system?

I'm not sure you need. Packages from 

  deb http://rapid.dotsrc.org/rapid sarge main

already have ztdummy and ztdummy is on by default in the zaptel-source
package.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Asterisk PBX in Debian

[Carlos Prieto]

Hi everyone.

I've installed Asterisk PBX using apt packages, but i don't have actually any Digium card, so i want to use ztdummy.

I've tried to modify the Makefiles in the debian source package, i
don't get any error, but still the ztdummy module doesn't get compiled.

Does anybody has idea how to get the ztdummy module using the debian package system?

Thanks for your help.
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Re: [Asterisk-Users] Asterisk PBX

[Francesco Peeters]

On Wed, September 21, 2005 15:30, Nathan Pralle said:
>>  + site:lists.difium.com
>
> The above is good when searching for information on Joe Diffie --
> Otherwise, you'll want:
>
>  + site:lists.digium.com
>
> :)
>
> Nathan
>
>


I already corrected myself... I canna help them list servers take so long!
 ;-)

---FP
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Re: [Asterisk-Users] Asterisk PBX

[Nathan Pralle]

 + site:lists.difium.com


The above is good when searching for information on Joe Diffie -- 
Otherwise, you'll want:


 + site:lists.digium.com

:)

Nathan


--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
-
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Re: [Asterisk-Users] Asterisk PBX

[Francesco Peeters]

On Wed, September 21, 2005 15:19, Francesco Peeters said:
> On Wed, September 21, 2005 15:14, Tom Rymes said:
>> On Wed, 21 Sep 2005, kapil dhawan wrote:
> 
>> If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this 
>> list
>> will be your best tools.
>>
>> Tom
>
> I'd like to add Google to that shortlist:
>
>  + site:voip-info.org
> or
>  + site:lists.difium.com
>
> will help you quickly search the wiki and list archives...
>
> Good luck!
>
Oops! Typo!

 + site:lists.digium.com

is the correct syntax...

Sorry!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] Asterisk PBX

[Francesco Peeters]

On Wed, September 21, 2005 15:14, Tom Rymes said:
> On Wed, 21 Sep 2005, kapil dhawan wrote:

> If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this 
> list
> will be your best tools.
>
> Tom

I'd like to add Google to that shortlist:

 + site:voip-info.org
or
 + site:lists.difium.com

will help you quickly search the wiki and list archives...

Good luck!


-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] Asterisk PBX

[Tom Rymes]

On Wed, 21 Sep 2005, kapil dhawan wrote:



Hi List

I am very new to Asterisk but have been alloted a job to replace my  
traditional PBX with it. Kindly provide me some useful info (PDF's  
etc) to setup Asterisk with FXO and FXS both.


I have to cater some 60 users with 10 simultaneous calls.

Regards


Go to asteriskathome.sourceforge.net and download [EMAIL PROTECTED] Be  
*VERY*** *VERY* *VERY* careful with the CD  
you burn, though, as when booted it will erase your hard drive and  
install CentOS ***WITHOUT WARNING


Other than that, it is a very good way to replace a PBX for an office  
of that size. Of course, you will need to select phones, server  
hardware, PSTN interconnect hardware, etc. as well. Considering how  
important phones are to the average business, you might want to  
consider hiring a consultant (might I recommend   me?)  
to help get you up to speed. It's just a thought, but getting help  
from someone who has already done this might keep you from making a  
few expensive mistakes (ie: buying equipment that is over/ 
underpowered, unreliable, low quality, etc.)


If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this list  
will be your best tools.


Tom
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Re: [Asterisk-Users] Asterisk PBX

[Sahil Gupta]

Hi Kapil,
AFAIK, there are no such PDF's that exist unless someone has really spent 
time compiling such information, which will be great to see.


However, if you check out www.voip-info.org, its a complete mine of useful 
information regarding doing what you wish to.


Regards,


Sahil Gupta
VoiceValley

On Wed, 21 Sep 2005, kapil dhawan wrote:


Hi List

I am very new to Asterisk but have been alloted a job to replace my 
traditional PBX with it. Kindly provide me some useful info (PDF's etc) to 
setup Asterisk with FXO and FXS both.


I have to cater some 60 users with 10 simultaneous calls.

Regards

_
Biography of Shah Rukh. His profile, awards, films. 
http://server1.msn.co.in/Profile/shahrukh.asp Find more here!


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[Asterisk-Users] Asterisk PBX

[kapil dhawan]

Hi List

I am very new to Asterisk but have been alloted a job to replace my 
traditional PBX with it. Kindly provide me some useful info (PDF's etc) to 
setup Asterisk with FXO and FXS both.


I have to cater some 60 users with 10 simultaneous calls.

Regards

_
Biography of Shah Rukh. His profile, awards, films. 
http://server1.msn.co.in/Profile/shahrukh.asp Find more here!


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[Asterisk-Users] asterisk PBX and Siemens Hipath 3750

[Varun Pabrai]

Hello
   I am planning to build a small PBX using
TDM22B.

We have a Siemens Hipath 3750 in operation
already.

When I manage to complete my PBX using TDM22B
I would ofcourse like to be able to connect my Asterisks PBX
with the Siemens Hipath 3750 PBX.

Will there be any issues regarding my plan ?
Or is there any other issues that I need to take
into account vis-a-vis Siemens PBX.

I have never done all this before so I would
appreciate any inputs.

Thanks in advance

Varun
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Re: [Asterisk-Users] Asterisk PBX with X100P in India

[Min Hwan Chang]

My Setup is below:
X100P connected to POTS
Siemens IP Phone
[EMAIL PROTECTED] v0.6

When I try dialing out the POTS line in India using a Siemens IP
phone, I'll hear the phone ring for a second, then hear loud beeps.
The loud beeps I believe are from the POTS line?

I am currently trying to dial a Cell phone number within India, so the
general format is 09X.

I have my extensions.conf set up like so for Outbound Context:

--
[outbound-local]
exten => _X,1,Macro(dialout-default,09${EXTEN})

[outbound-ld]
exten => _001NXXNXX,1,Macro(dialout-default,${EXTEN})

-
For the life of me I can't' seem to figure out what the problem is.
Thanks for the help!

Regards,
Min


On 4/15/05, Julio Saura <[EMAIL PROTECTED]> wrote:
> 
> what kind of problems do u have?
> 
> can u explain more in detail so we can try helping you?
> 
> best regards
> 
> El vie, 15-04-2005 a las 01:29 -0700, Min Hwan Chang escribió:
> > I'm currently trying to set up an Asterisk PBX system in India.
> > However I'm having trouble configuring the X100P to dial out on the
> > POTS line.  Does anyone have any knowledge about this?
> >
> > I know the telephone system is a bit different in India, so would the
> > X100P not be suitable?  Is there a change I need to make in the
> > Zaptel.conf or zapata.conf?
> >
> > Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also
> > pretty frustrating...
> >
> > Any help here would be appreciated.
> > ___
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> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Asterisk PBX with X100P in India

[Julio Saura]

what kind of problems do u have?

can u explain more in detail so we can try helping you?

best regards


El vie, 15-04-2005 a las 01:29 -0700, Min Hwan Chang escribió:
> I'm currently trying to set up an Asterisk PBX system in India.
> However I'm having trouble configuring the X100P to dial out on the
> POTS line.  Does anyone have any knowledge about this?
> 
> I know the telephone system is a bit different in India, so would the
> X100P not be suitable?  Is there a change I need to make in the
> Zaptel.conf or zapata.conf?
> 
> Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also
> pretty frustrating...
> 
> Any help here would be appreciated.
> ___
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[Asterisk-Users] Asterisk PBX with X100P in India

[Min Hwan Chang]

I'm currently trying to set up an Asterisk PBX system in India.
However I'm having trouble configuring the X100P to dial out on the
POTS line.  Does anyone have any knowledge about this?

I know the telephone system is a bit different in India, so would the
X100P not be suitable?  Is there a change I need to make in the
Zaptel.conf or zapata.conf?

Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also
pretty frustrating...

Any help here would be appreciated.
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Re: [Asterisk-Users] Asterisk PBX Manager

[C F]

I didn't test it on a live system. Just on their demo but it looks very good.


On Tue, 1 Mar 2005 11:46:40 -0500, Michael Di Martino <[EMAIL PROTECTED]> wrote:
>  
>  
> 
> Does anyone on this list have any experience Thirdlane.com's Asterisk PBX
> Manager? 
> And if so what do you think of it? 
> 
> Regards, 
> Michael DiMartino 
> Director of MIS 
> The telx Group, Inc. 
> 17 State St, 33rd Floor 
> New York, NY 10004 
> T: 212.480.3300 X2022 
> C: 646.207.6603 
>   
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[Asterisk-Users] Asterisk PBX Manager

[Michael Di Martino]

Title: Asterisk PBX Manager







Does anyone on this list have any experience Thirdlane.com's Asterisk PBX Manager?

And if so what do you think of it?


Regards,

Michael DiMartino

Director of MIS

The telx Group, Inc.

17 State St, 33rd Floor

New York, NY 10004

T: 212.480.3300 X2022

C: 646.207.6603

 



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[Asterisk-Users] Asterisk Pbx Manager Equivalent (in plain text - apologies to those that dont like HTML mail!!)

[Paul Brock]

http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from
Thirdlane) looks like a great program for "eye candy configuration" of
Asterisk.

However it costs lost of $, and I'm currently only an "experimenter" so to
speak.

Anyone advice of a decent alternative that is similar?? Currently, we only
have VOIP connections, but will have a couple of Digium fxs/fxo's soon to
have a play with, so would be advantageous if it worked with these too.

Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? 

Thx

Paul

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[Asterisk-Users] Asterisk Pbx Manager Equivalent

[Paul Brock]

http://www.thirdlane.com/screenshots.htm
(Asterisk PBX Manager from Thirdlane) looks like a 

great program for “eye candy configuration” of
Asterisk.

 

However it costs lost of $, and I’m currently only an “experimenter”
so to speak.

 

Anyone advice of a decent alternative that is similar??
Currently, we only have VOIP connections, 

but will have a couple of Digium fxs/fxo’s soon to
have a play with, so would be advantageous if it 

worked with these too…

 

Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? 

 

Thx

 

Paul 






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Re: [Asterisk-Users] Asterisk PBX Manager

[Gregory Junker]

Not yet. It's under development.
Greg
Alex Brecher wrote:
Is there anything open source out there that has the same or better feature
set than Asterisk PBX Manager ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter
Sent: Tuesday, November 30, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PBX Manager
I have this, it comes as a webmin module. I also got it with the intention
of bundling it for clients.
It costs $300  , and the license is tied to the NIC.
While it wont do EVERYTHING, it will probably be sufficient for the user
to set up extensions/phones/menus/voicemail/conferences. One thing that I
am not happy with, is that it allows raw editing of the conf files. Gawd
help us if a user gets into that lot.
I emailed Third lane, and they replied staright away with an address where
I could download an evaluation. I'd publish the url here, but there must
be a reason why they don't show it on their web site.
Oh, and by the way (this from a beginner), I found it by searching on the
WIKI
Clive
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RE: [Asterisk-Users] Asterisk PBX Manager

[dean collins]

AMP but you already knew that.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Brecher
Sent: Wednesday, December 01, 2004 6:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk PBX Manager

Is there anything open source out there that has the same or better
feature
set than Asterisk PBX Manager ?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clive
Carter
Sent: Tuesday, November 30, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PBX Manager

I have this, it comes as a webmin module. I also got it with the
intention
of bundling it for clients.
It costs $300  , and the license is tied to the NIC.
While it wont do EVERYTHING, it will probably be sufficient for the user
to set up extensions/phones/menus/voicemail/conferences. One thing that
I
am not happy with, is that it allows raw editing of the conf files. Gawd
help us if a user gets into that lot.
I emailed Third lane, and they replied staright away with an address
where
I could download an evaluation. I'd publish the url here, but there must
be a reason why they don't show it on their web site.
Oh, and by the way (this from a beginner), I found it by searching on
the
WIKI

Clive

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RE: [Asterisk-Users] Asterisk PBX Manager

[Alex Brecher]

Is there anything open source out there that has the same or better feature
set than Asterisk PBX Manager ?


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter
Sent: Tuesday, November 30, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PBX Manager

I have this, it comes as a webmin module. I also got it with the intention
of bundling it for clients.
It costs $300  , and the license is tied to the NIC.
While it wont do EVERYTHING, it will probably be sufficient for the user
to set up extensions/phones/menus/voicemail/conferences. One thing that I
am not happy with, is that it allows raw editing of the conf files. Gawd
help us if a user gets into that lot.
I emailed Third lane, and they replied staright away with an address where
I could download an evaluation. I'd publish the url here, but there must
be a reason why they don't show it on their web site.
Oh, and by the way (this from a beginner), I found it by searching on the
WIKI

Clive

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[Asterisk-Users] Asterisk PBX Manager

[Clive Carter]

I have this, it comes as a webmin module. I also got it with the intention
of bundling it for clients.
It costs $300  , and the license is tied to the NIC.
While it wont do EVERYTHING, it will probably be sufficient for the user
to set up extensions/phones/menus/voicemail/conferences. One thing that I
am not happy with, is that it allows raw editing of the conf files. Gawd
help us if a user gets into that lot.
I emailed Third lane, and they replied staright away with an address where
I could download an evaluation. I'd publish the url here, but there must
be a reason why they don't show it on their web site.
Oh, and by the way (this from a beginner), I found it by searching on the
WIKI

Clive

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Re: [Asterisk-Users] Asterisk PBX Manager

[Robert Lawrence]

I have tried to email them multiple times to get more information, but 
they have not responded to my requests.  I do know that it is NOT GPLed 
and it is commercial software.I have emailed ThirdLane to ask about 
a demo and costs, as I am looking for something to bundle for my 
customers to use.

Robert
Darren Bentley wrote:
Hi,
I haven't seen any mention of this on the list.
I'm curious if anyone has tried it and can share some opinions on it?
http://www.thirdlane.com/screenshots.htm
http://www.thirdlane.com/opensource.htm#manager
Defaults Manager - initial PBX configuration
Device Manager - management of devices (phones)
Mailbox Manager - configuration of user mailboxes
Extensions Manager - dialplan management and assignment of scripts to
extensions
Voice Menu Manager - configuration of Auto Attendant and multi level
voice menus
Script Manager - creation of scripts for call handling (used by
Extensions Manager)
Conference Manager - configuration of conference rooms
Configuration Editor - direct access to Asterisk configuration files  
Command Shell- web interface to Asterisk command line interface
File Manager - intelligent upload and download for various configuration
and support files

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RE: [Asterisk-Users] Asterisk PBX Manager

[dean collins]

I've heard it mentioned but I've never seen where to download it, and if
it isn't gpl'd then how much they want for it.


Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Bentley
Sent: Tuesday, November 30, 2004 2:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PBX Manager

Hi,

I haven't seen any mention of this on the list.

I'm curious if anyone has tried it and can share some opinions on it?

http://www.thirdlane.com/screenshots.htm
http://www.thirdlane.com/opensource.htm#manager

Defaults Manager - initial PBX configuration
Device Manager - management of devices (phones)
Mailbox Manager - configuration of user mailboxes
Extensions Manager - dialplan management and assignment of scripts to
extensions
Voice Menu Manager - configuration of Auto Attendant and multi level
voice menus
Script Manager - creation of scripts for call handling (used by
Extensions Manager)
Conference Manager - configuration of conference rooms
Configuration Editor - direct access to Asterisk configuration files  
Command Shell- web interface to Asterisk command line interface
File Manager - intelligent upload and download for various configuration
and support files

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[Asterisk-Users] Asterisk PBX Manager

[Darren Bentley]

Hi,

I haven't seen any mention of this on the list.

I'm curious if anyone has tried it and can share some opinions on it?

http://www.thirdlane.com/screenshots.htm
http://www.thirdlane.com/opensource.htm#manager

Defaults Manager - initial PBX configuration
Device Manager - management of devices (phones)
Mailbox Manager - configuration of user mailboxes
Extensions Manager - dialplan management and assignment of scripts to
extensions
Voice Menu Manager - configuration of Auto Attendant and multi level
voice menus
Script Manager - creation of scripts for call handling (used by
Extensions Manager)
Conference Manager - configuration of conference rooms
Configuration Editor - direct access to Asterisk configuration files  
Command Shell- web interface to Asterisk command line interface
File Manager - intelligent upload and download for various configuration
and support files

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Re: [Asterisk-Users] Asterisk PBX and backup Circuits

[Rich Adamson]

> I am interested to know how one would calculate the amount of PSTN
> connection needed for backup on an Asterisk PBX that is being setup to
> receive its DIDs via a VoIP provide.  To sum up what I am
> implementing:  I am porting my DIDs to a VoIP provide so I will need a
> back up plan in place if the Data network fails.  In addition, 911
> will always be going out the PSTN so I know I need at least one POTs
> circuit.  Calls inbound and outbound will always routed through the
> data network.

In telco terms, you probably need to do a small Traffic Study; analyze
the existing traffic for maximum number of simultanous calls, etc.

If this is an existing business with an existing pbx, there are likely
some usage statistics available within the pbx. If that's not available,
"some" telephone companies will do the traffic study for you (don't
need to tell them why your doing it, but rather to determine the
number of telco lines needed for the business.) If that's not possible,
ask the telco to provide you with a list of "all calls with detail"
and run through the list to calculate the maximum number of
simultanous calls.

If this is a new installation with absolutely no history, your only
option is to guess at the maximum.


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[Asterisk-Users] Asterisk PBX and backup Circuits

[kurt x]

I am interested to know how one would calculate the amount of PSTN
connection needed for backup on an Asterisk PBX that is being setup to
receive its DIDs via a VoIP provide.  To sum up what I am
implementing:  I am porting my DIDs to a VoIP provide so I will need a
back up plan in place if the Data network fails.  In addition, 911
will always be going out the PSTN so I know I need at least one POTs
circuit.  Calls inbound and outbound will always routed through the
data network.

Thanks,

Kurt
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[Asterisk-Users] Asterisk PBX and backup Circuits

[Kurt Pasewaldt]

I am interested to know how one would calculate the amount of PSTN
connection needed for backup on an Asterisk PBX that is being setup to
receive its DIDs via a VoIP provide.  To sum up what I am
implementing:  I am porting my DIDs to a VoIP provide so I will need a
back up plan in place if the Data network fails.  In addition, 911
will always be going out the PSTN so I know I need at least one POTs
circuit.  Calls inbound and outbound will always routed through the
data network.

Thanks,

Kurt
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Karl Brose]

Craig Guy wrote:
cancallforward=yes
There is no such function in distributed  chan_sip.c,
ergo there can't be such a configuration parameter.
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Craig Guy]

Hmm,

Asterisk is certainly recognising the call forward.  I had this error when I
first tried it and resolved by adding the 'cancallforward=yes' in the
sip.conf for that extension.  I think you may actually have to restart
asterisk to enable these functions rather than reload?  I am also running
the August 8 CVS head rather than the 'stable' version of Asterisk and my
grandstream firmware is 1.0.5.10.

Craig


- Original Message - 
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, August 21, 2004 1:40 AM
Subject: RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone


Hi Craig,
Thank you very much for the helpful information. I did enable that setting
and it seems to have worked but not all the way.  I do a *72 for an
unconditional call forward + the number to forward to. Then when I dial the
grandstream that has it enabled, asterisk just reponds that the extension is
busy, the BT does not foward the call. I also get the following on the CLI

  -- Executing Dial("Zap/8-1", "SIP/2000|20") in new stack
-- Called 2000
-- Got SIP response 302 "Moved Temporarily" back from 64.201.13.50
-- SIP/2000-42e8 is busy


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig Guy
Sent: Friday, August 20, 2004 12:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone


Hi James,

This is a feature that needs to be enabled on both the phones and on
Asterisk.  So after enabling on your BT100 you need to add
'cancallforward=yes' to each extension in sip.conf you would like to add
this feature to as in :-

[9500]
context=internal
type=friend
username=9500
host=dynamic
callerid=9500
disallow=all
allow=ulaw
allow=alaw
dtmfmode=info
mailbox=9500
callgroup=1
pickupgroup=1
cancallforward=yes

Craig

- Original Message - 
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, August 21, 2004 12:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone


Hi All,

I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.

Thanks,

James
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Chris Shaw]

http://www.voip-info.org/wiki-asterisk+pbx+functions
http://www.voip-info.org/wiki-asterisk+vertical+service+activation+codes

- Original Message -
From: "Robert Rozman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 3:02 PM
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone


> Hi,
>
> sorry for interruption, but are there any guides for all possible Asterisk
> PBX functions that are available with no particular dialplan handling ?
>
> Thanks,
>
> Robert.
>
> - Original Message -
> From: "James Freire" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, August 20, 2004 6:09 PM
> Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone
>
>
> Hi All,
>
> I am using a Grandstream BT100 and I have been trying to get the PBX
> features to work for DND, call foward, etc. These functions do work when I
> use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
> functions (ie *78, *79) to work using my SIP phones. Is there a feature
that
> has to be enabled to do this? I know these functions are available within
> the GS phone but all of them seem to just show the phone as being busy,
even
> though, say, call foward is supposed to foward. It just makes the phone
> busy. I figure it would be easier just to have asterisk handling all those
> PBX functions.
>
> Thanks,
>
> James
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Robert Rozman]

Hi,

sorry for interruption, but are there any guides for all possible Asterisk
PBX functions that are available with no particular dialplan handling ?

Thanks,

Robert.

- Original Message - 
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 6:09 PM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone


Hi All,

I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.

Thanks,

James
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Chris Shaw]

- Original Message -
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 1:37 PM
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone


> Chris Shaw wrote:
>
> >>I am suprised that one would have to create a dialplan since its an
> >
> > already built in function that works with regular POTS phones. Or is it
> > because of the way DTMF is sent via SIP?
> >
> > I don't know digium's long range plans, but looking through chan_sip.c
NONE
> > of the vertical service codes are mentioned anywhere... A quick look
through
> > chan_zap reveals all of them... So for right now it's not implemented in
> > SIP...
>
> Well, here we stumble over the SIP religion again.
>
> First, a phone connected to an RJ11 jack in a Digium card is a stupid
phone. All
> the intelligence lies in the zaptel driver and asterisk.
>
> Most SIP phones are more clever (at least expected to be much more clever
than
> the GS :-).
>
> Look at the SIPURA, where you are able to implement vertical service codes
> in the SIPura. Asterisk should not bother with DND and forwards, the SIP
phone
> does. Just send the call to the phone. Some of these phones are complete
> Linux systems with IPsec, multiple lines and a lot of routing
intelligence.
>
> There's also a discussion between Asterisk developers on whether these
> codes should be fixed in the channel or in the dial plan. At least, they
> should be configurable since there's no global standard (again).
> Or there may be, but there are still differences between countries
> and providers...
>
> * Executive summary: SIP is designed for very intelligent end-points.
> * A PBX with analogue lines is designed for central intelligence.
> * Asterisk will always be in the middle of these kind of discussions,
>and it'll be fun each time we try to sort it out.
>
> /Olle

No, I agree completely with the way it works now, in fact I think it
SHOULDN'T be implemented in SIP myself... Doing it in the dialplan (if your
phone doesn't support it) works fine and doesn't break anything (that's the
key right there). We need some more docs on how to do different things and
I'm sure many people could contribute those, myself included... Some already
have...

The only thing is, if any of the apps you've written in your dialplan become
obsoleted or change syntax, your whole implementation will get screwed
over... I guess that's true with anything though...

-Chris

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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Olle E. Johansson]

Chris Shaw wrote:
I am suprised that one would have to create a dialplan since its an
already built in function that works with regular POTS phones. Or is it
because of the way DTMF is sent via SIP?
I don't know digium's long range plans, but looking through chan_sip.c NONE
of the vertical service codes are mentioned anywhere... A quick look through
chan_zap reveals all of them... So for right now it's not implemented in
SIP...
Well, here we stumble over the SIP religion again.
First, a phone connected to an RJ11 jack in a Digium card is a stupid phone. All
the intelligence lies in the zaptel driver and asterisk.
Most SIP phones are more clever (at least expected to be much more clever than
the GS :-).
Look at the SIPURA, where you are able to implement vertical service codes
in the SIPura. Asterisk should not bother with DND and forwards, the SIP phone
does. Just send the call to the phone. Some of these phones are complete
Linux systems with IPsec, multiple lines and a lot of routing intelligence.
There's also a discussion between Asterisk developers on whether these
codes should be fixed in the channel or in the dial plan. At least, they
should be configurable since there's no global standard (again).
Or there may be, but there are still differences between countries
and providers...
* Executive summary: SIP is designed for very intelligent end-points.
* A PBX with analogue lines is designed for central intelligence.
* Asterisk will always be in the middle of these kind of discussions,
  and it'll be fun each time we try to sort it out.
/Olle
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Chris Shaw]

> I am suprised that one would have to create a dialplan since its an
already built in function that works with regular POTS phones. Or is it
because of the way DTMF is sent via SIP?

I don't know digium's long range plans, but looking through chan_sip.c NONE
of the vertical service codes are mentioned anywhere... A quick look through
chan_zap reveals all of them... So for right now it's not implemented in
SIP...

-Chris

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RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[James Freire]

I am suprised that one would have to create a dialplan since its an already built in 
function that works with regular POTS phones. Or is it because of the way DTMF is sent 
via SIP?

> Someone correct me if I'm wrong but I believe you'll need the dialplan for
> this one...
> 
> What I envision is doing something like this...
> 
> [verticalservice]
> 
> exten => *78,1,DbGet(${dnd}=features/dnd)
> exten => *78,2,DbPut(features/dnd=1)
> exten => *78,3,Playback(pbx-dndenabled)
> exten => *78,4,Hangup()
> exten => *78,102,GotoIf($[${dnd} = '0')]?103:104)
> exteh => *78,103,DbPut(features/dnd=1)
> exten => *78,104,Playback(pbx-dndenabled)
> exten => *78,105,Hangup()
> 
> exten => *79 ... etc...

Wouldn't you need to track each extension? something like:
exten => *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten => *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten => *78,3,Playback(pbx-dndenabled)
exten => *78,4,Hangup()
etc.?

The wiki has an exmple for call forwarding:
http://www.voip-info.org/wiki-Asterisk+call+forwarding

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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Chris Shaw]

> Wouldn't you need to track each extension? something like:
> exten => *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
> exten => *78,2,DbPut(dnd/${CALLERIDNUM}=1)
> exten => *78,3,Playback(pbx-dndenabled)
> exten => *78,4,Hangup()
> etc.?
>
Yep! good catch! that's why I asked someone to correct me, I was in a hurry
and this was an on-the-fly kind of example...

You would need to do something like this, or make a key like
features/dnd-${CALLERIDNUM} would be best... Would also work for
forwarding...

-Chris

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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Walt Reed]

On Fri, Aug 20, 2004 at 10:13:16AM -0700, Chris Shaw said:
> - Original Message -
> From: "James Freire" <[EMAIL PROTECTED]>
> 
> > I am using a Grandstream BT100 and I have been trying to get the PBX
> features to work for DND, call foward, etc. These functions do work when I
> use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
> functions (ie *78, *79) to work using my SIP phones. Is there a feature that
> has to be enabled to do this? I know these functions are available within
> the GS phone but all of them seem to just show the phone as being busy, even
> though, say, call foward is supposed to foward. It just makes the phone
> busy. I figure it would be easier just to have asterisk handling all those
> PBX functions.
> >
> > James
> 
> Someone correct me if I'm wrong but I believe you'll need the dialplan for
> this one...
> 
> What I envision is doing something like this...
> 
> [verticalservice]
> 
> exten => *78,1,DbGet(${dnd}=features/dnd)
> exten => *78,2,DbPut(features/dnd=1)
> exten => *78,3,Playback(pbx-dndenabled)
> exten => *78,4,Hangup()
> exten => *78,102,GotoIf($[${dnd} = '0')]?103:104)
> exteh => *78,103,DbPut(features/dnd=1)
> exten => *78,104,Playback(pbx-dndenabled)
> exten => *78,105,Hangup()
> 
> exten => *79 ... etc...

Wouldn't you need to track each extension? something like:
exten => *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten => *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten => *78,3,Playback(pbx-dndenabled)
exten => *78,4,Hangup()
etc.?

The wiki has an exmple for call forwarding:
http://www.voip-info.org/wiki-Asterisk+call+forwarding

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RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[James Freire]

Hi Craig, 
Thank you very much for the helpful information. I did enable that setting and it 
seems to have worked but not all the way.  I do a *72 for an unconditional call 
forward + the number to forward to. Then when I dial the grandstream that has it 
enabled, asterisk just reponds that the extension is busy, the BT does not foward the 
call. I also get the following on the CLI

  -- Executing Dial("Zap/8-1", "SIP/2000|20") in new stack
-- Called 2000
-- Got SIP response 302 "Moved Temporarily" back from 64.201.13.50
-- SIP/2000-42e8 is busy


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig Guy
Sent: Friday, August 20, 2004 12:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone


Hi James,

This is a feature that needs to be enabled on both the phones and on
Asterisk.  So after enabling on your BT100 you need to add
'cancallforward=yes' to each extension in sip.conf you would like to add
this feature to as in :-

[9500]
context=internal
type=friend
username=9500
host=dynamic
callerid=9500
disallow=all
allow=ulaw
allow=alaw
dtmfmode=info
mailbox=9500
callgroup=1
pickupgroup=1
cancallforward=yes

Craig

- Original Message - 
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, August 21, 2004 12:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone


Hi All,

I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.

Thanks,

James
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Chris Shaw]

- Original Message -
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 9:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone


> Hi All,
>
> I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.
>
> Thanks,
>
> James

Someone correct me if I'm wrong but I believe you'll need the dialplan for
this one...

What I envision is doing something like this...

[verticalservice]

exten => *78,1,DbGet(${dnd}=features/dnd)
exten => *78,2,DbPut(features/dnd=1)
exten => *78,3,Playback(pbx-dndenabled)
exten => *78,4,Hangup()
exten => *78,102,GotoIf($[${dnd} = '0')]?103:104)
exteh => *78,103,DbPut(features/dnd=1)
exten => *78,104,Playback(pbx-dndenabled)
exten => *78,105,Hangup()

exten => *79 ... etc...


Then in your extension calling macro, you're going to want to check against
the DB like this...

[macro-insidedial]

exten => s,1,DbGet(${dnd}=features/dnd)
exten => s,2,DbGet(${fw}=features/fw)
exten => s,3,Dial(${ARG1},25,tT)
exten => s,4,VoiceMail(u${ARG1})
exten => s,5,Hangup()
exten => s,102,GotoIf($[${dnd} = '1']?200:2)
exten => s,103,GotoIf($[${fw} = '1']?300:3)
exten => s,104,VoiceMail(b${ARG1})

exten => s,200,VoiceMail(b${ARG1})
exten => s,201,Hangup()

exten => s,300,Dial(SIP/[EMAIL PROTECTED],60)
exten => s,301,Congestion()

be sure to include [verticalservice] in your inside-office context...

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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

[Craig Guy]

Hi James,

This is a feature that needs to be enabled on both the phones and on
Asterisk.  So after enabling on your BT100 you need to add
'cancallforward=yes' to each extension in sip.conf you would like to add
this feature to as in :-

[9500]
context=internal
type=friend
username=9500
host=dynamic
callerid=9500
disallow=all
allow=ulaw
allow=alaw
dtmfmode=info
mailbox=9500
callgroup=1
pickupgroup=1
cancallforward=yes

Craig

- Original Message - 
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, August 21, 2004 12:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone


Hi All,

I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.

Thanks,

James
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[Asterisk-Users] Asterisk PBX Functions via SIP phone

[James Freire]

Hi All,

I am using a Grandstream BT100 and I have been trying to get the PBX features to work 
for DND, call foward, etc. These functions do work when I use my POTS phones hooked up 
to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP 
phones. Is there a feature that has to be enabled to do this? I know these functions 
are available within the GS phone but all of them seem to just show the phone as being 
busy, even though, say, call foward is supposed to foward. It just makes the phone 
busy. I figure it would be easier just to have asterisk handling all those PBX 
functions.

Thanks,

James
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RE: [Asterisk-Users] Asterisk, PBX, VoIP and PRI

[Peter Svensson]

On Wed, 28 Jul 2004, Chris Johnson wrote:

> On Wed, 28 Jul 2004, Chris Johnson wrote:
> 
> >Why not plug the PRI into a TE410P in the asterisk box and handle both the 
> >ppp and the voice via asterisk? 
> 
> If it'll work, sounds great!!  Anyone doing this?

Sorry, there are no analog softmodem drivers yet. I have been living in
isdn land for too long and forgot about the analog modems. There are 
indeed drivers for ppp over isdn direct but no softmodem for analog calls.

We have solved this by routing those calls to our old pbx (over an E1 
PRI) from which a couple of BRI:s go to isdn modems with analog 
capabilities.

So, you should be able to hook up your old PRI equipment to Asterisk, i.e.

 PSTN  -PRI-  Asterisk  -PRI-  Old_equipment
 \
  ---lan--- sip stuff

This is sort of what we are doing. As far as I can tell the digital 
channels are passed transparently from one pri to the other once a call is 
set up. We can do both analog termination and direct isdn connections. 

Peter




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Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI

[Jim Lund]

Can the TE410P  handle the ppp ( analog 56K inboud modem calls ( 
be a modem pool server )) without any extra hardware?

At 04:07 PM 07/28/2004, you wrote:
On Wed, 28 Jul 2004, Chris Johnson wrote:
> 1)   Convert one of our PRIs into a two way voice/data bundle.
> 2)   Plug this PRI into a piece of equipment (AS5300?) which can
> terminate both PPP, *AND* feed our Asterisk system.  We'd transfer our
> POTS service over to the PRI and pick up the SIP signaling there.
> 3)   In the extended model, use this setup to resell VoIP to our
> business ISP clients.
>
> Has anyone out there done this?  If so, is the AS5300 the right choice?
> What caveats will I face?
Why not plug the PRI into a TE410P in the asterisk box and handle both the
ppp and the voice via asterisk?
I am not sure what you mean by "Convert one of our PRIs into a two way
voice/data bundle". Isn't that sort of what a PRI is anyway?
Peter

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=
   Jim Lund  Fox Business Systems KansasNet
[EMAIL PROTECTED] 531 Ft. Riley Blvd
(785)776-1452Manhattan, KS  66502 http://www.kansas.net 

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Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI

[Scott Laird]

On Jul 28, 2004, at 2:07 PM, Peter Svensson wrote:
On Wed, 28 Jul 2004, Chris Johnson wrote:
1)   Convert one of our PRIs into a two way voice/data bundle.
2)   Plug this PRI into a piece of equipment (AS5300?) which can
terminate both PPP, *AND* feed our Asterisk system.  We'd transfer our
POTS service over to the PRI and pick up the SIP signaling there.
3)   In the extended model, use this setup to resell VoIP to our
business ISP clients.
Has anyone out there done this?  If so, is the AS5300 the right 
choice?
What caveats will I face?
Why not plug the PRI into a TE410P in the asterisk box and handle both 
the
ppp and the voice via asterisk?
He's (presumably) talking about PPP over a 56k modem, not PPP to a ISDN 
BRI.  Asterisk can't do an internal 56k modem, at least not yet.  I 
thought that someone said that you need the AS5350 to do combined voice 
and data, but I might be mistaken.  It's been years since I've dealt 
with dialup.

Scott
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RE: [Asterisk-Users] Asterisk, PBX, VoIP and PRI

[Chris Johnson]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Wednesday, July 28, 2004 5:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI

On Wed, 28 Jul 2004, Chris Johnson wrote:

> 1)   Convert one of our PRIs into a two way voice/data bundle.
> 2)   Plug this PRI into a piece of equipment (AS5300?) which can
> terminate both PPP, *AND* feed our Asterisk system.  We'd transfer our
> POTS service over to the PRI and pick up the SIP signaling there.
> 3)   In the extended model, use this setup to resell VoIP to our
> business ISP clients.
>  
> Has anyone out there done this?  If so, is the AS5300 the right
choice?
> What caveats will I face?

>Why not plug the PRI into a TE410P in the asterisk box and handle both
the 
>ppp and the voice via asterisk? 

If it'll work, sounds great!!  Anyone doing this?

>>I am not sure what you mean by "Convert one of our PRIs into a two way

>>voice/data bundle". Isn't that sort of what a PRI is anyway?

Yes, but most ISPs use inward only PRI.

Peter



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RE: [Asterisk-Users] Asterisk, PBX, VoIP and PRI

[Andrew Brown]

 

  This is similar what we are looking to do with our 
  customers,  ISP providing VoIP with our * system to broadband 
  customers. 
   
  Having trouble getting the Mysql to work 
  on RedHat 9.0 atm for the billing. but would be interested in 
  talking more and sharing experiences.
   
   
  
  Hi everyone, I’m new to the list, 
  so I apologize if this is a trite question.  We’re an ISP with a barely used PRI, 
  and also currently use asterisk with POTS service on an audiocodes gateway.  
  We have a notion that we want to take on a new revenue stream, with our 
  model being as such:
   
  1)   
  Convert one of our PRIs into a two way voice/data 
  bundle.
  2)   
  Plug this PRI into a piece of 
  equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system.  We’d transfer our POTS service over to 
  the PRI and pick up the SIP signaling there.
  3)   
  In the extended model, use this 
  setup to resell VoIP to our business ISP 
  clients.
   
  Has anyone out there done 
  this?  If so, is the AS5300 the 
  right choice?  What caveats will I 
  face?

Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI

[Peter Svensson]

On Wed, 28 Jul 2004, Chris Johnson wrote:

> 1)   Convert one of our PRIs into a two way voice/data bundle.
> 2)   Plug this PRI into a piece of equipment (AS5300?) which can
> terminate both PPP, *AND* feed our Asterisk system.  We'd transfer our
> POTS service over to the PRI and pick up the SIP signaling there.
> 3)   In the extended model, use this setup to resell VoIP to our
> business ISP clients.
>  
> Has anyone out there done this?  If so, is the AS5300 the right choice?
> What caveats will I face?

Why not plug the PRI into a TE410P in the asterisk box and handle both the 
ppp and the voice via asterisk? 

I am not sure what you mean by "Convert one of our PRIs into a two way 
voice/data bundle". Isn't that sort of what a PRI is anyway?

Peter



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[Asterisk-Users] Asterisk, PBX, VoIP and PRI

[Chris Johnson]

Hi everyone, I’m new to the list, so I apologize if this is
a trite question.  We’re an ISP with a barely
used PRI, and also currently use asterisk with POTS service on an audiocodes gateway. 
We have a notion that we want to take on a new revenue stream, with our
model being as such:

 

1)   Convert one
of our PRIs into a two way voice/data bundle.

2)   Plug this
PRI into a piece of equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system.  We’d transfer our POTS service over to the
PRI and pick up the SIP signaling there.

3)   In the
extended model, use this setup to resell VoIP to our
business ISP clients.

 

Has anyone out there done this?  If so, is the AS5300 the right choice?  What caveats will I face?

Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb

[Michael Graves]

Interestingly enough, I just tried this on a SIP extension and it
worked fine...although I did not see a report on the command line as
expected. I used a Polycom IP600 to call an extension which is an
analog phone plugged into one side of a Spiura SPA-2000. Keying *78
into the phone one the SPA generated a pulsing dialtone for 3 seconds
then the dialtone returned to normal. I hungup then dialed that
extension again and was routed directly to voicemail.

I do have an X100p in my * server and zaptel is loaded. However, I
don't use it as my fxo anymore, I just left it there a timing source
for the conferences.

Michael


On Wed, 23 Jun 2004 10:08:46 +0200, Stephen Rosebush wrote:

>Yes this is where I wanted to use it the most, I use Asterisk as a 
>software-only based PBX using SIP and IAX, I have an ATA device on my 
>network connected to the PBX but it nor the actual analog phone has a 
>DND function. I am hoping to implement an Asterisk-side based DND 
>somehow but was wondering where to go from there.
>
>Thanks
>
>Steve
>
>Dean Collins wrote:
>
>>That could explain why it wouldn't work on any of my sip extensions I
>>tried it on this morning when I first read about it and thought cool the
>>things you learn.
>>
>>Is there anyway to make it work on Sip extensions?
>>
>>Cheers,
>>Dean
>>
>>
>>
>>-Original Message-
>>From: [EMAIL PROTECTED]
>>[mailto:[EMAIL PROTECTED] On Behalf Of Aaron J.
>>Angel
>>Sent: Wednesday, 23 June 2004 10:33 AM
>>To: [EMAIL PROTECTED]
>>Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
>>
>>john lawler [EMAIL PROTECTED] wrote:
>>  
>>
>>>You don't have to put this in the dialplan.  It's one of
>>>those low-level functions in Asterisk (possibly controlled at
>>>the driver level-- I'm not sure about that).  If you have an
>>>extension defined, pick up the handset and dial '*78', you
>>>should see on the Asterisk CLI:
>>>
>>> Enabled DND on channel 
>>>
>>>
>>
>>That assumes your using zaptel, no?  This doesn't exist for other
>>channels
>>as it's not built in to the channel driver for anything but zaptel.
>>
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>>  
>>
>
>
>-- 
>Stephen Rosebush,
>[EMAIL PROTECTED]
>http://www.desynched.org/
>
>// PSTN
>USA:   1-248-724-4452  x201
>Netherlands:   +31-(0)20-6598858 x63420 x201
>
>// IP Phone
>FWD:   63420 x201
>IAXTEL:1-700-356-6191 x201
>SIP:   sip:[EMAIL PROTECTED]
>
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--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

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

"One day in your life shouldn't be a problem" 
- 54-40 from One Day
 
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Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb

[Stephen Rosebush]

Yes this is where I wanted to use it the most, I use Asterisk as a 
software-only based PBX using SIP and IAX, I have an ATA device on my 
network connected to the PBX but it nor the actual analog phone has a 
DND function. I am hoping to implement an Asterisk-side based DND 
somehow but was wondering where to go from there.

Thanks
Steve
Dean Collins wrote:
That could explain why it wouldn't work on any of my sip extensions I
tried it on this morning when I first read about it and thought cool the
things you learn.
Is there anyway to make it work on Sip extensions?
Cheers,
Dean

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron J.
Angel
Sent: Wednesday, 23 June 2004 10:33 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
john lawler [EMAIL PROTECTED] wrote:
 

You don't have to put this in the dialplan.  It's one of
those low-level functions in Asterisk (possibly controlled at
the driver level-- I'm not sure about that).  If you have an
extension defined, pick up the handset and dial '*78', you
should see on the Asterisk CLI:
	Enabled DND on channel 
   

That assumes your using zaptel, no?  This doesn't exist for other
channels
as it's not built in to the channel driver for anything but zaptel.
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--
Stephen Rosebush,
[EMAIL PROTECTED]
http://www.desynched.org/
// PSTN
USA:1-248-724-4452  x201
Netherlands:+31-(0)20-6598858 x63420 x201
// IP Phone
FWD:63420 x201
IAXTEL: 1-700-356-6191 x201
SIP:sip:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb

[Dean Collins]

That could explain why it wouldn't work on any of my sip extensions I
tried it on this morning when I first read about it and thought cool the
things you learn.

Is there anyway to make it work on Sip extensions?

Cheers,
Dean



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron J.
Angel
Sent: Wednesday, 23 June 2004 10:33 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb

john lawler [EMAIL PROTECTED] wrote:
> You don't have to put this in the dialplan.  It's one of
> those low-level functions in Asterisk (possibly controlled at
> the driver level-- I'm not sure about that).  If you have an
> extension defined, pick up the handset and dial '*78', you
> should see on the Asterisk CLI:
> 
>   Enabled DND on channel 

That assumes your using zaptel, no?  This doesn't exist for other
channels
as it's not built in to the channel driver for anything but zaptel.

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RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb

[Aaron J. Angel]

john lawler [EMAIL PROTECTED] wrote:
> You don't have to put this in the dialplan.  It's one of
> those low-level functions in Asterisk (possibly controlled at
> the driver level-- I'm not sure about that).  If you have an
> extension defined, pick up the handset and dial '*78', you
> should see on the Asterisk CLI:
> 
>   Enabled DND on channel 

That assumes your using zaptel, no?  This doesn't exist for other channels
as it's not built in to the channel driver for anything but zaptel.

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Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb

[john lawler]

You don't have to put this in the dialplan.  It's one of those low-level
functions in Asterisk (possibly controlled at the driver level-- I'm not
sure about that).  If you have an extension defined, pick up the handset
and dial '*78', you should see on the Asterisk CLI:
Enabled DND on channel 
And then if someone tries to call that channel, they'd get handled as if 
the person had the phone off the hook, even if it weren't.  It's that 
simple.  To turn it back off, you use '*79'.

jl
p.s.  I think I actually added that info that you quoted from voip-info. 
 If you think it's unclear, grab an account and reword it!

Stephen Rosebush wrote:
I quote from the wiki on voip-info:
In Asterisk, DND is controlled by dialing:
  *78 to turn *on* Do Not Disturb mode and
  *79 to turn *off* Do Not Disturb mode
How would one implement this?? I've seen dialplans that created these
extensions and you would need to use the Asterisk DB to set the DND
flag on or off.. I am still pretty new and I would like to know how to
implement this and maybe some other 'star' services?
Thanks a bunch guys!

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[Asterisk-Users] Asterisk -- PBX Do Not Disturb

[Stephen Rosebush]

I quote from the wiki on voip-info:
In Asterisk, DND is controlled by dialing:
  *78 to turn *on* Do Not Disturb mode and
  *79 to turn *off* Do Not Disturb mode
How would one implement this?? I've seen dialplans that created these
extensions and you would need to use the Asterisk DB to set the DND
flag on or off.. I am still pretty new and I would like to know how to
implement this and maybe some other 'star' services?
Thanks a bunch guys!
--
Stephen Rosebush,
[EMAIL PROTECTED]
http://www.desynched.org/
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[Asterisk-Users] Asterisk PBX -> RT Integration

[Michael]

Greetings,

I had been working on Asterisk (http://asteriskpbx.org) about 2 years ago .

http://www.marko.net/asterisk/archives/0210/0107.html

Last night with the help of Jesse's rt-soap-server.pl (and some prodding) 
I implemented a much cleaner, more repeatable * -> RT phone gateway with some notes:

http://megaglobal.net/docs/asterisk/html/rtasterisk.html

Questions or comments appreciated.  


--Michael.



--
. Michael Jastremski ..
.. Network Engineer > Megaglobal Networks > Megaglobal.net 
.. Photographer > Open Photo Project  > Openphoto.net .
.. Resident > West Philadelphia   > Westphila.net .
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RE: Re[4]: [Asterisk-Users] Asterisk PBX Billing

[James Golovich]

The mysql schema is available in the doc/cdr_mysql.txt file (from the
asterisk source dir)

James

On Thu, 3 Jul 2003, Kim C. Callis wrote:

> You can find the comma delimited file at /var/log/asterisk/cdr-csv or if
> you are looking to do some easy querying on a database, you need to
> create a schema that I am sure someone on the channel has defined
> somewhere. At that point you clean up the /etc/asterisk/cdr_mysql.conf
> file to point to the appropriate database and authentication
> information.
> 
> Kim C. Callis
> 
> -Original Message-
> From: Angelo Sampietro [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, July 02, 2003 8:07 AM
> To: Kim C. Callis
> Cc: [EMAIL PROTECTED]
> Subject: Re[4]: [Asterisk-Users] Asterisk PBX Billing
> 
> thanks a lot!
> can you tell me where can i find more info about the CDR?
> probably this will be the better way to give to the company a summary
> with all the phone traffic :)
> 
> Angelo
> 
> 
> 
> Thursday, July 3, 2003, 4:37:32 PM, you wrote:
> 
> KCC> There is a CDR (Call Detail Record) which is accessible in two
> different
> KCC> ways. The first is via a simple comma delimited file which can be
> parsed
> KCC> and fed into whatever database that you want. The second way is to
> dump
> KCC> the CDR directly into MySQL, and extract accordingly. So the only
> trick
> KCC> there is to create a database for billing and create a relationship
> that
> KCC> will extract from the CDR database.
> 
> KCC> Kim C. Callis
> 
> KCC> -Original Message-
> KCC> From: [EMAIL PROTECTED]
> KCC> [mailto:[EMAIL PROTECTED] On Behalf Of Angelo
> KCC> Sampietro
> KCC> Sent: Wednesday, July 02, 2003 7:06 AM
> KCC> To: Scott Stingel
> KCC> Cc: [EMAIL PROTECTED]
> KCC> Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing
> 
> KCC> i think that the problem could be something more easy:
> 
> KCC> it is possible inside asterisk to log all che calls of all the
> users
> KCC> and know the timing and the number called for each call?
> KCC> if it is possible to do that, could be possible to make a program
> KCC> that takes this files and generate the costs reading the log
> KCC> informations...
> 
> KCC> so for me the real question is: there is a log of all the phone
> call
> KCC> that are made by asterisk?
> 
> KCC> Angelo
> 
> 
> 
> KCC> Wednesday, July 2, 2003, 3:28:22 PM, you wrote:
> 
> SS>> Shepherd-
> 
> SS>> Having designed one of these in the past (in a higher level voice
> SS>> environment), I can tell you that this is not a small undertaking.
> KCC> It's at
> SS>> least as much an SQL job as a voice task.
> 
> SS>> Usually the way to accomplish this is to establish more-or-less a
> KCC> pre-paid
> SS>> phone card system, where the shop prepays an overall amount for
> SS>> international calling access.  Then you have to time each call as
> it
> KCC> is
> SS>> occurring, debiting each account, and the master account, in
> KCC> real-time. This
> SS>> can be a bit complex when you have 20 or 30 calls going at one
> time.
> KCC> You
> SS>> have to cut them off promptly when the money runs out (big
> problem).
> KCC> And
> SS>> you have to provide call detail and charges to them at the end of
> KCC> each call,
> SS>> using their own retail tariff.
> 
> SS>> To add to the complexity, each country has a different tariff from
> KCC> the long
> SS>> distance carrier, and within the country, major cities often have
> KCC> special
> SS>> rates per minute.  Mobiles have a different tariff too.  Phone card
> SS>> platforms usually include a least-cost routing system which chooses
> KCC> a
> SS>> carrier real time based on the call.  Tariffs change weekly and
> must
> KCC> be
> SS>> updated in the system.
> 
> SS>> Anyway, I'm just scratching the surface!  I'll write more when I
> KCC> can!
> 
> SS>> Cheers
> SS>> Scott Stingel
> 
> 
> SS>> Scott M. Stingel 
> SS>> Emerging Voice Technology Inc.
> 
> SS>> Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
> SS>> URL:www.evtmedia.com <http://www.evtmedia.com>   
> 
> 
> 
> >>> -Original Message-
> >>> From: [EMAIL PROTECTED] 
> >>> [mailto:[EMAIL PROTECTED] On Behalf Of 
> >>> shepherd fungayi
> >>> Sent: Wednesday, July 02, 2003 10:49 AM
> >>> To: [EMAIL PROTECTED]
> &

RE: Re[4]: [Asterisk-Users] Asterisk PBX Billing

[Kim C. Callis]

You can find the comma delimited file at /var/log/asterisk/cdr-csv or if
you are looking to do some easy querying on a database, you need to
create a schema that I am sure someone on the channel has defined
somewhere. At that point you clean up the /etc/asterisk/cdr_mysql.conf
file to point to the appropriate database and authentication
information.

Kim C. Callis

-Original Message-
From: Angelo Sampietro [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, July 02, 2003 8:07 AM
To: Kim C. Callis
Cc: [EMAIL PROTECTED]
Subject: Re[4]: [Asterisk-Users] Asterisk PBX Billing

thanks a lot!
can you tell me where can i find more info about the CDR?
probably this will be the better way to give to the company a summary
with all the phone traffic :)

Angelo



Thursday, July 3, 2003, 4:37:32 PM, you wrote:

KCC> There is a CDR (Call Detail Record) which is accessible in two
different
KCC> ways. The first is via a simple comma delimited file which can be
parsed
KCC> and fed into whatever database that you want. The second way is to
dump
KCC> the CDR directly into MySQL, and extract accordingly. So the only
trick
KCC> there is to create a database for billing and create a relationship
that
KCC> will extract from the CDR database.

KCC> Kim C. Callis

KCC> -Original Message-
KCC> From: [EMAIL PROTECTED]
KCC> [mailto:[EMAIL PROTECTED] On Behalf Of Angelo
KCC> Sampietro
KCC> Sent: Wednesday, July 02, 2003 7:06 AM
KCC> To: Scott Stingel
KCC> Cc: [EMAIL PROTECTED]
KCC> Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing

KCC> i think that the problem could be something more easy:

KCC> it is possible inside asterisk to log all che calls of all the
users
KCC> and know the timing and the number called for each call?
KCC> if it is possible to do that, could be possible to make a program
KCC> that takes this files and generate the costs reading the log
KCC> informations...

KCC> so for me the real question is: there is a log of all the phone
call
KCC> that are made by asterisk?

KCC> Angelo



KCC> Wednesday, July 2, 2003, 3:28:22 PM, you wrote:

SS>> Shepherd-

SS>> Having designed one of these in the past (in a higher level voice
SS>> environment), I can tell you that this is not a small undertaking.
KCC> It's at
SS>> least as much an SQL job as a voice task.

SS>> Usually the way to accomplish this is to establish more-or-less a
KCC> pre-paid
SS>> phone card system, where the shop prepays an overall amount for
SS>> international calling access.  Then you have to time each call as
it
KCC> is
SS>> occurring, debiting each account, and the master account, in
KCC> real-time. This
SS>> can be a bit complex when you have 20 or 30 calls going at one
time.
KCC> You
SS>> have to cut them off promptly when the money runs out (big
problem).
KCC> And
SS>> you have to provide call detail and charges to them at the end of
KCC> each call,
SS>> using their own retail tariff.

SS>> To add to the complexity, each country has a different tariff from
KCC> the long
SS>> distance carrier, and within the country, major cities often have
KCC> special
SS>> rates per minute.  Mobiles have a different tariff too.  Phone card
SS>> platforms usually include a least-cost routing system which chooses
KCC> a
SS>> carrier real time based on the call.  Tariffs change weekly and
must
KCC> be
SS>> updated in the system.

SS>> Anyway, I'm just scratching the surface!  I'll write more when I
KCC> can!

SS>> Cheers
SS>> Scott Stingel


SS>> Scott M. Stingel 
SS>> Emerging Voice Technology Inc.

SS>> Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
SS>> URL:    www.evtmedia.com <http://www.evtmedia.com>   



>>> -Original Message-
>>> From: [EMAIL PROTECTED] 
>>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>>> shepherd fungayi
>>> Sent: Wednesday, July 02, 2003 10:49 AM
>>> To: [EMAIL PROTECTED]
>>> Subject: [Asterisk-Users] Asterisk PBX Billing
>>> 
>>> 
>>> Hi
>>> 
>>> I would like to use the Asterisk PBX as part of a phone shop 
>>> system instead 
>>> of the usual PBX plus PC. How can I do the the billing in a 
>>> way that is 
>>> convinient to the phone shop attendant?
>>> 
>>> Regards
>>> 
>>> Shepherd
>>> 
>>> _
>>> Add photos to your messages with MSN 8. Get 2 months FREE*. 
>>> http://join.msn.com/?page=features/featuredemail
>>> 
>>> ___
>>> Asterisk-Users mai

Re[4]: [Asterisk-Users] Asterisk PBX Billing

[Angelo Sampietro]

thanks a lot!
can you tell me where can i find more info about the CDR?
probably this will be the better way to give to the company a summary
with all the phone traffic :)

Angelo



Thursday, July 3, 2003, 4:37:32 PM, you wrote:

KCC> There is a CDR (Call Detail Record) which is accessible in two different
KCC> ways. The first is via a simple comma delimited file which can be parsed
KCC> and fed into whatever database that you want. The second way is to dump
KCC> the CDR directly into MySQL, and extract accordingly. So the only trick
KCC> there is to create a database for billing and create a relationship that
KCC> will extract from the CDR database.

KCC> Kim C. Callis

KCC> -Original Message-
KCC> From: [EMAIL PROTECTED]
KCC> [mailto:[EMAIL PROTECTED] On Behalf Of Angelo
KCC> Sampietro
KCC> Sent: Wednesday, July 02, 2003 7:06 AM
KCC> To: Scott Stingel
KCC> Cc: [EMAIL PROTECTED]
KCC> Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing

KCC> i think that the problem could be something more easy:

KCC> it is possible inside asterisk to log all che calls of all the users
KCC> and know the timing and the number called for each call?
KCC> if it is possible to do that, could be possible to make a program
KCC> that takes this files and generate the costs reading the log
KCC> informations...

KCC> so for me the real question is: there is a log of all the phone call
KCC> that are made by asterisk?

KCC> Angelo



KCC> Wednesday, July 2, 2003, 3:28:22 PM, you wrote:

SS>> Shepherd-

SS>> Having designed one of these in the past (in a higher level voice
SS>> environment), I can tell you that this is not a small undertaking.
KCC> It's at
SS>> least as much an SQL job as a voice task.

SS>> Usually the way to accomplish this is to establish more-or-less a
KCC> pre-paid
SS>> phone card system, where the shop prepays an overall amount for
SS>> international calling access.  Then you have to time each call as it
KCC> is
SS>> occurring, debiting each account, and the master account, in
KCC> real-time. This
SS>> can be a bit complex when you have 20 or 30 calls going at one time.
KCC> You
SS>> have to cut them off promptly when the money runs out (big problem).
KCC> And
SS>> you have to provide call detail and charges to them at the end of
KCC> each call,
SS>> using their own retail tariff.

SS>> To add to the complexity, each country has a different tariff from
KCC> the long
SS>> distance carrier, and within the country, major cities often have
KCC> special
SS>> rates per minute.  Mobiles have a different tariff too.  Phone card
SS>> platforms usually include a least-cost routing system which chooses
KCC> a
SS>> carrier real time based on the call.  Tariffs change weekly and must
KCC> be
SS>> updated in the system.

SS>> Anyway, I'm just scratching the surface!  I'll write more when I
KCC> can!

SS>> Cheers
SS>> Scott Stingel


SS>> Scott M. Stingel 
SS>> Emerging Voice Technology Inc.

SS>> Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
SS>> URL:    www.evtmedia.com <http://www.evtmedia.com>   



>>> -Original Message-
>>> From: [EMAIL PROTECTED] 
>>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>>> shepherd fungayi
>>> Sent: Wednesday, July 02, 2003 10:49 AM
>>> To: [EMAIL PROTECTED]
>>> Subject: [Asterisk-Users] Asterisk PBX Billing
>>> 
>>> 
>>> Hi
>>> 
>>> I would like to use the Asterisk PBX as part of a phone shop 
>>> system instead 
>>> of the usual PBX plus PC. How can I do the the billing in a 
>>> way that is 
>>> convinient to the phone shop attendant?
>>> 
>>> Regards
>>> 
>>> Shepherd
>>> 
>>> _
>>> Add photos to your messages with MSN 8. Get 2 months FREE*. 
>>> http://join.msn.com/?page=features/featuredemail
>>> 
>>> ___
>>> Asterisk-Users mailing list
>>> [EMAIL PROTECTED]
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> 
>>> 



SS>> ___
SS>> Asterisk-Users mailing list
SS>> [EMAIL PROTECTED]
SS>> http://lists.digium.com/mailman/listinfo/asterisk-users






-- 
Angelo Sampietro
IT Manager
ARC Interactive

"After a certain high level of technical skill is achieved, 
Science and art tend to coalesce in esthetics, plasticity, and form. 
The greatest scientists are always artists as well."
 
Albert Einstein 

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RE: Re[2]: [Asterisk-Users] Asterisk PBX Billing

[Kim C. Callis]

There is a CDR (Call Detail Record) which is accessible in two different
ways. The first is via a simple comma delimited file which can be parsed
and fed into whatever database that you want. The second way is to dump
the CDR directly into MySQL, and extract accordingly. So the only trick
there is to create a database for billing and create a relationship that
will extract from the CDR database.

Kim C. Callis

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angelo
Sampietro
Sent: Wednesday, July 02, 2003 7:06 AM
To: Scott Stingel
Cc: [EMAIL PROTECTED]
Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing

i think that the problem could be something more easy:

it is possible inside asterisk to log all che calls of all the users
and know the timing and the number called for each call?
if it is possible to do that, could be possible to make a program
that takes this files and generate the costs reading the log
informations...

so for me the real question is: there is a log of all the phone call
that are made by asterisk?

Angelo



Wednesday, July 2, 2003, 3:28:22 PM, you wrote:

SS> Shepherd-

SS> Having designed one of these in the past (in a higher level voice
SS> environment), I can tell you that this is not a small undertaking.
It's at
SS> least as much an SQL job as a voice task.

SS> Usually the way to accomplish this is to establish more-or-less a
pre-paid
SS> phone card system, where the shop prepays an overall amount for
SS> international calling access.  Then you have to time each call as it
is
SS> occurring, debiting each account, and the master account, in
real-time. This
SS> can be a bit complex when you have 20 or 30 calls going at one time.
You
SS> have to cut them off promptly when the money runs out (big problem).
And
SS> you have to provide call detail and charges to them at the end of
each call,
SS> using their own retail tariff.

SS> To add to the complexity, each country has a different tariff from
the long
SS> distance carrier, and within the country, major cities often have
special
SS> rates per minute.  Mobiles have a different tariff too.  Phone card
SS> platforms usually include a least-cost routing system which chooses
a
SS> carrier real time based on the call.  Tariffs change weekly and must
be
SS> updated in the system.

SS> Anyway, I'm just scratching the surface!  I'll write more when I
can!

SS> Cheers
SS> Scott Stingel


SS> Scott M. Stingel 
SS> Emerging Voice Technology Inc.

SS> Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
SS> URL:www.evtmedia.com <http://www.evtmedia.com>   



>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> shepherd fungayi
>> Sent: Wednesday, July 02, 2003 10:49 AM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] Asterisk PBX Billing
>> 
>> 
>> Hi
>> 
>> I would like to use the Asterisk PBX as part of a phone shop 
>> system instead 
>> of the usual PBX plus PC. How can I do the the billing in a 
>> way that is 
>> convinient to the phone shop attendant?
>> 
>> Regards
>> 
>> Shepherd
>> 
>> _
>> Add photos to your messages with MSN 8. Get 2 months FREE*. 
>> http://join.msn.com/?page=features/featuredemail
>> 
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED]
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>> 



SS> ___
SS> Asterisk-Users mailing list
SS> [EMAIL PROTECTED]
SS> http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Angelo Sampietro
IT Manager
ARC Interactive

"After a certain high level of technical skill is achieved, 
Science and art tend to coalesce in esthetics, plasticity, and form. 
The greatest scientists are always artists as well."
 
Albert Einstein 

___
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RE: Re[2]: [Asterisk-Users] Asterisk PBX Billing

[Chris Mason]

That's all I would need, it would be easy enough to work out the cost after
that.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angelo Sampietro
Sent: Wednesday, July 02, 2003 10:06 AM
To: Scott Stingel
Cc: [EMAIL PROTECTED]
Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing


i think that the problem could be something more easy:

it is possible inside asterisk to log all che calls of all the users and
know the timing and the number called for each call? if it is possible to do
that, could be possible to make a program that takes this files and generate
the costs reading the log informations...

so for me the real question is: there is a log of all the phone call that
are made by asterisk?

Angelo



Wednesday, July 2, 2003, 3:28:22 PM, you wrote:

SS> Shepherd-

SS> Having designed one of these in the past (in a higher level voice 
SS> environment), I can tell you that this is not a small undertaking.  
SS> It's at least as much an SQL job as a voice task.

SS> Usually the way to accomplish this is to establish more-or-less a 
SS> pre-paid phone card system, where the shop prepays an overall amount 
SS> for international calling access.  Then you have to time each call 
SS> as it is occurring, debiting each account, and the master account, 
SS> in real-time. This can be a bit complex when you have 20 or 30 calls 
SS> going at one time.  You have to cut them off promptly when the money 
SS> runs out (big problem).  And you have to provide call detail and 
SS> charges to them at the end of each call, using their own retail 
SS> tariff.

SS> To add to the complexity, each country has a different tariff from 
SS> the long distance carrier, and within the country, major cities 
SS> often have special rates per minute.  Mobiles have a different 
SS> tariff too.  Phone card platforms usually include a least-cost 
SS> routing system which chooses a carrier real time based on the call.  
SS> Tariffs change weekly and must be updated in the system.

SS> Anyway, I'm just scratching the surface!  I'll write more when I 
SS> can!

SS> Cheers
SS> Scott Stingel


SS> Scott M. Stingel
SS> Emerging Voice Technology Inc.

SS> Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
SS> URL:www.evtmedia.com <http://www.evtmedia.com>   



>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> shepherd fungayi
>> Sent: Wednesday, July 02, 2003 10:49 AM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] Asterisk PBX Billing
>> 
>> 
>> Hi
>> 
>> I would like to use the Asterisk PBX as part of a phone shop
>> system instead 
>> of the usual PBX plus PC. How can I do the the billing in a 
>> way that is 
>> convinient to the phone shop attendant?
>> 
>> Regards
>> 
>> Shepherd
>> 
>> _
>> Add photos to your messages with MSN 8. Get 2 months FREE*.
>> http://join.msn.com/?page=features/featuredemail
>> 
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED] 
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>> 



SS> ___
SS> Asterisk-Users mailing list
SS> [EMAIL PROTECTED] 
SS> http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Angelo Sampietro
IT Manager
ARC Interactive

"After a certain high level of technical skill is achieved, 
Science and art tend to coalesce in esthetics, plasticity, and form. 
The greatest scientists are always artists as well."
 
Albert Einstein 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
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Re[2]: [Asterisk-Users] Asterisk PBX Billing

[Angelo Sampietro]

i think that the problem could be something more easy:

it is possible inside asterisk to log all che calls of all the users
and know the timing and the number called for each call?
if it is possible to do that, could be possible to make a program
that takes this files and generate the costs reading the log
informations...

so for me the real question is: there is a log of all the phone call
that are made by asterisk?

Angelo



Wednesday, July 2, 2003, 3:28:22 PM, you wrote:

SS> Shepherd-

SS> Having designed one of these in the past (in a higher level voice
SS> environment), I can tell you that this is not a small undertaking.  It's at
SS> least as much an SQL job as a voice task.

SS> Usually the way to accomplish this is to establish more-or-less a pre-paid
SS> phone card system, where the shop prepays an overall amount for
SS> international calling access.  Then you have to time each call as it is
SS> occurring, debiting each account, and the master account, in real-time. This
SS> can be a bit complex when you have 20 or 30 calls going at one time.  You
SS> have to cut them off promptly when the money runs out (big problem).  And
SS> you have to provide call detail and charges to them at the end of each call,
SS> using their own retail tariff.

SS> To add to the complexity, each country has a different tariff from the long
SS> distance carrier, and within the country, major cities often have special
SS> rates per minute.  Mobiles have a different tariff too.  Phone card
SS> platforms usually include a least-cost routing system which chooses a
SS> carrier real time based on the call.  Tariffs change weekly and must be
SS> updated in the system.

SS> Anyway, I'm just scratching the surface!  I'll write more when I can!

SS> Cheers
SS> Scott Stingel


SS> Scott M. Stingel 
SS> Emerging Voice Technology Inc.

SS> Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
SS> URL:www.evtmedia.com <http://www.evtmedia.com>   



>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> shepherd fungayi
>> Sent: Wednesday, July 02, 2003 10:49 AM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] Asterisk PBX Billing
>> 
>> 
>> Hi
>> 
>> I would like to use the Asterisk PBX as part of a phone shop 
>> system instead 
>> of the usual PBX plus PC. How can I do the the billing in a 
>> way that is 
>> convinient to the phone shop attendant?
>> 
>> Regards
>> 
>> Shepherd
>> 
>> _
>> Add photos to your messages with MSN 8. Get 2 months FREE*. 
>> http://join.msn.com/?page=features/featuredemail
>> 
>> ___
>> Asterisk-Users mailing list
>> [EMAIL PROTECTED]
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>> 



SS> ___
SS> Asterisk-Users mailing list
SS> [EMAIL PROTECTED]
SS> http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Angelo Sampietro
IT Manager
ARC Interactive

"After a certain high level of technical skill is achieved, 
Science and art tend to coalesce in esthetics, plasticity, and form. 
The greatest scientists are always artists as well."
 
Albert Einstein 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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RE: [Asterisk-Users] Asterisk PBX Billing

[Scott Stingel]

Shepherd-

Having designed one of these in the past (in a higher level voice
environment), I can tell you that this is not a small undertaking.  It's at
least as much an SQL job as a voice task.

Usually the way to accomplish this is to establish more-or-less a pre-paid
phone card system, where the shop prepays an overall amount for
international calling access.  Then you have to time each call as it is
occurring, debiting each account, and the master account, in real-time. This
can be a bit complex when you have 20 or 30 calls going at one time.  You
have to cut them off promptly when the money runs out (big problem).  And
you have to provide call detail and charges to them at the end of each call,
using their own retail tariff.

To add to the complexity, each country has a different tariff from the long
distance carrier, and within the country, major cities often have special
rates per minute.  Mobiles have a different tariff too.  Phone card
platforms usually include a least-cost routing system which chooses a
carrier real time based on the call.  Tariffs change weekly and must be
updated in the system.

Anyway, I'm just scratching the surface!  I'll write more when I can!

Cheers
Scott Stingel


Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>   
URL:www.evtmedia.com <http://www.evtmedia.com>   



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> shepherd fungayi
> Sent: Wednesday, July 02, 2003 10:49 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk PBX Billing
> 
> 
> Hi
> 
> I would like to use the Asterisk PBX as part of a phone shop 
> system instead 
> of the usual PBX plus PC. How can I do the the billing in a 
> way that is 
> convinient to the phone shop attendant?
> 
> Regards
> 
> Shepherd
> 
> _
> Add photos to your messages with MSN 8. Get 2 months FREE*. 
> http://join.msn.com/?page=features/featuredemail
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 



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RE: [Asterisk-Users] Asterisk PBX Billing

[Chris Mason]

I'm very interested in the same thing for a hotel system I would like to
implement. Anyone know if the country codes be tied to a pricing lookup
table?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi
Sent: Wednesday, July 02, 2003 5:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PBX Billing


Hi

I would like to use the Asterisk PBX as part of a phone shop system instead 
of the usual PBX plus PC. How can I do the the billing in a way that is 
convinient to the phone shop attendant?

Regards

Shepherd

_
Add photos to your messages with MSN 8. Get 2 months FREE*. 
http://join.msn.com/?page=features/featuredemail

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Re: [Asterisk-Users] Asterisk PBX Billing

[Gary]

ok, I'll bite :-)

What the heck is a "phone shop system"  ??

On Wed, 02 Jul 2003 09:48:44 +, shepherd fungayi wrote:

>Hi
>
>I would like to use the Asterisk PBX as part of a phone shop system instead 
>of the usual PBX plus PC. How can I do the the billing in a way that is 
>convinient to the phone shop attendant?
>
>Regards
>
>Shepherd
>
>_
>Add photos to your messages with MSN 8. Get 2 months FREE*. 
>http://join.msn.com/?page=features/featuredemail
>
>___
>Asterisk-Users mailing list
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.



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[Asterisk-Users] Asterisk PBX Billing

[shepherd fungayi]

Hi

I would like to use the Asterisk PBX as part of a phone shop system instead 
of the usual PBX plus PC. How can I do the the billing in a way that is 
convinient to the phone shop attendant?

Regards

Shepherd

_
Add photos to your messages with MSN 8. Get 2 months FREE*. 
http://join.msn.com/?page=features/featuredemail

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