RE: [asterisk-users] ekiga register problems

[Brad Sumrall]

204/20466.176.193.46D  5063 Unmonitored

It just came up after a reboot on its own???

Go figure, windows problem!

Thank you

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, May 29, 2007 12:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ekiga register problems

On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote:
> returning newbie. 
> Trying to register ekiga for the first time to my asterisk server only.
> 
> [204] 
> user=204 
> context=internal 
> type=friend 
> secret=xxx 
> insecure=very 
> canreinvite=no 
> host=dynamic 
> disallow=all 
> allow=ulaw 
> allow=alaw 
> nat=no 
> 
> Can anyone tell me what I am missing? 
> I am not behind NAT or a firewall

What exactly is the problem you get?

What is the line for "204" in 'sip show peers'? 


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

[Tzafrir Cohen]

On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote:
> returning newbie. 
> Trying to register ekiga for the first time to my asterisk server only.
> 
> [204] 
> user=204 
> context=internal 
> type=friend 
> secret=xxx 
> insecure=very 
> canreinvite=no 
> host=dynamic 
> disallow=all 
> allow=ulaw 
> allow=alaw 
> nat=no 
> 
> Can anyone tell me what I am missing? 
> I am not behind NAT or a firewall

What exactly is the problem you get?

What is the line for "204" in 'sip show peers'? 


-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] language in zapata.conf [was: Language in Zaptel.conf]

[Tzafrir Cohen]

On Mon, May 28, 2007 at 06:08:38PM -0500, Carlos Chavez wrote:
> On Tue, 2007-05-29 at 00:53 +0300, Tzafrir Cohen wrote:
> > On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote:
> > > On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:
> > > 
> > > > As you can see I set the language=es parameter (and do this for 
> > > > all
> > > > interfaces).  I installed the spanish sound set for Asterisk
> > > > in /var/lib/asterisk/sounds/es (with links to the appropriate
> > > > directories for letter, digits, phonetic, etc).  All Zap interfaces
> > > > still play all sounds in English.  Only if I do a
> > > > Set(CHANNEL(language)=es) do I get the sounds in Spanish.  When I do a
> > > > "zap show channels" the language column is blank.
> > > 
> > > Duh. It seems that the value of language in zapata.conf is indeed 
> > > ignored. My bad. Writing a fix.
> > 
> > And sadly I was looking at an obsolete copy of the SVN. That has already 
> > been fixed in the SVN after 1.4.4 was released:
> > 
> >   http://bugs.digium.com/view.php?id=9626
> >   http://svn.digium.com/view/asterisk?rev=62331&view=rev
> > 
> > (reported by sergee and fixed by russel)
> > 
> > Well. At least I fixed the title of the thread...
> > 
> 
>   So is this fixed by using Asterisk SVN or Zaptel SVN?

Asterisk SVN. Or by applying the above small fix to 1.4.4 .

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

[Sahil Gupta]

Hi,
You need to enable overlapdial.

Regards,


Sahil Gupta
Chief Executive Officer
VoiceValley Group of Companies

Phone: +61-7-30188403
Fax: +61-7-30188499

On Tue, 29 May 2007, Carlos Hernandez wrote:


Hi all:

We are looking for someone with experience in Alcatel PBX  - PRI - Asterisk 
integration


Please get in touch off list.. We're wanting to hire a professional 
subcontractor, developer or company to get around some issues like these:


Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to 
Asterisk results in a disc tone

(Asterisk do send calls properly into Alcatel)

If / when we manage to get anything from Alcatel, we get just the first digit 
of the number the user is intending to call.. Asterisk expects the whole 
number at once, so it fails..
Most of the time we get nothing at all from Alcatel, we think something is 
missing, so Alcatel sees the link is down.


Please let me know if you have done this type of work before. We are not 
wanting to involve the Alcatel people, unless really required.


Is there any special way to set up zaptel/zapata so Alcatel detects the PRI 
to be operational?

Is there any special way to receive the calls once the PRI is up?

Right now asterisk is set with:  pri_net 
Any information or hints will be greatly appreciated


Thank you,
Carlos
NZ
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[asterisk-users] Alcatel - Asterisk setup

[Carlos Hernandez]

Hi all:

We are looking for someone with experience in Alcatel PBX  - PRI - 
Asterisk  integration


Please get in touch off list.. We're wanting to hire a professional 
subcontractor, developer or company to get around some issues like these:


Asterisk shows PRI to Alcatel is up, but when trying to dial from 
Alcatel to Asterisk results in a disc tone

(Asterisk do send calls properly into Alcatel)

If / when we manage to get anything from Alcatel, we get just the first 
digit of the number the user is intending to call.. Asterisk expects the 
whole number at once, so it fails..
Most of the time we get nothing at all from Alcatel, we think something 
is missing, so Alcatel sees the link is down.


Please let me know if you have done this type of work before. We are not 
wanting to involve the Alcatel people, unless really required.


Is there any special way to set up zaptel/zapata so Alcatel detects the 
PRI to be operational?

Is there any special way to receive the calls once the PRI is up?

Right now asterisk is set with:  pri_net  


Any information or hints will be greatly appreciated

Thank you,
Carlos
NZ
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[asterisk-users] ekiga register problems

[Brad Sumrall]

returning newbie. 
Trying to register ekiga for the first time to my asterisk server only.

[204] 
user=204 
context=internal 
type=friend 
secret=xxx 
insecure=very 
canreinvite=no 
host=dynamic 
disallow=all 
allow=ulaw 
allow=alaw 
nat=no 

Can anyone tell me what I am missing? 
I am not behind NAT or a firewall

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[asterisk-users] [1.2.18] Wrong steps in extensions.conf?

[Gilles Ganault]

Hello,

	Sometimes, when a call comes in from the PSTN through our VoIP gateway, 
the information that is sent to our web page that logs calls includes the 
original CID name instead of the one that is we expect to be rewritten on 
the fly using Asterisk's LookupCIDName:


=
;extensions.conf
[internal]
exten => group,1,LookupCIDName
exten => group,n,AGI(web.agi|${CALLERID(num)}|${CALLERID(name)})
exten => group,n,Dial(${EXT204})
=
;/var/lib/asterisk/agi-bin/web.agi
#!/usr/bin/perl

#use LWP::Simple;
use URI::Escape;
use LWP 5.64;

open STDOUT, '>/dev/null';
#Causes double entry?
fork and exit;

my $cidnum = $ARGV[0];
my $cidname = $ARGV[1];

$safe_cidname = uri_escape($cidname);

my $browser = LWP::UserAgent->new;

my $url = "http://www.acme.com/input.php?";;
$url .= "name=" . $safe_cidname . "&";
$url .= "number=" . $cidnum . "&";

($min, $hrs, $day, $month, $year) = (localtime) [1,2,3,4,5];
$currentdate = sprintf("%02d/%02d/%02d", $day, $month+1, $year % 100);
$currenttime = sprintf("%02d:%02d", $hrs,$min);
$url .= "date=" . $currentdate . "&";
$url .= "time=" . $currenttime;
#print $url . "\n";

my $response = $browser->get( $url );
die "Can't get $url -- ", $response->status_line unless $response->is_success;
print $response->content;
=

Could it be that, sometimes, Asterisk doesn't wait for the previous step to 
be completed before moving on to the next?


Thank you.

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RE: [asterisk-users] RE: Bottom line on fax reception

[Steve Totaro]

> >>
> >
> > Let me further qualify my results.  This was done with whatever the
> > current stable versions of Asterisk, Hylafax, and IAXmodem were
> > available in January of this year.  The faxes were outbound.  PDFs
put
> > into a Samba share and a cron job moving them over to the Hylafax
> > monitored directory.
> >
> 
> for my application I am more concerned with inbound working, outbound
> is just a bonus if it works. one of the big points is when you have a
> distributed workforce conventional fax machines don't work out since
> you get a paper result in one place and the recipient in another.
> Hylafax output can easily be redirected from a general delivery
> mailbox, or people can have their own fax extensions or DID to
> automate delivery even more. In my application voip itself really
> doesn't factor in either, the fax setup is on the same box the analog
> lines physically terminate at.
> 
> I have had pretty good luck with an old slow machine, ancient
> asterisk, low quality channel bank, and a physical fax modem on the
> same box as asterisk running hylafax, analog line in - pbx - analog
> line out - faxmodem, occasionally I get errors on faxes, and rarely
> someone can't get a fax through, but giving them the extension of a
> physical fax machine always works. So I am not convinced that problem
> is purely to blame on anything other than the far end station.
> 
> What I would like to eliminate is the fxs port and physical faxmodem
> from the setup and use iaxmodem instead (frees up a port, plus doesn't
> need faxmodem at all, and less complicated) it sounds like this sort
> of configuration works pretty well according to most of the posters. I
> know there are some issues with fax autodetection, but normally the
> sender fax is programmed to retry a few times, and failing that, your
> answer message could include a message to hit start on the fax machine
> if it does not start automatically, or dial an extension manually to
> start it.
> 
> another thing I like to do is if I scribble something down on a piece
> of paper, I just drop it in the fax machine and send it to the fax
> modem by calling its extension, I get a nicely scanned pdf in the mail
> that I can then forward to  anyone without knowing their fax number or
> paying for a fax call, great for emailing diagrams of things without
> taking the time to draw them on the computer.

Yes, I suppose the thread title is "reception".  

I am pretty sure the PDF decoding or encoding is what eats up the
processor cycles, tiff would probably be much less processing.  

The poor man's scanner option is also pretty nice.  Depending on the fax
machine, it could be on par with a Panafax which is a costly little and
awesome piece of equipment.

Thanks,
Steve

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RE: [asterisk-users] RE: Bottom line on fax reception

[Jon Pounder]

Quoting Steve Totaro <[EMAIL PROTECTED]>:




-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Pounder
Sent: Monday, May 28, 2007 9:10 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] RE: Bottom line on fax reception

Quoting Steve Totaro <[EMAIL PROTECTED]>:

> If you are a junk spam faxer then it should suit your needs.
>
> If you occasionally send faxes and if you do not receive one or the
> other party does not receive one or it spits out junk but that is

OK,

> then it should fit your needs.
>
> If you are faxing contracts or other important documents that are

worth

> something, then go for a more reliable solution.
>
> On a 3ghz HP DL320 with a gig of RAM, each fax took about 5%

indicated

> by top.  I would not want to go above ten simultaneous faxes so I

setup

> ten IAX Modems (50% in top).  Even at that rate, there were a lot of
> failures.  I did not bother to figure out why because these were

legal

> contracts, in bulk, amounting to big dollars.

anyone have a comparison with a multicpu machine with the same or
lower clock rate ?



Let me further qualify my results.  This was done with whatever the
current stable versions of Asterisk, Hylafax, and IAXmodem were
available in January of this year.  The faxes were outbound.  PDFs put
into a Samba share and a cron job moving them over to the Hylafax
monitored directory.



for my application I am more concerned with inbound working, outbound  
is just a bonus if it works. one of the big points is when you have a  
distributed workforce conventional fax machines don't work out since  
you get a paper result in one place and the recipient in another.  
Hylafax output can easily be redirected from a general delivery  
mailbox, or people can have their own fax extensions or DID to  
automate delivery even more. In my application voip itself really  
doesn't factor in either, the fax setup is on the same box the analog  
lines physically terminate at.


I have had pretty good luck with an old slow machine, ancient  
asterisk, low quality channel bank, and a physical fax modem on the  
same box as asterisk running hylafax, analog line in - pbx - analog  
line out - faxmodem, occasionally I get errors on faxes, and rarely  
someone can't get a fax through, but giving them the extension of a  
physical fax machine always works. So I am not convinced that problem  
is purely to blame on anything other than the far end station.


What I would like to eliminate is the fxs port and physical faxmodem  
from the setup and use iaxmodem instead (frees up a port, plus doesn't  
need faxmodem at all, and less complicated) it sounds like this sort  
of configuration works pretty well according to most of the posters. I  
know there are some issues with fax autodetection, but normally the  
sender fax is programmed to retry a few times, and failing that, your  
answer message could include a message to hit start on the fax machine  
if it does not start automatically, or dial an extension manually to  
start it.


another thing I like to do is if I scribble something down on a piece  
of paper, I just drop it in the fax machine and send it to the fax  
modem by calling its extension, I get a nicely scanned pdf in the mail  
that I can then forward to  anyone without knowing their fax number or  
paying for a fax call, great for emailing diagrams of things without  
taking the time to draw them on the computer.





Thanks,
Steve Totaro
www.asteriskhelpdesk.com





>
> The variables are very simple for any of these kind of decisions.

Don't

> think about savings, think about costs.
>
> Costs of equipment
> Costs of time (resources) implementing
> Costs of maintenance
> Costs of losing data (faxes in this case)
> Costs of going back and doing it the right way if you find the above
> costs are higher than another solution.
>
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
> KB3OPB



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Jon Pounder

   _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
_/_/_/  _/  _/ _/_/_/  _/  _/_/
   _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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RE: [asterisk-users] RE: Bottom line on fax reception

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jon Pounder
> Sent: Monday, May 28, 2007 9:10 PM
> To: asterisk-users@lists.digium.com
> Subject: RE: [asterisk-users] RE: Bottom line on fax reception
> 
> Quoting Steve Totaro <[EMAIL PROTECTED]>:
> 
> > If you are a junk spam faxer then it should suit your needs.
> >
> > If you occasionally send faxes and if you do not receive one or the
> > other party does not receive one or it spits out junk but that is
OK,
> > then it should fit your needs.
> >
> > If you are faxing contracts or other important documents that are
worth
> > something, then go for a more reliable solution.
> >
> > On a 3ghz HP DL320 with a gig of RAM, each fax took about 5%
indicated
> > by top.  I would not want to go above ten simultaneous faxes so I
setup
> > ten IAX Modems (50% in top).  Even at that rate, there were a lot of
> > failures.  I did not bother to figure out why because these were
legal
> > contracts, in bulk, amounting to big dollars.
> 
> anyone have a comparison with a multicpu machine with the same or
> lower clock rate ?
> 

Let me further qualify my results.  This was done with whatever the
current stable versions of Asterisk, Hylafax, and IAXmodem were
available in January of this year.  The faxes were outbound.  PDFs put
into a Samba share and a cron job moving them over to the Hylafax
monitored directory.

Thanks,
Steve Totaro
www.asteriskhelpdesk.com

> 
> 
> 
> >
> > The variables are very simple for any of these kind of decisions.
Don't
> > think about savings, think about costs.
> >
> > Costs of equipment
> > Costs of time (resources) implementing
> > Costs of maintenance
> > Costs of losing data (faxes in this case)
> > Costs of going back and doing it the right way if you find the above
> > costs are higher than another solution.
> >
> > Thanks,
> > Steve Totaro
> > http://www.asteriskhelpdesk.com
> > KB3OPB


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RE: [asterisk-users] TE205P, E1, Panasonic PBX and hang-up issues

[Steve Totaro]

According to your CLI output, the channel is being torn down.  

Is there a lag on the CLI between the inside channel and the outfacing
channel getting the hangup request?

Does this only happen on mobile phones?  I know if I call my cell and
hangup, it will continue to ring a couple or even a few more times.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of C F
> Sent: Monday, May 28, 2007 6:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] TE205P, E1, Panasonic PBX and hang-up
issues
> 
> Yes that makes more sense. Now to the problem, please post your
> zapata.conf as well as your zaptel.conf. Also if you don't mind
> downloading the config file from the Panasonic TD1232 and email to me
> off list so I can take a look at it and make sure the settings are ok
> on the panasonic side.
> 
> Thank you
> 
> On 5/28/07, Barry O'Donovan <[EMAIL PROTECTED]> wrote:
> > On Fri 25 May 2007, C F wrote:
> > > Are you sure the panasonic is TVP 100? I have installed over 50
> > > Panasonic systems in my life, and service many more, I have never
> > > heard of that system, and a quick google shows it's just a
VoiceMail
> > > system and not a PBX.
> >
> > Thanks for the reply. Does D1232 Digital Super Hybrid System make
more
> sense?
> >
> > Thanks,
> > Barry
> >
> > >
> > > On 5/23/07, Barry O'Donovan
<[EMAIL PROTECTED]>
> wrote:
> > > > Hey folks,
> > > >
> > > > I have a Digium TE205P working as a man in the middle:
> > > >
> > > > PRI line  Asterisk/TE205P  PBX
> > > >
> > > > The PBX is a Panasonic KX - TVP 100.
> > > >
> > > > Everything is working great except for one little issue.
Asterisk
> isn't
> > > > hanging up the PRI B channel when the PBX channel is hung up.
> > > >
> > > > I don't want to overload you with information but please ask if
more
> is
> > > > needed. I suspect I'm really hoping someone who had a similar
> problem
> > > > with just say "ah, I know what that is!".
> > > >
> > > > Versions in use for Zaptel, LibPRI and Asterisk are all the SVN
1.4
> > > > branch.
> > > >
> > > > To replicate:
> > > >
> > > > 1. dial a mobile (say) from one of the PBX phones;
> > > > 2. when you here a ring tone, hang up the PBX phone;
> > > > 3. the mobile continues to ring.
> > > >
> > > > The verbose output is:
> > > >
> > > > -- Accepting overlap call from '' to '' on
channel
> 0/17,
> > > > span 2
> > > > -- Starting simple switch on 'Zap/48-1'
> > > > -- Executing [EMAIL PROTECTED]:1]
> > > > Set("Zap/48-1", "RECORDFILE=/srv/recordings/live/1179858572.0")
in
> new
> > > > stack -- Executing [EMAIL PROTECTED]:2]
> > > > MixMonitor("Zap/48-1",
"/srv/recordings/live/1179858572.0.wav|b") in
> new
> > > > stack
> > > > -- Executing [EMAIL PROTECTED]:3] SetCallerPres("Zap/48-1",
> "allowed")
> > > > in new stack
> > > > -- Executing [EMAIL PROTECTED]:4] SetCallerID("Zap/48-1",
"5400")
> in
> > > > new stack
> > > > -- Executing [EMAIL PROTECTED]:5] Dial("Zap/48-1",
> "Zap/g0/0868017669")
> > > > in new stack
> > > > -- Requested transfer capability: 0x00 - SPEECH
> > > > -- Called g0/0868017669
> > > >   == Begin MixMonitor Recording Zap/48-1
> > > > -- Zap/1-1 is ringing
> > > > -- Channel 0/17, span 2 got hangup request, cause 16
> > > > -- Zap/1-1 answered Zap/48-1
> > > > -- Channel 0/1, span 1 got hangup request, cause 0
> > > > -- Hungup 'Zap/1-1'
> > > >   == Spawn extension (pbx, 0868017669, 5) exited non-zero on
> 'Zap/48-1'
> > > >   == End MixMonitor Recording Zap/48-1
> > > > -- Hungup 'Zap/48-1'
> > > > asterisk1*CLI>
> > > >
> > > >
> > > > Any suggestions or fixes that you might have from prior
instances
> would
> > > > be greatly appreciated.
> > > >
> > > > Thanks a million,
> > > >
> > > > Barry O'Donovan
> > > > http://www.barryodonovan.com/
> > > >
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> > > > asterisk-users mailing list
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> > > asterisk-users mailing list
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> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > --
> > Kind regards,
> > Barry O'Donovan
> > +353 86 801 7669
> >
> > http://www.barryodonovan.com/
> >
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RE: [asterisk-users] RE: Bottom line on fax reception

[Jon Pounder]

Quoting Steve Totaro <[EMAIL PROTECTED]>:


If you are a junk spam faxer then it should suit your needs.

If you occasionally send faxes and if you do not receive one or the
other party does not receive one or it spits out junk but that is OK,
then it should fit your needs.

If you are faxing contracts or other important documents that are worth
something, then go for a more reliable solution.

On a 3ghz HP DL320 with a gig of RAM, each fax took about 5% indicated
by top.  I would not want to go above ten simultaneous faxes so I setup
ten IAX Modems (50% in top).  Even at that rate, there were a lot of
failures.  I did not bother to figure out why because these were legal
contracts, in bulk, amounting to big dollars.


anyone have a comparison with a multicpu machine with the same or  
lower clock rate ?







The variables are very simple for any of these kind of decisions.  Don't
think about savings, think about costs.

Costs of equipment
Costs of time (resources) implementing
Costs of maintenance
Costs of losing data (faxes in this case)
Costs of going back and doing it the right way if you find the above
costs are higher than another solution.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB



-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Pounder
Sent: Monday, May 28, 2007 4:54 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] RE: Bottom line on fax reception

Quoting Steve Totaro <[EMAIL PROTECTED]>:

> Please qualify your usage.  A couple faxes a day, a couple hundred,

a

> couple thousand, or a couple hundred thousand?


well what is your usage where it doesn't work ?

I would like to know where it does and doesn't work as well but so far
various groups have conflicting opinions.



>
>
>
> Are you running asterisk and hylafax on the same machine?  What is

your

> TDM connectivity?
>
>
>
> Hylafax uses quite a lot of CPU juice.  Anyone ever scale up a quad
> T1/E1 server for faxing using asterisk and hylafax?  Must be a heck

of a

> server!
>
>
>
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
>

>
> KB3OPB
>
>
>   _
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Greg
> Kennedy
> Sent: Monday, May 28, 2007 4:02 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] RE: Bottom line on fax reception
>
>
>
>
> I gave up on the rxfax business as it never worked for me. I use
> iaxmodem and hylafax and it works perfectly, every single time i use

it.

> inbound or outbound doesnt matter.
> I have not read about anyone using iaxmodem and hylafax having any
> issues. and its fairly easy to setup. Took me about 1 hour total to

get

> everything installed and configured.
>
>
>> Message: 3
>> Date: Mon, 28 May 2007 08:20:22 -0700
>> From: "shadowym" <[EMAIL PROTECTED]>
>> Subject: RE: [asterisk-users] Bottom line on fax reception
>> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>> 
>> Message-ID: <[EMAIL PROTECTED]>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> Anybody??
>>
>> -Original Message-
>> From: shadowym [mailto:[EMAIL PROTECTED]
>> Sent: Thursday, May 24, 2007 9:35 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Bottom line on fax reception
>>
>>
>>
>> So what is the bottom line? Does it work or not. I've heard stories

it

>> works, it doesn't work, it kinda sorta works when it's not raining

out

> side.
>> Everything under the rainbow.
>>
>> What's the bottom line with recent updates on 1.2.x? Is it

production

> ready
>> for fax? By production ready I mean that it just works all the time
> and
>> doesn't need any babysitting. Do I have to worry about dropped

lines,

>> sometimes not detecting incoming fax toneyada yada.
>>
>> I know I don't have to use fax on Asterisk but I really want to for
> various
>> reasons. Mostly incoming but outgoing is a nice to have. Should I

use

> an
>> addon package and if so which one? Any help would be appreciated.
>>
>> Message: 6
>> Date: Mon, 28 May 2007 11:38:01 -0400
>> From: "Steve Totaro" <[EMAIL PROTECTED]>
>> Subject: RE: [asterisk-users] Bottom line on fax reception
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> 
>> Message-ID:
>>
>

<[EMAIL PROTECTED]

> .com>
>>
>> Content-Type: text/plain; charset="US-ASCII"
>>
>> Someone already answered this question. The answer is no, it does

not

>> work by your definition of production ready.
>>
>> Thanks,
>> Steve Totaro
>> http://www.asteriskhelpdesk.com
>> KB3OPB
>>
>>
>
>



Jon Pounder

_/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
 _/_/_/  _/  _/ _/_/_/  _/  _/_/
_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.i

RE: [asterisk-users] Multiple TDM400p cards in one machine -- nolonger an issue?

[Steve Totaro]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Lee Jenkins
> Sent: Monday, May 28, 2007 8:57 PM
> To: Chris Earle; Asterisk Users Mailing List - Non-Commercial
Discussion
> Subject: Re: [asterisk-users] Multiple TDM400p cards in one machine --
> nolonger an issue?
> 
> Chris Earle wrote:
> > Hi all,
> >
> > Years ago, I was pretty sure attempting to use two TDM400p cards in
one
> > machine was recommended against by Digium ... probably because the
cards
> > couldn't hack it, and/or interrupt problems etc
> >
> > I have seen some posts recently that seem to indicate it is in fact
> possible
> > these days thanks to some updated firmware perhaps? . I just
need to
> > have two in the server because the 4 ports aren't enough ...
> >
> > I'd rather just expand by one card rather than get a TDM2400 (or
> TDM800??)
> >
> > Anyone had recent success/failure with this sort of thing?
> >
> >
> 
> Sangoma Remora Card may be an option?
> 
> http://www.sangoma.com/datasheets/p_a200-specs
> 
> --
> 
> Warm Regards,
> 
> Lee
> 

Maybe it is time to look at a fractional T1?  I recently used
www.shopfort1.com for realtime pricing.  

Thanks,
Steve

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Re: [asterisk-users] Multiple TDM400p cards in one machine -- no longer an issue?

[Lee Jenkins]

Chris Earle wrote:

Hi all,

Years ago, I was pretty sure attempting to use two TDM400p cards in one
machine was recommended against by Digium ... probably because the cards
couldn't hack it, and/or interrupt problems etc

I have seen some posts recently that seem to indicate it is in fact possible
these days thanks to some updated firmware perhaps? . I just need to
have two in the server because the 4 ports aren't enough ...

I'd rather just expand by one card rather than get a TDM2400 (or TDM800??)

Anyone had recent success/failure with this sort of thing?




Sangoma Remora Card may be an option?

http://www.sangoma.com/datasheets/p_a200-specs

--

Warm Regards,

Lee



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RE: [asterisk-users] RE: Bottom line on fax reception

[Steve Totaro]

If you are a junk spam faxer then it should suit your needs.  

If you occasionally send faxes and if you do not receive one or the
other party does not receive one or it spits out junk but that is OK,
then it should fit your needs. 

If you are faxing contracts or other important documents that are worth
something, then go for a more reliable solution.

On a 3ghz HP DL320 with a gig of RAM, each fax took about 5% indicated
by top.  I would not want to go above ten simultaneous faxes so I setup
ten IAX Modems (50% in top).  Even at that rate, there were a lot of
failures.  I did not bother to figure out why because these were legal
contracts, in bulk, amounting to big dollars.

The variables are very simple for any of these kind of decisions.  Don't
think about savings, think about costs.

Costs of equipment
Costs of time (resources) implementing
Costs of maintenance 
Costs of losing data (faxes in this case) 
Costs of going back and doing it the right way if you find the above
costs are higher than another solution. 

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jon Pounder
> Sent: Monday, May 28, 2007 4:54 PM
> To: asterisk-users@lists.digium.com
> Subject: RE: [asterisk-users] RE: Bottom line on fax reception
> 
> Quoting Steve Totaro <[EMAIL PROTECTED]>:
> 
> > Please qualify your usage.  A couple faxes a day, a couple hundred,
a
> > couple thousand, or a couple hundred thousand?
> 
> 
> well what is your usage where it doesn't work ?
> 
> I would like to know where it does and doesn't work as well but so far
> various groups have conflicting opinions.
> 
> 
> 
> >
> >
> >
> > Are you running asterisk and hylafax on the same machine?  What is
your
> > TDM connectivity?
> >
> >
> >
> > Hylafax uses quite a lot of CPU juice.  Anyone ever scale up a quad
> > T1/E1 server for faxing using asterisk and hylafax?  Must be a heck
of a
> > server!
> >
> >
> >
> > Thanks,
> > Steve Totaro
> > http://www.asteriskhelpdesk.com
> >
 >>
> > KB3OPB
> >
> >
> >   _
> >
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Greg
> > Kennedy
> > Sent: Monday, May 28, 2007 4:02 PM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] RE: Bottom line on fax reception
> >
> >
> >
> >
> > I gave up on the rxfax business as it never worked for me. I use
> > iaxmodem and hylafax and it works perfectly, every single time i use
it.
> > inbound or outbound doesnt matter.
> > I have not read about anyone using iaxmodem and hylafax having any
> > issues. and its fairly easy to setup. Took me about 1 hour total to
get
> > everything installed and configured.
> >
> >
> >> Message: 3
> >> Date: Mon, 28 May 2007 08:20:22 -0700
> >> From: "shadowym" <[EMAIL PROTECTED]>
> >> Subject: RE: [asterisk-users] Bottom line on fax reception
> >> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> >> 
> >> Message-ID: <[EMAIL PROTECTED]>
> >> Content-Type: text/plain; charset="us-ascii"
> >>
> >> Anybody??
> >>
> >> -Original Message-
> >> From: shadowym [mailto:[EMAIL PROTECTED]
> >> Sent: Thursday, May 24, 2007 9:35 AM
> >> To: asterisk-users@lists.digium.com
> >> Subject: [asterisk-users] Bottom line on fax reception
> >>
> >>
> >>
> >> So what is the bottom line? Does it work or not. I've heard stories
it
> >> works, it doesn't work, it kinda sorta works when it's not raining
out
> > side.
> >> Everything under the rainbow.
> >>
> >> What's the bottom line with recent updates on 1.2.x? Is it
production
> > ready
> >> for fax? By production ready I mean that it just works all the time
> > and
> >> doesn't need any babysitting. Do I have to worry about dropped
lines,
> >> sometimes not detecting incoming fax toneyada yada.
> >>
> >> I know I don't have to use fax on Asterisk but I really want to for
> > various
> >> reasons. Mostly incoming but outgoing is a nice to have. Should I
use
> > an
> >> addon package and if so which one? Any help would be appreciated.
> >>
> >> Message: 6
> >> Date: Mon, 28 May 2007 11:38:01 -0400
> >> From: "Steve Totaro" <[EMAIL PROTECTED]>
> >> Subject: RE: [asterisk-users] Bottom line on fax reception
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >> 
> >> Message-ID:
> >>
> >
<[EMAIL PROTECTED]
> > .com>
> >>
> >> Content-Type: text/plain; charset="US-ASCII"
> >>
> >> Someone already answered this question. The answer is no, it does
not
> >> work by your definition of production ready.
> >>
> >> Thanks,
> >> Steve Totaro
> >> http://www.asteriskhelpdesk.com
> >> KB3OPB
> >>
> >>
> >
> >
> 
> 
> 
> Jon Pounder
> 
> _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
>  _/_/_/  _/  _/ _/_/_/  _/  _/_/
> _/_/  _/_/  _/ _/_/  _/_/  _/
> _/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/

Re: [asterisk-users] RE: Bottom line on fax reception

[Darrick Hartman]

Tim Litwiller wrote:

Greg Kennedy wrote:


I gave up on the rxfax business as it never worked for me. I use 
iaxmodem and hylafax and it works perfectly, every single time i use 
it. inbound or outbound doesnt matter.
I have not read about anyone using iaxmodem and hylafax having any 
issues. and its fairly easy to setup. Took me about 1 hour total to 
get everything installed and configured.


Where is a how to on this and does it pass thru to a fax machine?  
I've been fighting with getting faxing to work on my home asterisk 
machine and have given up.  But if you say it works well and reliably 
on large volume I'd be willing to try again on my home machine.


We get a quite low volume of faxes 10 or less per week. But they are 
how my wifes  family communicates and when our fax does work we start 
to loose contact with them, so she would argue that we should go back 
to plain old telephones. 
Fax reception does work reliably IF (big IF) you are faxing using a 
transport media that is conducive to faxing.  The internet is not a 
transport method that will result in 100% reliable connectivity.  If 
you're on a reasonably good internet connection with low-latency and 
jitter between you and your voip service AND you are using ulaw, you 
should get acceptable results for residential purposes.  I've connected 
my Brother MFC to a Digium TDM400 card and successfully sent and 
received faxes over the internet in the past.  I've also had my share of 
failures with the same connection when a large file was being 
downloaded, even with traffic shaping enabled. 

People need to be VERY clear about this when they say they are faxing 
successfully through Asterisk.  Lee has all sorts of ammunition why you 
shouldn't even try it over IP, at least not in a business setting.


Darrick

--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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RE: [asterisk-users] Bottom line on fax reception

[Paul Hales]

A colleague of mine did some testing the other day, with a digium TDM400
with FXS modules hooked up to fax machines and a TE120P hooked up to our
testing E1 line.

It seems to work pretty well, and she said it was easy to configure.

PaulH

On Mon, 2007-05-28 at 14:55 -0700, shadowym wrote:
> Thanks for all the replies.  Seems there are at least 2 or 3 people giving
> strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade
> production) solution.  That is just the sort of feedback I was looking for.
> 
> My application is just standard business reception of faxes.  Right now they
> use WinFax and probably receive about 30 to 50 faxes a day.  I want to wean
> them off Winfax as it's not really supported anymore and I dislike all
> things Symantec in general.
> 
> Receiving faxes on Asterisk has the added benefit of being able to use the
> fax line as an extra outgoing line when the rest are in use.  That is what
> they are doing now on their key system and they don't want to lose that
> ability.  I don't blame them.
> 
> -Original Message-
> From: Duncan Turnbull [mailto:[EMAIL PROTECTED] 
> Sent: Monday, May 28, 2007 2:18 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Bottom line on fax reception
> 
> I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no
> baby sitting, I receive about 20 and it requires no baby sitting
> 
> Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and
> hylafax lists for much bigger examples
> 
> Cheers Duncan
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of shadowym
> Sent: Tuesday, 29 May 2007 7:34 a.m.
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Bottom line on fax reception
> 
> Sorry but I must have missed it if someone else responded.  If the built in
> fax reception doesn't work very well what about the 3rd party stuff
> mentioned on the Asterisk Wiki?
> 
> -Original Message-
> From: Steve Totaro [mailto:[EMAIL PROTECTED]
> Sent: Monday, May 28, 2007 8:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Bottom line on fax reception
> 
> Someone already answered this question.  The answer is no, it does not work
> by your definition of production ready.
> 
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
> KB3OPB
>  
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> > [EMAIL PROTECTED] On Behalf Of shadowym
> > Sent: Monday, May 28, 2007 11:20 AM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [asterisk-users] Bottom line on fax reception
> > 
> > Anybody??
> > 
> > -Original Message-
> > From: shadowym [mailto:[EMAIL PROTECTED]
> > Sent: Thursday, May 24, 2007 9:35 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Bottom line on fax reception
> > 
> > 
> > 
> > So what is the bottom line?  Does it work or not.  I've heard stories
> it
> > works, it doesn't work, it kinda sorta works when it's not raining out 
> > side.
> > Everything under the rainbow.
> > 
> > What's the bottom line with recent updates on 1.2.x?  Is it production 
> > ready for fax?  By production ready I mean that it just works all the 
> > time
> and
> > doesn't need any babysitting.  Do I have to worry about dropped lines, 
> > sometimes not detecting incoming fax toneyada yada.
> > 
> > I know I don't have to use fax on Asterisk but I really want to for 
> > various reasons.  Mostly incoming but outgoing is a nice to have.
> > Should I
> use an
> > addon package and if so which one?  Any help would be appreciated.
> > 
> > 
> > 
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
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> 
> 
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Re: [asterisk-users] RE: Bottom line on fax reception

[Tim Litwiller]

Greg Kennedy wrote:


I gave up on the rxfax business as it never worked for me. I use 
iaxmodem and hylafax and it works perfectly, every single time i use 
it. inbound or outbound doesnt matter.
I have not read about anyone using iaxmodem and hylafax having any 
issues. and its fairly easy to setup. Took me about 1 hour total to 
get everything installed and configured.


Where is a how to on this and does it pass thru to a fax machine?  I've 
been fighting with getting faxing to work on my home asterisk machine 
and have given up.  But if you say it works well and reliably on large 
volume I'd be willing to try again on my home machine.


We get a quite low volume of faxes 10 or less per week. But they are how 
my wifes  family communicates and when our fax does work we start to 
loose contact with them, so she would argue that we should go back to 
plain old telephones.





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Re: [asterisk-users] language in zapata.conf [was: Language in Zaptel.conf]

[Carlos Chavez]

On Tue, 2007-05-29 at 00:53 +0300, Tzafrir Cohen wrote:
> On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote:
> > On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:
> > 
> > >   As you can see I set the language=es parameter (and do this for all
> > > interfaces).  I installed the spanish sound set for Asterisk
> > > in /var/lib/asterisk/sounds/es (with links to the appropriate
> > > directories for letter, digits, phonetic, etc).  All Zap interfaces
> > > still play all sounds in English.  Only if I do a
> > > Set(CHANNEL(language)=es) do I get the sounds in Spanish.  When I do a
> > > "zap show channels" the language column is blank.
> > 
> > Duh. It seems that the value of language in zapata.conf is indeed 
> > ignored. My bad. Writing a fix.
> 
> And sadly I was looking at an obsolete copy of the SVN. That has already 
> been fixed in the SVN after 1.4.4 was released:
> 
>   http://bugs.digium.com/view.php?id=9626
>   http://svn.digium.com/view/asterisk?rev=62331&view=rev
> 
> (reported by sergee and fixed by russel)
> 
> Well. At least I fixed the title of the thread...
> 

So is this fixed by using Asterisk SVN or Zaptel SVN?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] TE205P, E1, Panasonic PBX and hang-up issues

[C F]

Yes that makes more sense. Now to the problem, please post your
zapata.conf as well as your zaptel.conf. Also if you don't mind
downloading the config file from the Panasonic TD1232 and email to me
off list so I can take a look at it and make sure the settings are ok
on the panasonic side.

Thank you

On 5/28/07, Barry O'Donovan <[EMAIL PROTECTED]> wrote:

On Fri 25 May 2007, C F wrote:
> Are you sure the panasonic is TVP 100? I have installed over 50
> Panasonic systems in my life, and service many more, I have never
> heard of that system, and a quick google shows it's just a VoiceMail
> system and not a PBX.

Thanks for the reply. Does D1232 Digital Super Hybrid System make more sense?

Thanks,
Barry

>
> On 5/23/07, Barry O'Donovan <[EMAIL PROTECTED]> wrote:
> > Hey folks,
> >
> > I have a Digium TE205P working as a man in the middle:
> >
> > PRI line  Asterisk/TE205P  PBX
> >
> > The PBX is a Panasonic KX - TVP 100.
> >
> > Everything is working great except for one little issue. Asterisk isn't
> > hanging up the PRI B channel when the PBX channel is hung up.
> >
> > I don't want to overload you with information but please ask if more is
> > needed. I suspect I'm really hoping someone who had a similar problem
> > with just say "ah, I know what that is!".
> >
> > Versions in use for Zaptel, LibPRI and Asterisk are all the SVN 1.4
> > branch.
> >
> > To replicate:
> >
> > 1. dial a mobile (say) from one of the PBX phones;
> > 2. when you here a ring tone, hang up the PBX phone;
> > 3. the mobile continues to ring.
> >
> > The verbose output is:
> >
> > -- Accepting overlap call from '' to '' on channel 0/17,
> > span 2
> > -- Starting simple switch on 'Zap/48-1'
> > -- Executing [EMAIL PROTECTED]:1]
> > Set("Zap/48-1", "RECORDFILE=/srv/recordings/live/1179858572.0") in new
> > stack -- Executing [EMAIL PROTECTED]:2]
> > MixMonitor("Zap/48-1", "/srv/recordings/live/1179858572.0.wav|b") in new
> > stack
> > -- Executing [EMAIL PROTECTED]:3] SetCallerPres("Zap/48-1", "allowed")
> > in new stack
> > -- Executing [EMAIL PROTECTED]:4] SetCallerID("Zap/48-1", "5400") in
> > new stack
> > -- Executing [EMAIL PROTECTED]:5] Dial("Zap/48-1", "Zap/g0/0868017669")
> > in new stack
> > -- Requested transfer capability: 0x00 - SPEECH
> > -- Called g0/0868017669
> >   == Begin MixMonitor Recording Zap/48-1
> > -- Zap/1-1 is ringing
> > -- Channel 0/17, span 2 got hangup request, cause 16
> > -- Zap/1-1 answered Zap/48-1
> > -- Channel 0/1, span 1 got hangup request, cause 0
> > -- Hungup 'Zap/1-1'
> >   == Spawn extension (pbx, 0868017669, 5) exited non-zero on 'Zap/48-1'
> >   == End MixMonitor Recording Zap/48-1
> > -- Hungup 'Zap/48-1'
> > asterisk1*CLI>
> >
> >
> > Any suggestions or fixes that you might have from prior instances would
> > be greatly appreciated.
> >
> > Thanks a million,
> >
> > Barry O'Donovan
> > http://www.barryodonovan.com/
> >
> > ___
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Kind regards,
Barry O'Donovan
+353 86 801 7669

http://www.barryodonovan.com/


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Re: [asterisk-users] Blindside Web Conferencing

[Stefan Reuter]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hey Brad,

I am not sure if you know about the Asterisk-IM plugin for Openfire.
Basically it supports dialing contacts and arbitrary numbers through
Spark and updates presence based on being on call or not.
One of our next steps would be to integrate conferencing so you could
setup (and control) a voice conference much the same way you can do with
Jabber groupchat.
We also have a web conferencing app in a pre beta state sitting around
for some time now (based on Asterisk-Java, DWR and Tomcat) with the
original intend to use it for a commercial service which never got
really started though.
I am not sure if we could come together in some way but if you are
interested feel free to contact me off-list.

=Stefan
-BEGIN PGP SIGNATURE-
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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Re: [asterisk-users] DTMFToText Installation process

[Tzafrir Cohen]

On Mon, May 28, 2007 at 07:18:59PM +0530, rajesh koniki wrote:
> Hi,
> I was looking for a way to pass alphanumeric variables to asterisk via
> the keypad, found this application app_dtmftotext.c , And I already tried 
> with 'spandsp' application for this. But I am getting errors.[I followed 
> the instructions at http://www.soft-switch.org/installing-spandsp.html].
> 
> Can anybody be of help Me on this getting DTMFToText() application on 
> asterisk with the help of app_dtmftotext.c and/or spandsp application is 
> appreciated.

app_dtmftotext.c is an asterisk module that uses the library spandsp to
provide the dialplan application DTMFToText .

What version of Asterisk do you have? What version of spandsp did you 
try to install? Which linux distribution do you use?

-- 
   Tzafrir Cohen   
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RE: [asterisk-users] Blindside Web Conferencing

[Watkins, Bradley]

Thanks Stefan!  I was just thinking the other day that it would be great
if I could whiteboard in Spark.

Back on topic, I'm definitely interested in this web conferencing app.
I'll have to check it out once a .war is made available and I have a few
spare moments.

- Brad 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Stefan Reuter
> Sent: Monday, May 28, 2007 4:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Blindside Web Conferencing
> 
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> > Yes, we are looking for that. Do you know of any projects that 
> > provides those? I know one written in TCL/TK.
> 
> You might also want to have a look at
> http://www.version2software.com/v2whiteboard.html - its a 
> plugin for the Java based Jabber client Spark (from 
> igniterealtime.org)
> 
> =Stefan
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.6 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
> 
> iD8DBQFGW0DZcVCZDfrn+pMRAq4KAJ961ZBIsSNhn7p4+SQI4RPPe1gsHwCdG4dv
> pQOw6ugERcCUKy7pjDHf/qs=
> =JI7J
> -END PGP SIGNATURE-
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RE: [asterisk-users] Bottom line on fax reception

[shadowym]

Thanks for all the replies.  Seems there are at least 2 or 3 people giving
strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade
production) solution.  That is just the sort of feedback I was looking for.

My application is just standard business reception of faxes.  Right now they
use WinFax and probably receive about 30 to 50 faxes a day.  I want to wean
them off Winfax as it's not really supported anymore and I dislike all
things Symantec in general.

Receiving faxes on Asterisk has the added benefit of being able to use the
fax line as an extra outgoing line when the rest are in use.  That is what
they are doing now on their key system and they don't want to lose that
ability.  I don't blame them.

-Original Message-
From: Duncan Turnbull [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 28, 2007 2:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bottom line on fax reception

I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no
baby sitting, I receive about 20 and it requires no baby sitting

Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and
hylafax lists for much bigger examples

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Tuesday, 29 May 2007 7:34 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bottom line on fax reception

Sorry but I must have missed it if someone else responded.  If the built in
fax reception doesn't work very well what about the 3rd party stuff
mentioned on the Asterisk Wiki?

-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Monday, May 28, 2007 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bottom line on fax reception

Someone already answered this question.  The answer is no, it does not work
by your definition of production ready.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of shadowym
> Sent: Monday, May 28, 2007 11:20 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Bottom line on fax reception
> 
> Anybody??
> 
> -Original Message-
> From: shadowym [mailto:[EMAIL PROTECTED]
> Sent: Thursday, May 24, 2007 9:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Bottom line on fax reception
> 
> 
> 
> So what is the bottom line?  Does it work or not.  I've heard stories
it
> works, it doesn't work, it kinda sorta works when it's not raining out 
> side.
> Everything under the rainbow.
> 
> What's the bottom line with recent updates on 1.2.x?  Is it production 
> ready for fax?  By production ready I mean that it just works all the 
> time
and
> doesn't need any babysitting.  Do I have to worry about dropped lines, 
> sometimes not detecting incoming fax toneyada yada.
> 
> I know I don't have to use fax on Asterisk but I really want to for 
> various reasons.  Mostly incoming but outgoing is a nice to have.
> Should I
use an
> addon package and if so which one?  Any help would be appreciated.
> 
> 
> 
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Re: [asterisk-users] language in zapata.conf [was: Language in Zaptel.conf]

[Tzafrir Cohen]

On Tue, May 29, 2007 at 12:34:33AM +0300, Tzafrir Cohen wrote:
> On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:
> 
> > As you can see I set the language=es parameter (and do this for all
> > interfaces).  I installed the spanish sound set for Asterisk
> > in /var/lib/asterisk/sounds/es (with links to the appropriate
> > directories for letter, digits, phonetic, etc).  All Zap interfaces
> > still play all sounds in English.  Only if I do a
> > Set(CHANNEL(language)=es) do I get the sounds in Spanish.  When I do a
> > "zap show channels" the language column is blank.
> 
> Duh. It seems that the value of language in zapata.conf is indeed 
> ignored. My bad. Writing a fix.

And sadly I was looking at an obsolete copy of the SVN. That has already 
been fixed in the SVN after 1.4.4 was released:

  http://bugs.digium.com/view.php?id=9626
  http://svn.digium.com/view/asterisk?rev=62331&view=rev

(reported by sergee and fixed by russel)

Well. At least I fixed the title of the thread...

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] RE: Bottom line on fax reception

[Lee Howard]

Steve Totaro wrote:

Please qualify your usage.  A couple faxes a day, a couple hundred, a 
couple thousand, or a couple hundred thousand?




Couple hundred thousand per month - at least on one installation.

Are you running asterisk and hylafax on the same machine?  What is 
your TDM connectivity?




Yes, same machine, TDM is PRI, usually... at least it is on the 
installation I am mentioning.



Hylafax uses quite a lot of CPU juice.



Huh?  Certainly much, much less than Asterisk.

  Anyone ever scale up a quad T1/E1 server for faxing using asterisk 
and hylafax?  Must be a heck of a server!




It's okay.  :-)

Lee.
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[asterisk-users] language in zapata.conf [was: Language in Zaptel.conf]

[Tzafrir Cohen]

On Mon, May 28, 2007 at 04:08:35PM -0500, Carlos Chavez wrote:

>   As you can see I set the language=es parameter (and do this for all
> interfaces).  I installed the spanish sound set for Asterisk
> in /var/lib/asterisk/sounds/es (with links to the appropriate
> directories for letter, digits, phonetic, etc).  All Zap interfaces
> still play all sounds in English.  Only if I do a
> Set(CHANNEL(language)=es) do I get the sounds in Spanish.  When I do a
> "zap show channels" the language column is blank.

Duh. It seems that the value of language in zapata.conf is indeed 
ignored. My bad. Writing a fix.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Bottom line on fax reception

[Duncan Turnbull]

I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no 
baby sitting, I receive about 20 and it requires no baby
sitting

Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and 
hylafax lists for much bigger examples

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Tuesday, 29 May 2007 7:34 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bottom line on fax reception

Sorry but I must have missed it if someone else responded.  If the built in
fax reception doesn't work very well what about the 3rd party stuff
mentioned on the Asterisk Wiki?

-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 28, 2007 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bottom line on fax reception

Someone already answered this question.  The answer is no, it does not work
by your definition of production ready.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of shadowym
> Sent: Monday, May 28, 2007 11:20 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Bottom line on fax reception
> 
> Anybody??
> 
> -Original Message-
> From: shadowym [mailto:[EMAIL PROTECTED]
> Sent: Thursday, May 24, 2007 9:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Bottom line on fax reception
> 
> 
> 
> So what is the bottom line?  Does it work or not.  I've heard stories
it
> works, it doesn't work, it kinda sorta works when it's not raining out 
> side.
> Everything under the rainbow.
> 
> What's the bottom line with recent updates on 1.2.x?  Is it production 
> ready for fax?  By production ready I mean that it just works all the 
> time
and
> doesn't need any babysitting.  Do I have to worry about dropped lines, 
> sometimes not detecting incoming fax toneyada yada.
> 
> I know I don't have to use fax on Asterisk but I really want to for 
> various reasons.  Mostly incoming but outgoing is a nice to have.  
> Should I
use an
> addon package and if so which one?  Any help would be appreciated.
> 
> 
> 
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] ZAPTEL problem

[Tzafrir Cohen]

On Mon, May 28, 2007 at 02:22:34PM +0530, ram wrote:
> Hi
> 
> I have 100XP Digium clone card
> 
> Installed in my pc
> 
> and compiled zaptel and asterisk again after installing the card
> 
> but after i  rebooted
> 
> i can load zaptel and wcfxo modprobe with out any problem
> 
> but when i intiated ztcfg -
> 
> i get the following error
> 
> 
> Zaptel Configuration
> ==
> 
> 
> 1 channels configured.
> ZT_CHANCONFIG failed on channel 1: No such device or address (6)

No device to talk to. Why?

> 
> 
> 
> -
> 
> dmesg errors
> 
> Zapata Telephony Interface Unloaded
> Zapata Telephony Interface Registered on major 196
> Zaptel Version: SVN-branch-1.2-r1725 Echo Canceller: KB1
> Failed to initailize DAA, giving up...
> wcfxo: probe of :00:10.0 failed with error -5

The problem is here. Defective card? PCI-level problems?

An example of PCI-level issues is the following "magic" that worked
for me with several cards with some computers (at least with some 
specific kernels) - adding the boot parameter 'pci=noacpi' to the kernel 
command-line.

> 
> --
> 
> lspci
> 
> 00:0c.1 SCSI storage controller: Adaptec AIC-7896U2/7897U2
> 00:10.0 Communication controller: Motorola Unknown device 5608

I assume that the second line is the X100P card.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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[asterisk-users] Language in Zaptel.conf

[Carlos Chavez]

I am having a problem setting the default language for Zap interfaces.
I have an Asterisk 1.4.4 server on CentOS 5 with two Astribank 8 units
for analog devices.  Here is a sample configuration on one of the ports:

language=es
context=oficina
callerid="Miriam Perez Vite"<100>
mailbox=100
usecallerid=yes
callwaiting=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echotraining=800
rxgain=0.0
txgain=0.0
busydetect=yes
busycount=6
callprogress=no
accountcode=General
amaflags=default
signalling=fxo_ks
pickupgroup=1
callgroup=1
faxdetect=no
group=0
channel => 32

As you can see I set the language=es parameter (and do this for all
interfaces).  I installed the spanish sound set for Asterisk
in /var/lib/asterisk/sounds/es (with links to the appropriate
directories for letter, digits, phonetic, etc).  All Zap interfaces
still play all sounds in English.  Only if I do a
Set(CHANNEL(language)=es) do I get the sounds in Spanish.  When I do a
"zap show channels" the language column is blank.

The server also has an E1 with Unicall where I have also set the
default language to Spanish and all calls coming through the E1 do play
the sounds in Spanish.  Any ideas why the Zap channels do not want to
set the default language?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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RE: [asterisk-users] RE: Bottom line on fax reception

[Jon Pounder]

Quoting Steve Totaro <[EMAIL PROTECTED]>:


Please qualify your usage.  A couple faxes a day, a couple hundred, a
couple thousand, or a couple hundred thousand?



well what is your usage where it doesn't work ?

I would like to know where it does and doesn't work as well but so far  
various groups have conflicting opinions.








Are you running asterisk and hylafax on the same machine?  What is your
TDM connectivity?



Hylafax uses quite a lot of CPU juice.  Anyone ever scale up a quad
T1/E1 server for faxing using asterisk and hylafax?  Must be a heck of a
server!



Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com

Subject: RE: [asterisk-users] Bottom line on fax reception
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"

Anybody??

-Original Message-
From: shadowym [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 24, 2007 9:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Bottom line on fax reception



So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out

side.

Everything under the rainbow.

What's the bottom line with recent updates on 1.2.x? Is it production

ready

for fax? By production ready I mean that it just works all the time

and

doesn't need any babysitting. Do I have to worry about dropped lines,
sometimes not detecting incoming fax toneyada yada.

I know I don't have to use fax on Asterisk but I really want to for

various

reasons. Mostly incoming but outgoing is a nice to have. Should I use

an

addon package and if so which one? Any help would be appreciated.

Message: 6
Date: Mon, 28 May 2007 11:38:01 -0400
From: "Steve Totaro" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] Bottom line on fax reception
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:


<[EMAIL PROTECTED]
.com>


Content-Type: text/plain; charset="US-ASCII"

Someone already answered this question. The answer is no, it does not
work by your definition of production ready.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB









Jon Pounder

   _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
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   _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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Re: [asterisk-users] Blindside Web Conferencing

[Stefan Reuter]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

> Yes, we are looking for that. Do you know of any projects that provides
> those? I know one written in TCL/TK.

You might also want to have a look at
http://www.version2software.com/v2whiteboard.html - its a plugin for the
Java based Jabber client Spark (from igniterealtime.org)

=Stefan
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFGW0DZcVCZDfrn+pMRAq4KAJ961ZBIsSNhn7p4+SQI4RPPe1gsHwCdG4dv
pQOw6ugERcCUKy7pjDHf/qs=
=JI7J
-END PGP SIGNATURE-
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[asterisk-users] Send parked call extension to set

[Dave Bour]

Anyone know how to send a post transfer (and possibly post hangup) message to 
an aastra set with the pickup extension
Dave

Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]

For those who just want it to work...
Giving you complete IT peace of mind. 

(Sent via Blackberry - hence message may be shorter than my usual verbose 
responses)
PIN 4cc364db (as of March 24, 2007)  
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RE: [asterisk-users] RE: Bottom line on fax reception

[Steve Totaro]

Please qualify your usage.  A couple faxes a day, a couple hundred, a
couple thousand, or a couple hundred thousand?

 

Are you running asterisk and hylafax on the same machine?  What is your
TDM connectivity?

 

Hylafax uses quite a lot of CPU juice.  Anyone ever scale up a quad
T1/E1 server for faxing using asterisk and hylafax?  Must be a heck of a
server!

 

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
 
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Kennedy
Sent: Monday, May 28, 2007 4:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RE: Bottom line on fax reception

 


I gave up on the rxfax business as it never worked for me. I use
iaxmodem and hylafax and it works perfectly, every single time i use it.
inbound or outbound doesnt matter. 
I have not read about anyone using iaxmodem and hylafax having any
issues. and its fairly easy to setup. Took me about 1 hour total to get
everything installed and configured.


> Message: 3
> Date: Mon, 28 May 2007 08:20:22 -0700
> From: "shadowym" <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] Bottom line on fax reception
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
> 
> Anybody?? 
> 
> -Original Message-
> From: shadowym [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, May 24, 2007 9:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Bottom line on fax reception
> 
> 
> 
> So what is the bottom line? Does it work or not. I've heard stories it
> works, it doesn't work, it kinda sorta works when it's not raining out
side.
> Everything under the rainbow.
> 
> What's the bottom line with recent updates on 1.2.x? Is it production
ready
> for fax? By production ready I mean that it just works all the time
and
> doesn't need any babysitting. Do I have to worry about dropped lines,
> sometimes not detecting incoming fax toneyada yada. 
> 
> I know I don't have to use fax on Asterisk but I really want to for
various
> reasons. Mostly incoming but outgoing is a nice to have. Should I use
an
> addon package and if so which one? Any help would be appreciated. 
> 
> Message: 6
> Date: Mon, 28 May 2007 11:38:01 -0400
> From: "Steve Totaro" <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] Bottom line on fax reception
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Message-ID:
>
<[EMAIL PROTECTED]
.com>
> 
> Content-Type: text/plain; charset="US-ASCII"
> 
> Someone already answered this question. The answer is no, it does not
> work by your definition of production ready.
> 
> Thanks,
> Steve Totaro
> http://www.asteriskhelpdesk.com
> KB3OPB
> 
>

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RE: [asterisk-users] Polycom Static IP

[Steve Totaro]

Sounds like a firmware bug, VLAN or other network configuration bug in
the phone (subnet perhaps?)

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com 
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forum
Sent: Monday, May 28, 2007 3:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom Static IP

 

I am still having issues with my Polycom 301 phones when I disable DHCP.
I give the phone a static address and I keep getting the error 'could
not contact boot server using existing config'.  As soon as I set it
back to DHCP enabled the phone can see the boot server and I'm online.

 

Steve

 

 

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RE: [asterisk-users] Blindside Web Conferencing

[Steve Totaro]

http://www.thecoccinella.org/ looks pretty nice

 

I have not tried this one.  It has been a couple of years since I played
around with IM clients and I cannot remember what I was using.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com/ 
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Alam
Sent: Monday, May 28, 2007 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blindside Web Conferencing

 

Hi Steve,

 

Yes, we are looking for that. Do you know of any projects that provides
those? I know one written in TCL/TK.

 

Thanks.

 

Richard

 

On 5/28/07, Steve Totaro <[EMAIL PROTECTED]> wrote: 

Sounds cool.  You could probably use some code from the various open
source jabber clients that allow for shared whiteboard and pushing URLs
too. 

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com   
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] ] On Behalf Of Richard
Alam
Sent: Monday, May 28, 2007 10:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Blindside Web Conferencing

 

Hello,

 

We are creating a web-based conferencing application using Asterisk as
the voice conferencing server. 

 

This as an open source project. We are trying to determine if there is
interest of the community and perhaps work together to improve the
application. 

 

Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
(mute/unmute/kick), chat, participants can raise/lower hand to indicate
she wants to talk when muted. 

 

Please visit http://present.blindsideproject.org
 (http://present.sce.carleton.ca
  if that doesn't work) to try it out.
Click on the "requirements" link for instructions on how to setup your
Idefisk. 

 

Project website is at http://www.blindsideproject.org
 (http://www.blindsideproject.org:8080
  if that doesn't work). We are
requesting the University permissions to configure DNS/firewall, so
please bear with us. 

 

Please send feedback/suggestions to this mailing list. We're also trying
to setup our own. Hope the moderators of this list don't mind. 

 

Hoping for your feedbacks.

 

Thanks.

 

Blindside Project Team


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Re: [asterisk-users] Bottom line on fax reception

[Andrew Joakimsen]

On 5/24/07, shadowym <[EMAIL PROTECTED]> wrote:

So what is the bottom line?  Does it work or not.  I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.

What's the bottom line with recent updates on 1.2.x?  Is it production ready
for fax?  By production ready I mean that it just works all the time and
doesn't need any babysitting.  Do I have to worry about dropped lines,
sometimes not detecting incoming fax toneyada yada.

I know I don't have to use fax on Asterisk but I really want to for various
reasons.  Mostly incoming but outgoing is a nice to have.  Should I use an
addon package and if so which one?  Any help would be appreciated.




No. Digium still refuses to include proper faxing support in Asterisk.
OpenPBX is still unstable and developers really aren't too excited
about getting it to work... to much other stuff to fix before they
worry about fax. FWIW OPBX has seemed sort of dead recently.

If you want proper (T.38) fax support then pick up a Cisco AS53xx,
AS54xx, AS58xx, 2600, 3600, 7200. You need IOS 12.1 or above, double
check Cisco for specific WIC, RAM and other misc. requirements of
course!
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[asterisk-users] RE: Bottom line on fax reception

[Greg Kennedy]

I gave up on the rxfax business as it never worked for me. I use iaxmodem and 
hylafax and it works perfectly, every single time i use it. inbound or outbound 
doesnt matter. 
I have not read about anyone using iaxmodem and hylafax having any issues. and 
its fairly easy to setup. Took me about 1 hour total to get everything 
installed and configured.> Message: 3> Date: Mon, 28 May 2007 08:20:22 -0700> 
From: "shadowym" <[EMAIL PROTECTED]>> Subject: RE: [asterisk-users] Bottom line 
on fax reception> To: "'Asterisk Users Mailing List - Non-Commercial 
Discussion'"> > Message-ID: <[EMAIL 
PROTECTED]>> Content-Type: text/plain; charset="us-ascii"> > Anybody?? > > 
-Original Message-> From: shadowym [mailto:[EMAIL PROTECTED] > Sent: 
Thursday, May 24, 2007 9:35 AM> To: asterisk-users@lists.digium.com> Subject: 
[asterisk-users] Bottom line on fax reception> > > > So what is the bottom 
line? Does it work or not. I've heard stories it> works, it doesn't work, it 
kinda sorta works when it's not raining out side.> Everything under the 
rainbow.> > What's the bottom line with recent updates on 1.2.x? Is it 
production ready> for fax? By production ready I mean that it just works all 
the time and> doesn't need any babysitting. Do I have to worry about dropped 
lines,> sometimes not detecting incoming fax toneyada yada. > > I know I 
don't have to use fax on Asterisk but I really want to for various> reasons. 
Mostly incoming but outgoing is a nice to have. Should I use an> addon package 
and if so which one? Any help would be appreciated. > > Message: 6> Date: Mon, 
28 May 2007 11:38:01 -0400> From: "Steve Totaro" <[EMAIL PROTECTED]>> Subject: 
RE: [asterisk-users] Bottom line on fax reception> To: "Asterisk Users Mailing 
List - Non-Commercial Discussion"> > 
Message-ID:> <[EMAIL PROTECTED]>> > Content-Type: text/plain; 
charset="US-ASCII"> > Someone already answered this question. The answer is no, 
it does not> work by your definition of production ready.> > Thanks,> Steve 
Totaro> http://www.asteriskhelpdesk.com> KB3OPB> >___
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Re: [asterisk-users] Bottom line on fax reception

[Doug Lytle]

Doug Lytle wrote:

shadowym wrote:
 
So what is the bottom line?  Does it work or not.  I've heard stories it
  


As it has been said many many times before, Fax detection is an art 
and most of the time is not reliable.  Faxing on the other hand, using 
iaxmodem along with HylaFAX+ works very well.  Search the archives.


Doug




Reposting

Doug

--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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[asterisk-users] Polycom Static IP

[Forum]

I am still having issues with my Polycom 301 phones when I disable DHCP.  I
give the phone a static address and I keep getting the error 'could not
contact boot server using existing config'.  As soon as I set it back to
DHCP enabled the phone can see the boot server and I'm online.

 

Steve

 

 

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[asterisk-users] help on asterisk sipp

[khawla khawla]

I was wondering whether someone could help me. I installed sipp on my Asterisk 
server but I don't really understand how does it fonction! Has someone ever 
tried it?If you can explain to me the principle, I would be extremely 
grateful.Thank you very much in advance.

Lancez des recherches en toute sécurité depuis n'importe quelle page Web. 
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maintenant ! 
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RE: [asterisk-users] Bottom line on fax reception

[shadowym]

Sorry but I must have missed it if someone else responded.  If the built in
fax reception doesn't work very well what about the 3rd party stuff
mentioned on the Asterisk Wiki?

-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 28, 2007 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bottom line on fax reception

Someone already answered this question.  The answer is no, it does not work
by your definition of production ready.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of shadowym
> Sent: Monday, May 28, 2007 11:20 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Bottom line on fax reception
> 
> Anybody??
> 
> -Original Message-
> From: shadowym [mailto:[EMAIL PROTECTED]
> Sent: Thursday, May 24, 2007 9:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Bottom line on fax reception
> 
> 
> 
> So what is the bottom line?  Does it work or not.  I've heard stories
it
> works, it doesn't work, it kinda sorta works when it's not raining out 
> side.
> Everything under the rainbow.
> 
> What's the bottom line with recent updates on 1.2.x?  Is it production 
> ready for fax?  By production ready I mean that it just works all the 
> time
and
> doesn't need any babysitting.  Do I have to worry about dropped lines, 
> sometimes not detecting incoming fax toneyada yada.
> 
> I know I don't have to use fax on Asterisk but I really want to for 
> various reasons.  Mostly incoming but outgoing is a nice to have.  
> Should I
use an
> addon package and if so which one?  Any help would be appreciated.
> 
> 
> 
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RE: [asterisk-users] Octasic echo cancellation

[shadowym]

Hi Sebastien,

I'm just a lowly user but I will tell you what I think I understand about
it.  

There is nothing in the Octasic documentation that suggests you can have
continuosly updated statistics but I agree that would be a nice to have
feature.  Have you tried contacting Octasic about that?

You can set the ERL and tail length in the octveqd.conf file which you need
to manually copy into the /etc/ directory as per the Octasic documentation
and octveqd.conf file included with softecho.  By default they are set to
9db and 64ms respectively.

It is my understanding EC takes place AFTER the tx/rx gain as part of the
Zaptel driver function.  Therefore your tx/rx gain WILL have an effect on
it's operation and (I think) the ERL values.

If your using Digium cards you should run fxotune and set your gains.  With
Sangoma cards you only need to set your gains.  If that is all set up ok
softecho should get rid of all your echo.   If not I would suggest you try
increase the tail length to 128ms in octveqd.conf.  I don't think you should
need to change any of the other settings. 
 

-Original Message-
From: Sebastien Leclere [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 28, 2007 10:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Octasic echo cancellation

Hi,

I'm currently testing SoftEcho, an echo cancellation software for Asterisk
from Octasic. I noticed an important increase of the quality of my coms, but
I still have a few echo problems.

There is an ERL parameter which corresponds to an initial ERL value probably
to optimize the echo training or something in that kind.

Is there a way to monitor, using one of the zttools, the instant ERL value,
to be able to set this parameter correctly?

In that purpose, I would like to know where do the echo cancellation take
place in the communication chain, is it after or before the amplification
by the Rx/Tx gain parameters? Will this gain apply on the echo cancelled
signal or on the gross signal?

Anybody with experience on this product could give me some advice in the aim
of removing all trace of echo?


Thanks by advance

Sebastien


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Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

[Matthew J. Roth]

Mark Coccimiglio wrote:
Sounds like you are running into the hardware limitations of your 
systems PCI or "Front Side Bus" (FSB) and not necessarily an issue of 
asterisk.  In short there is a limited amount of bandwidth on the 
computer's PCI Bus (33 MHz)  and the FSB (100-800MHz).  One thing to 
remember is that ALL cores and data streams need to share the PCI and 
FSB.Asterisk is very processor and memory intensive.  At the 
extreme level of usage more cores won't help if data is "stuck in the 
pipe".  So the performance planing you described would be expected.

Mark,

That is a great theory and I'd like to follow up on it.  Do you know if 
the PCI or FSB buses are instrumented by Linux?  If not, are you aware 
of any way to gather statistics about their utilization?   I'd like to 
see if the numbers support your idea and, if so, which bus is saturated.


Let me add a little bit of extra information to this discussion.  The 
CPU utilization does not flatten out at 50%.  In fact, as more calls are 
added, Asterisk will eventually drive the idle percentage down to single 
digits with surprisingly few problems.  If PCI or FSB bandwidth were the 
limiting factor, wouldn't the CPU utilization top out at the point that 
the available bandwidth was used?


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] RE: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

[Matthew J. Roth]

JR Richardson wrote:

The Dual-Core system you are working with must have cost a bundle, several
thousand.  My approach has been to stick with single cpu, single core
servers and add more servers to the cluster, versus building bigger, faster
Proc servers. With sub $1000 servers, I can achieve 150-200 calls per
server, cluster several servers together and for the same price as a quad
proc dual-core server have 700-1000 call capacity.

Now, with that said, a cluster becomes harder to build and operate than a 1
server Asterisk implementation and does not work well in some environments,
such as with large call queues.  But when you are talking straight call
capacity, multiple servers will usually dominate singe servers in relation
to cost.
At the start of our Asterisk project, scaling vertically seemed like the 
simpler approach and to a certain extent it is necessary because of our 
call queues.  Now that it seems like we've pushed this approach as far 
as it will go, my eagerness to study and implement Asterisk clusters is 
growing.  I know that horizontal scaling will have its own set of 
problems, but at this point I think they will be more manageable than 
what we're currently dealing with.


Could you please give me a rough overview of your clustering 
architecture?  I don't need too many details, but a list of the 
technologies/programs you are using would be a great basis for my 
research.  I currently planning to look into SER and DUNDi, and another 
poster suggested looking at OpenSSI. 

Nice discussion, and thanks for posting your benchmark results and feedback.
  
You're welcome.  I'll post some large scale numbers off of our 
production server soon.  I'm also going to start looking at SIPp, which 
may give me a way to gather large scale statistics in a more controlled 
environment.


Thank you for your responses,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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RE: [asterisk-users] Asterisk and cell phones

[Dean Collins]

Hi Olivier,

Do a search on my blog www.collins.net.pr/blog for Orative as a
suggested application.

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

    
 
 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Monday, 28 May 2007 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk and cell phones

 

Hello,

PBX vendors used to sell software extensions providing enterprise
services to cell phones.

Main features were :
- 4 digit dialing or directory access,
- call forwarding,
- unified messaging.

Now that WiFi and dual mode cell phones get more popular, these
Java-based software should be more successful on the market as bandwidth
costs  decrease.
 
My question are :
1. How Asterisk would compare on cell phone integration with PBX or IPBX
vendors ? 
2. Is there now a way to enhance high end cell phones with third party
software to gain access to Asterisk features ?
3. More specifically, has anyone tested Nokia EXX dialing capabilities ?
Would it be easy to customise some callback features  with such phones 
For example :
you dial a number and hit a home-made Callback key : your phone then
dials a callback server number instead, hangs up, waits for this
callback server to call and lastly replies the real number you would
like to be connected with (this could be useful when receiving charges
are (much) cheaper than emitting charges). 

Best regards



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Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks

[Matthew J. Roth]

William Moore wrote:

Are you recording memory figures as well and have you checked the
total used memory?  Or did I miss it somewhere?  Thanks for doing
this, scalability testing is always good.


William,

This round of benchmarking is heavily focused on CPU utilization,
because it is causing an immediate problem for me.  However, I am
tracking some other statistics on a daily basis including memory
utilization, swap utilization, load averages, and active channels and calls.

One of my colleagues takes the text file I produce and creates graphs
using Cacti and rrdtool.  You'll be interested in these two (sorry for 
the format of the URLS, but otherwise the list was eating my posts):


- Percent CPU Used With No. Calls and No. Channels

- Asterisk Memory Used (KB)


Note that even with a peak call volume of approximately 400 active calls
and 550 active SIP channels, the memory utilization never surpasses 600
KB.  I'd estimate that most Asterisk installations would avoid swapping
with 1 GB of RAM.  A 2nd GB might be useful to provide plenty of room
for file caching so that your hard disk doesn't become a bottleneck.  We
also record all of our calls to a 6 GB RAM disk, so our server has a
total of 8 GB of RAM but that isn't necessary in most circumstances.

Overall, Asterisk seems to be very efficiently coded as far as memory is
concerned.  Note that for other reasons we perform a nightly reboot, so
I don't know if there are any memory leaks that would surface over time.

Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer




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Re: [asterisk-users] Blindside Web Conferencing

[Roberto Fichera]

At 19.56 28/05/2007, you wrote:



>On 5/28/07, Roberto Fichera <[EMAIL PROTECTED]> 
>wrote: 
>At 19.19 28/05/2007, you wrote:
>
>
>>On 5/28/07, Roberto Fichera <>PROTECTED]>[EMAIL PROTECTED]> wrote:
>>At 17.09 28/05/2007, Richard Alam wrote:
>>>Yes, we have some downloadable code. We are in the process of completing the 
>>>instructions (build/deploy/etc.). 
>>>
>>>Code is located here.
>>><<http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk
>>> 
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk>
>>> 
>>>http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk
>>
>>How can I get it?
>>
>>Hi,
>>
>>
>>Here is our first attempt at writing the doc.
>>< 
>>http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp>http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp
>> 
>>
>>Alternatively, you can
>>
>>c:\source> svn co 
>><http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core
>> 
>>>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core
>>c:\source> svn co < 
>>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main>
>> 
>>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main
>>c:\source> svn co < 
>>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web
>> 
>>
>>Then
>>c:\source>cd blindside-main
>>c:\source\blindside-main> mvn install (you must have maven 
>>installed/configured)
>>c:\source\blindside-main> cd ../blindside-web
>>c:\source\blindside-web> mvn jetty:run 
>>
>> From your browser, visit 
>> <http://localhost:8080/conference>http://localhost:8080/conference
>
>Does the Blindside run in linux full installation? 
>
> 
>Not sure exactly what you mean. 

Since I had a fast look over the installation docs which are talking about 
cygwin
and other things running on windows, I was supposing that it run on Windows.

>It is a J2EE/Ajax web application. So if you've got Tomcat 5.x running on 
>Linux, you can deploy the war file into it.

Yep ;-)!

> I'll upload a WAR file (blindside.war) into our server tonight or tomorrow. 
> That way you can download it, put it into your Tomcat's webapps directory, 
> modify the blindside.properties file and off you go.

That's will be really great and fast to test :-)!

> 
>Richard
> 
>
>>
>>Please let me know if you run into problems.
>
>Ok! I'll have a look on it, maybe in the middle/end of this week. 
>
>>
>>Richard
>>
>
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Re: [asterisk-users] RE: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

[Matthew J. Roth]

JR Richardson wrote:

Do you get any errors at max call capacity about "too many open files"?  You
may try increasing your file descriptors.

JR,

Thanks for the response, but I have the maximum number of open files 
available to Asterisk set to 65536.


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Asterisk and cell phones

[Olivier]

Hello,

PBX vendors used to sell software extensions providing enterprise services
to cell phones.

Main features were :
- 4 digit dialing or directory access,
- call forwarding,
- unified messaging.

Now that WiFi and dual mode cell phones get more popular, these Java-based
software should be more successful on the market as bandwidth costs
decrease.

My question are :
1. How Asterisk would compare on cell phone integration with PBX or IPBX
vendors ?
2. Is there now a way to enhance high end cell phones with third party
software to gain access to Asterisk features ?
3. More specifically, has anyone tested Nokia EXX dialing capabilities ?
Would it be easy to customise some callback features  with such phones
For example :
you dial a number and hit a home-made Callback key : your phone then dials a
callback server number instead, hangs up, waits for this callback server to
call and lastly replies the real number you would like to be connected with
(this could be useful when receiving charges are (much) cheaper than
emitting charges).

Best regards
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Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks - Correction

[Matthew J. Roth]

Luki wrote:

Perhaps a naive question, but how does 0.137% CPU utilization per call
equal 1735 MHz per call?

If 1735 MHz / 0.137% = 1735 MHz / 0.00137 => 1266423 MHz at 100%
utilization ??! Even with 4 CPUs, those would be 316 GHz CPUs.

I think you meant:
Average CPU utilization per call: 0.137% (~17 MHz)

Luki,

You are absolutely right.  Thank you for pointing out and correcting my 
mistake.


The corrected statistics are below.  Note that the MHz per call 
statistic is calculated with the following formula:


MHzPerCall = (numCPUs * CPUspeed) * (avgCPUperCall * .01)

Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


The Numbers (Corrected)
---

DC - Incoming SIP to the Playback() application
===
calls   %user   %system   %iowait %idle
   00.00  0.01  0.01 99.98
   10.02  0.04  0.00 99.94
   20.02  0.06  0.00 99.92
   30.03  0.11  0.00 99.86
   40.04  0.13  0.00 99.83
   50.05  0.16  0.00 99.80
   60.05  0.20  0.00 99.75
   70.07  0.24  0.00 99.70
   80.07  0.25  0.00 99.67
   90.08  0.27  0.00 99.65
  100.09  0.33  0.00 99.58

Average CPU utilization per call: 0.040% (~9.60 MHz)

SC - Incoming SIP to the Playback() application
===
calls   %user   %system   %iowait %idle
   00.01  0.02  0.00 99.98
   10.02  0.10  0.00 99.88
   20.03  0.17  0.00 99.80
   30.06  0.21  0.00 99.73
   40.08  0.28  0.00 99.63
   50.10  0.34  0.01 99.55
   60.11  0.48  0.00 99.41
   70.14  0.49  0.00 99.37
   80.16  0.57  0.00 99.28
   90.17  0.63  0.01 99.19
  100.18  0.75  0.00 99.07

Average CPU utilization per call: 0.091% (~11.52 MHz)

DC - Incoming SIP to the Queue() application - In queue
===
calls   %user   %system   %iowait %idle
   00.00  0.01  0.00 99.99
   10.01  0.03  0.00 99.96
   20.01  0.05  0.00 99.94
   30.01  0.08  0.00 99.91
   40.02  0.10  0.00 99.88
   50.03  0.12  0.00 99.84
   60.04  0.16  0.00 99.80
   70.03  0.17  0.00 99.80
   80.04  0.20  0.00 99.76
   90.03  0.22  0.00 99.75
  100.05  0.27  0.00 99.68

Average CPU utilization per call: 0.031% (~7.44 MHz)

SC - Incoming SIP to the Queue() application - In queue
===
calls   %user   %system   %iowait %idle
   00.02  0.02  0.00 99.96
   10.03  0.07  0.00 99.91
   20.03  0.13  0.00 99.83
   30.04  0.18  0.00 99.78
   40.05  0.23  0.00 99.72
   50.06  0.27  0.00 99.67
   60.07  0.33  0.00 99.60
   70.09  0.38  0.00 99.53
   80.09  0.40  0.00 99.51
   90.11  0.46  0.01 99.43
  100.11  0.48  0.00 99.41

Average CPU utilization per call: 0.055% (~6.97 MHz)

DC - Incoming SIP to the Queue() application - Bridged to an agent
==
calls   %user   %system   %iowait %idle
   00.00  0.01  0.00 99.99
   10.01  0.06  0.00 99.93
   20.02  0.14  0.00 99.84
   30.03  0.16  0.00 99.81
  
Average CPU utilization per call: 0.060% (~14.40 MHz)
 
SC - Incoming SIP to the Queue() application - Bridged to an agent

==
calls   %user   %system   %iowait %idle
   00.01  0.02  0.00 99.98
   10.02  0.16  0.00 99.82
   20.04  0.28  0.00 99.68
   30.07  0.36  0.00 99.57

Average CPU utilization per call: 0.137% (~17.35 MHz)

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Re: [asterisk-users] Divitas

[Yuan LIU]

From: EdPimentl <[EMAIL PROTECTED]>
Date: Sun, 27 May 2007 16:12:09 -0400

There will be a number of companies set to offer similar services.
In 3 months we will have a 24 port SIP<->GSM<->SKYPE gateway

-E

On 5/27/07, Dean Collins <[EMAIL PROTECTED]> wrote:


 I was cleaning through some old IT magazines this long weekend when I
came across a company called Divitas in the April 30th edition of Network
Computing.

I've never heard of them but has anyone else heard of them?

Basically they have a call control appliance that can deliver centrally
held up calls between not only GSM but also redirect the call to a wifi
hotspot if you are in range. It seems like a neat concept that shouldn't
necessarily be beyond the capabilities of Asterisk (apart from the fact 
that

the end Win Mobile 5 / Symbian handset would need some type of client).

Any thoughts?

At $550 per seat looks an expensive way to transfer calls between networks
but I've never seen another CPE piece of equipment that can do this.


According to another IT magzaine, Divitas indeed uses Asterisk.  But Divitas 
does not seem to be a pure CPE solution.  That may be why they could charge 
a premium.


Yuan Liu


http://www.divitas.com/products

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

[image: Call 
Button]



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Re: [asterisk-users] Blindside Web Conferencing

[Richard Alam]

On 5/28/07, Roberto Fichera <[EMAIL PROTECTED]> wrote:


At 19.19 28/05/2007, you wrote:


>On 5/28/07, Roberto Fichera <
[EMAIL PROTECTED]> wrote:
>At 17.09 28/05/2007, Richard Alam wrote:
>>Yes, we have some downloadable code. We are in the process of completing
the instructions (build/deploy/etc.).
>>
>>Code is located here.
>><<
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk
>
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk>

http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk
>
>How can I get it?
>
>Hi,
>
>
>Here is our first attempt at writing the doc.
><
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp
>
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp
>
>Alternatively, you can
>
>c:\source> svn co <
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core
>
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core
>c:\source> svn co <
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main
>
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main
>c:\source> svn co <
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web
>
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web
>
>Then
>c:\source>cd blindside-main
>c:\source\blindside-main> mvn install (you must have maven
installed/configured)
>c:\source\blindside-main> cd ../blindside-web
>c:\source\blindside-web> mvn jetty:run
>
> From your browser, visit 
http://localhost:8080/conference

Does the Blindside run in linux full installation?



Not sure exactly what you mean. It is a J2EE/Ajax web application. So if
you've got Tomcat 5.x running on Linux, you can deploy the war file into it.

I'll upload a WAR file (blindside.war) into our server tonight or tomorrow.
That way you can download it, put it into your Tomcat's webapps directory,
modify the blindside.properties file and off you go.

Richard




>Please let me know if you run into problems.

Ok! I'll have a look on it, maybe in the middle/end of this week.

>
>Richard
>

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Re: [asterisk-users] Blindside Web Conferencing

[Roberto Fichera]

At 19.19 28/05/2007, you wrote:


>On 5/28/07, Roberto Fichera <[EMAIL PROTECTED]> 
>wrote: 
>At 17.09 28/05/2007, Richard Alam wrote:
>>Yes, we have some downloadable code. We are in the process of completing the 
>>instructions (build/deploy/etc.). 
>>
>>Code is located here.
>><http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk>
>> 
>>http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk
>
>How can I get it?
>
>Hi,
>
> 
>Here is our first attempt at writing the doc.
>http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp
> 
>Alternatively, you can
> 
>c:\source> svn co 
>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core
>c:\source> svn co 
>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main
>c:\source> svn co 
>http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web
> 
>Then
>c:\source>cd blindside-main
>c:\source\blindside-main> mvn install (you must have maven 
>installed/configured)
>c:\source\blindside-main> cd ../blindside-web
>c:\source\blindside-web> mvn jetty:run
> 
> From your browser, visit 
> http://localhost:8080/conference

Does the Blindside run in linux full installation?

> 
>Please let me know if you run into problems.

Ok! I'll have a look on it, maybe in the middle/end of this week.

> 
>Richard
> 
>
> 
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Roberto Fichera. 

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[asterisk-users] Octasic echo cancellation

[Sebastien Leclere]

Hi,

I'm currently testing SoftEcho, an echo cancellation software for 
Asterisk from Octasic. I noticed an important increase of the quality of 
my coms, but I still have a few echo problems.


There is an ERL parameter which corresponds to an initial ERL value 
probably to optimize the echo training or something in that kind.


Is there a way to monitor, using one of the zttools, the instant ERL 
value, to be able to set this parameter correctly?


In that purpose, I would like to know where do the echo cancellation 
take place in the communication chain, is it after or before the 
amplification  by the Rx/Tx gain parameters? Will this gain apply on the 
echo cancelled signal or on the gross signal?


Anybody with experience on this product could give me some advice in the 
aim of removing all trace of echo?



Thanks by advance

Sebastien
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Re: [asterisk-users] Blindside Web Conferencing

[Richard Alam]

Hi Steve,

Yes, we are looking for that. Do you know of any projects that provides
those? I know one written in TCL/TK.

Thanks.

Richard


On 5/28/07, Steve Totaro <[EMAIL PROTECTED]> wrote:


 Sounds cool.  You could probably use some code from the various open
source jabber clients that allow for shared whiteboard and pushing URLs too.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB

  --

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Richard Alam
*Sent:* Monday, May 28, 2007 10:18 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Blindside Web Conferencing



Hello,



We are creating a web-based conferencing application using Asterisk as the
voice conferencing server.



This as an open source project. We are trying to determine if there
is interest of the community and perhaps work together to improve the
application.



Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
(mute/unmute/kick), chat, participants can raise/lower hand to indicate she
wants to talk when muted.



Please visit http://present.blindsideproject.org (
http://present.sce.carleton.ca if that doesn't work) to try it out. Click
on the "requirements" link for instructions on how to setup your Idefisk.



Project website is at http://www.blindsideproject.org (
http://www.blindsideproject.org:8080 if that doesn't work). We are
requesting the University permissions to configure DNS/firewall, so please
bear with us.



Please send feedback/suggestions to this mailing list. We're also trying
to setup our own. Hope the moderators of this list don't mind.



Hoping for your feedbacks.



Thanks.



Blindside Project Team

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Re: [asterisk-users] Blindside Web Conferencing

[Richard Alam]

On 5/28/07, Roberto Fichera <[EMAIL PROTECTED]> wrote:


At 17.09 28/05/2007, Richard Alam wrote:
>Yes, we have some downloadable code. We are in the process of completing
the instructions (build/deploy/etc.).
>
>Code is located here.
><
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk
>
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk

How can I get it?

Hi,



Here is our first attempt at writing the doc.
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideWebApp

Alternatively, you can

c:\source> svn co
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-core
c:\source> svn co
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-main
c:\source> svn co
http://www.blindsideproject.org:8080/svn/repos/blindside/conference/trunk/blindside-web

Then
c:\source>cd blindside-main
c:\source\blindside-main> mvn install (you must have maven
installed/configured)
c:\source\blindside-main> cd ../blindside-web
c:\source\blindside-web> mvn jetty:run


From your browser, visit http://localhost:8080/conference


Please let me know if you run into problems.

Richard
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[asterisk-users] Multiple TDM400p cards in one machine -- no longer an issue?

[Chris Earle]

Hi all,

Years ago, I was pretty sure attempting to use two TDM400p cards in one
machine was recommended against by Digium ... probably because the cards
couldn't hack it, and/or interrupt problems etc

I have seen some posts recently that seem to indicate it is in fact possible
these days thanks to some updated firmware perhaps? . I just need to
have two in the server because the 4 ports aren't enough ...

I'd rather just expand by one card rather than get a TDM2400 (or TDM800??)

Anyone had recent success/failure with this sort of thing?


--
Chris Earle



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[asterisk-users] Astmanproxy

[voip crazy]

Hello all,

Some of you are using astmanproxy with asttapi or activa TSP?
How does you make to work?

Thanks

VoipCrazy
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Re: [asterisk-users] Meet me

[Jaswinder Singh]

change conf => 222
to conf => 222
( remove | )

I had same problem as freepbx always put | removing it fixed the problem
On 29/05/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:





I am using asterisk 1.4.4 now and facing a problem with meetme,the code  I
was using with asterisk 1.2 is not functioning with 1.4 ,my code is

conf => 222| at meetme.conf

at meet_me_additional



like this

exten => 21,1,MeetMe(21,dq)

exten => 21,2,Playback(beep)



or this

exten => 222,1,GotoIfTime(*|mon-sun|08-08|may-may?223,1)

exten => 222,n,Playback(vm-goodbye)

exten => 222,n,Hangup

exten =>
STARTMEETME,1,MeetMe(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN})

exten => STARTMEETME,n,Hangup

exten => h,1,Hangup

 exten => 223,1,Set(MEETME_ROOMNUM=222)

 exten => 223,n,GotoIf($[${DIALSTATUS} = ANSWER]?READPIN)

 exten => 223,n,Answer

exten => 223,n,Wait(1)

 exten => 223,n(READPIN),Read(PIN,enter-conf-pin-number,,)

 exten => 223,n,GotoIf($[foo${PIN} = foo]?USER)

 exten => 223,n,GotoIf($[${PIN} = ]?ADMIN)

 exten => 223,n,Playback(conf-invalidpin)

 exten => 223,n,Goto(READPIN)

 exten => 223,n(ADMIN),Set(MEETME_OPTS=aAwciMs)

 exten => 223,n,Goto(STARTMEETME,1)

 exten => 223,n(USER),Set(MEETME_OPTS=ciMs)

 exten => 223,n,Goto(STARTMEETME,1)





please guide me



 
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RE: [asterisk-users] Bottom line on fax reception

[Steve Totaro]

Someone already answered this question.  The answer is no, it does not
work by your definition of production ready.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of shadowym
> Sent: Monday, May 28, 2007 11:20 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Bottom line on fax reception
> 
> Anybody??
> 
> -Original Message-
> From: shadowym [mailto:[EMAIL PROTECTED]
> Sent: Thursday, May 24, 2007 9:35 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Bottom line on fax reception
> 
> 
> 
> So what is the bottom line?  Does it work or not.  I've heard stories
it
> works, it doesn't work, it kinda sorta works when it's not raining out
> side.
> Everything under the rainbow.
> 
> What's the bottom line with recent updates on 1.2.x?  Is it production
> ready
> for fax?  By production ready I mean that it just works all the time
and
> doesn't need any babysitting.  Do I have to worry about dropped lines,
> sometimes not detecting incoming fax toneyada yada.
> 
> I know I don't have to use fax on Asterisk but I really want to for
> various
> reasons.  Mostly incoming but outgoing is a nice to have.  Should I
use an
> addon package and if so which one?  Any help would be appreciated.
> 
> 
> 
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RE: [asterisk-users] Blindside Web Conferencing

[Steve Totaro]

Sounds cool.  You could probably use some code from the various open
source jabber clients that allow for shared whiteboard and pushing URLs
too.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com 
KB3OPB
  

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Alam
Sent: Monday, May 28, 2007 10:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Blindside Web Conferencing

 

Hello,

 

We are creating a web-based conferencing application using Asterisk as
the voice conferencing server. 

 

This as an open source project. We are trying to determine if there is
interest of the community and perhaps work together to improve the
application.

 

Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
(mute/unmute/kick), chat, participants can raise/lower hand to indicate
she wants to talk when muted. 

 

Please visit http://present.blindsideproject.org
(http://present.sce.carleton.ca if that doesn't work) to try it out.
Click on the "requirements" link for instructions on how to setup your
Idefisk. 

 

Project website is at http://www.blindsideproject.org
(http://www.blindsideproject.org:8080 if that doesn't work). We are
requesting the University permissions to configure DNS/firewall, so
please bear with us. 

 

Please send feedback/suggestions to this mailing list. We're also trying
to setup our own. Hope the moderators of this list don't mind.

 

Hoping for your feedbacks.

 

Thanks.

 

Blindside Project Team

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Re: [asterisk-users] Blindside Web Conferencing

[Roberto Fichera]

At 17.09 28/05/2007, Richard Alam wrote:
>Yes, we have some downloadable code. We are in the process of completing the 
>instructions (build/deploy/etc.).
> 
>Code is located here.
>http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk

How can I get it?

> 
>Partial docs is located here.
>http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideDevelopment
> 
>Richard
>
> 
>On 5/28/07, Dean Collins <[EMAIL PROTECTED]> wrote: 
>
>I think this is a great potential application for Asterisk – I couldn't 
>actually determine if/where you had a downloadable POC or if it was still just 
>in development conceptualization at the moment. 
>
>Either way keep up the good work and put a paypal tip jar up once you have 
>something people can actually use. 
>
> 
>
> 
>
>Regards,
>
>Dean Collins
>Cognation Pty Ltd
>[EMAIL PROTECTED] 
>+1-212-203-4357 Ph
>+61-2-9016-5642 (Sydney in-dial).
>
>
> 
>
> 
>
>--
>From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] 
>On Behalf Of Richard Alam
>Sent: Monday, 28 May 2007 10:18 AM
>To: asterisk-users@lists.digium.com
>Subject: [asterisk-users] Blindside Web Conferencing
>
> 
>
>Hello,
>
> 
>
>We are creating a web-based conferencing application using Asterisk as the 
>voice conferencing server. 
>
> 
>
>This as an open source project. We are trying to determine if there is 
>interest of the community and perhaps work together to improve the 
>application. 
>
> 
>
>Using the web application, you can upload your powerpoint presentation, manage 
>the participants in the conference thru the web interface (mute/unmute/kick), 
>chat, participants can raise/lower hand to indicate she wants to talk when 
>muted. 
>
> 
>
>Please visit 
>http://present.blindsideproject.org 
>(http://present.sce.carleton.ca if that doesn't work) to try it out. Click on 
>the "requirements" link for instructions on how to setup your Idefisk. 
>
> 
>
>Project website is at 
>http://www.blindsideproject.org 
>(http://www.blindsideproject.org:8080 if that doesn't work). We are requesting 
>the University permissions to configure DNS/firewall, so please bear with us. 
>
> 
>
>Please send feedback/suggestions to this mailing list. We're also trying to 
>setup our own. Hope the moderators of this list don't mind. 
>
> 
>
>Hoping for your feedbacks.
>
> 
>
>Thanks.
>
> 
>
>Blindside Project Team
>
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>  
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Roberto Fichera. 

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RE: [asterisk-users] Bottom line on fax reception

[shadowym]

Anybody?? 

-Original Message-
From: shadowym [mailto:[EMAIL PROTECTED] 
Sent: Thursday, May 24, 2007 9:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Bottom line on fax reception


 
So what is the bottom line?  Does it work or not.  I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.

What's the bottom line with recent updates on 1.2.x?  Is it production ready
for fax?  By production ready I mean that it just works all the time and
doesn't need any babysitting.  Do I have to worry about dropped lines,
sometimes not detecting incoming fax toneyada yada.  

I know I don't have to use fax on Asterisk but I really want to for various
reasons.  Mostly incoming but outgoing is a nice to have.  Should I use an
addon package and if so which one?  Any help would be appreciated. 



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Re: [asterisk-users] Blindside Web Conferencing

[Richard Alam]

Yes, we have some downloadable code. We are in the process of completing the
instructions (build/deploy/etc.).

Code is located here.
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser/conference/trunk

Partial docs is located here.
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/wiki/BlindsideDevelopment

Richard


On 5/28/07, Dean Collins <[EMAIL PROTECTED]> wrote:


 I think this is a great potential application for Asterisk – I couldn't
actually determine if/where you had a downloadable POC or if it was still
just in development conceptualization at the moment.

Either way keep up the good work and put a paypal tip jar up once you have
something people can actually use.





Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

[image: Call 
Button]



  --

*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Richard Alam
*Sent:* Monday, 28 May 2007 10:18 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Blindside Web Conferencing



Hello,



We are creating a web-based conferencing application using Asterisk as the
voice conferencing server.



This as an open source project. We are trying to determine if there
is interest of the community and perhaps work together to improve the
application.



Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
(mute/unmute/kick), chat, participants can raise/lower hand to indicate she
wants to talk when muted.



Please visit http://present.blindsideproject.org (
http://present.sce.carleton.ca if that doesn't work) to try it out. Click
on the "requirements" link for instructions on how to setup your Idefisk.



Project website is at http://www.blindsideproject.org (
http://www.blindsideproject.org:8080 if that doesn't work). We are
requesting the University permissions to configure DNS/firewall, so please
bear with us.



Please send feedback/suggestions to this mailing list. We're also trying
to setup our own. Hope the moderators of this list don't mind.



Hoping for your feedbacks.



Thanks.



Blindside Project Team

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RE: [asterisk-users] Blindside Web Conferencing

[Dean Collins]

I think this is a great potential application for Asterisk - I couldn't
actually determine if/where you had a downloadable POC or if it was
still just in development conceptualization at the moment.

Either way keep up the good work and put a paypal tip jar up once you
have something people can actually use. 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

    
 
 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Alam
Sent: Monday, 28 May 2007 10:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Blindside Web Conferencing

 

Hello,

 

We are creating a web-based conferencing application using Asterisk as
the voice conferencing server. 

 

This as an open source project. We are trying to determine if there is
interest of the community and perhaps work together to improve the
application.

 

Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
(mute/unmute/kick), chat, participants can raise/lower hand to indicate
she wants to talk when muted. 

 

Please visit http://present.blindsideproject.org (
http://present.sce.carleton.ca if that doesn't work) to try it out.
Click on the "requirements" link for instructions on how to setup your
Idefisk. 

 

Project website is at http://www.blindsideproject.org (
http://www.blindsideproject.org:8080 if that doesn't work). We are
requesting the University permissions to configure DNS/firewall, so
please bear with us. 

 

Please send feedback/suggestions to this mailing list. We're also trying
to setup our own. Hope the moderators of this list don't mind.

 

Hoping for your feedbacks.

 

Thanks.

 

Blindside Project Team

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[asterisk-users] Blindside Web Conferencing

[Richard Alam]

Hello,

We are creating a web-based conferencing application using Asterisk as the
voice conferencing server.

This as an open source project. We are trying to determine if there
is interest of the community and perhaps work together to improve the
application.

Using the web application, you can upload your powerpoint presentation,
manage the participants in the conference thru the web interface
(mute/unmute/kick), chat, participants can raise/lower hand to indicate she
wants to talk when muted.

Please visit http://present.blindsideproject.org (
http://present.sce.carleton.ca if that doesn't work) to try it out. Click on
the "requirements" link for instructions on how to setup your Idefisk.

Project website is at http://www.blindsideproject.org (
http://www.blindsideproject.org:8080 if that doesn't work). We are
requesting the University permissions to configure DNS/firewall, so please
bear with us.

Please send feedback/suggestions to this mailing list. We're also trying to
setup our own. Hope the moderators of this list don't mind.

Hoping for your feedbacks.

Thanks.

Blindside Project Team
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Re: [asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000

[Olle E Johansson]

25 maj 2007 kl. 06.40 skrev JK:


Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to  
set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not  
working.
In our scenario  the SP is sending call to our ser server and ser  
is forwarding the call to asterisk. In the asterisk debug I can see  
the DTMF keys are coming but ivr does not recognice those keys at  
all. I can see this in the debug. We are using ulaw and alaw for  
codec.


May 24 20:14:00 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at  
XXX.XXX.XXX.XXX
May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at  
XXX.XXX.XXX.XXX
May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at  
XXX.XXX.XXX.XXX



Voice part works great. I mean if I forward that call to asterisk  
sip user we can talk.
Every thing is working great with other SP. The only difference I  
can see is the rtpmap:101 telephone-event/8000.

With the working SP the rtpmap is rtpmap:100 telephone-event/8000.
Your debug did not have any SIP messages. I need to see the INVITE  
and the 200 OK. Thanks.


/O

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RE: [asterisk-users] Zonbu

[Matthew Rubenstein]

How much does a Patton NanoServ 607x cost? Their page has no price, an
inactive "Ordering" tab, Google doesn't have ("nanoserv 6070" price) in
its index (except a couple unresponsive del.ic.ious pages). PingTel
announce a SIPxNano based on it, for "under $1000" in 2006Q3:
http://www.pingtel.com/page.php?id=70&view=117 . Is there pricing for
just the HW without whatever bundled SW or service these telcos are
bundling/subsidizing it with?


On Sun, 2007-05-27 at 19:51 -0700,
[EMAIL PROTECTED] wrote:
> Date: Sun, 27 May 2007 23:18:26 -0300
> From: "Gustavo Cordeiro" <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] Zonbu
> To: asterisk-users@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=iso-8859-1; format=flowed
> 
> 
>   $99,00 for one box, but you need a subscription plan...
> 
>   "Zonbu is $99 with a two-year subscription plan. With month to month
> plan, 
> Zonbu is $249."
> 
> 
> Sds,
> Gustavo
> 
> >From: "Nabeel Jafferali" <[EMAIL PROTECTED]>
> >Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial 
> >Discussion
> >To: "'Asterisk Users Mailing List - Non-Commercial 
> >Discussion'"
> >Subject: RE: [asterisk-users] Zonbu
> >Date: Sun, 27 May 2007 17:35:20 -0400
> >
> >Looks like a rebadged Patton 6075 to me:
> >
> >http://www.patton.com/products/pe_products.asp?category=337
> >
> >Nabeel
> >
> > > -Original Message-
> > > From:
> [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Dean Collins
> > > Sent: May 27, 2007 11:53 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [asterisk-users] Zonbu
> > >
> > > I just came across www.Zonbu.com   it's a
> > > fanless box about the size of a paperback book. It has no hard
> drive
> > > but runs it's Linux OS on a flash card - relying on document
> storage
> > > from an online service (rebadged Amazon S3).
> > >
> > > http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html
> > >
> > >
> > >
> > >
> > >
> > > I wonder who's going to be the first to hack an asterisk server
> onto
> > > this thing?
> > >
> > > At $99 it's a hell of an option for a fanless Asterisk server.
> > >
> > >
> > >
> > > Regards,
> > >
> > > Dean Collins
> > > [EMAIL PROTECTED]
> > > +1-212-203-4357 Ph
> > > +61-2-9016-5642 (Sydney in-dial).
> > >
> > > Call Button
> 
> > >  
-- 

(C) Matthew Rubenstein

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Re: [asterisk-users] Cisco remote reboot

[Daryl Jurbala]

I've fired a script from an AGI-BIN to accomplish that.

Try this one:

#!/usr/bin/perl
# mk 2004  feel free to distribute
#  [EMAIL PROTECTED], _Vile
#  perl script to reboot phones
#  try telnetting to your phone, first.
#
use Net::Telnet ();

$phone_ip = shift;

# Your Cisco 79xx prompt
$prompt = "Enter Your Prompt Here";

# Your Password
$password = "xx";

# Reset Command
$command = "reset";

if ($phone_ip eq "all")
{
reboot("xxx.xx.x.xx",$password,$command,$prompt);
reboot("xxx.xx.x.xx",$password,$command,$prompt);
reboot("xxx.xx.x.xx",$password,$command,$prompt);
reboot("xxx.xx.x.xx",$password,$command,$prompt);
} elsif ($phone_ip eq "") {
print "Enter an IP or 'all' for All.";
} else {
reboot($phone_ip,$password,$command,$prompt);
}

exit;

sub reboot{

my ($ip,$password,$command,$prompt) = @_;

$t = new Net::Telnet;
$t->open($ip);

$t->waitfor('/Password :.*$/');
$t->print($password);

$t->waitfor('/'.$prompt.'>.*$/');
$t->print($command);

}


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[asterisk-users] DTMFToText Installation process

[rajesh koniki]

Hi,
I was looking for a way to pass alphanumeric variables to asterisk via
the keypad, found this application app_dtmftotext.c , And I already tried 
with 'spandsp' application for this. But I am getting errors.[I followed the 
instructions at http://www.soft-switch.org/installing-spandsp.html].


Can anybody be of help Me on this getting DTMFToText() application on 
asterisk with the help of app_dtmftotext.c and/or spandsp application is 
appreciated.


Regards
K.Rajesh.

_
Voice your questions and our experts will answer them 
http://content.msn.co.in/Lifestyle/AskExpert/Default01.htm


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[asterisk-users] Advanced Billing System for Asterisk - MOR v0.4 released

[Mindaugas Kezys]

Hello,

We are proudly to present new version of our billing system MOR v0.4

What's new in MOR FREE v0.4

* Extended stability and reliability
* Extended configuration options for clients and providers
* User blocking
* Prepaid support
* Increased security
* New tariff/rating engine
* Registration
* PayPal integration
* Authorization by IP
* Extended calls view


What's new in MOR PRO v0.4

* Extended Calling Card engine
* CDR import from CSV file
* Device grouping
* Country Stats
* Providers' Stats
* Auto-Dialer
* Callback
* Click2Call
* Invoice generation
* Custom Rates
* Localization


More info in: http://www.kolmisoft.com/mor

Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com

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Re: [asterisk-users] CDR not recording accountcode on SIP Response 302 Call Forward From Phone

[Steve Murphy]

On Fri, 2007-05-25 at 17:17 -0500, JR Richardson wrote:
> Hi All,
> 
> Call comes into Asterisk
> Asterisk answers and Dials SIP Phone
> SIP phone has call forward enabled to a long distance number
> Asterisk receives a SIP response 302 "Moved Temporarily" back from phone
> Asterisk then forwards inbound call to 'Local/[EMAIL PROTECTED]' thanks to 
> phone
> 
> 2 problems with the CDR:
> 
> 1. intermittent 'bill sec' accuracy, sometimes 0 even when the call
> was answered and many minutes usage on call.
> 
> 2. no accountcode is recorded.
> 
> So the implication here is that if a phone user forwards their phone,
> I have no way of tracking the usage or what account the call should be
> billed to.  I have a feeling this is normal behavior for Asterisk as
> no real channel gets invoked with an accountcode parameter, but there
> has got to be something that accounts for this situation.  Does anyone
> have a work around or remedy?
> 
> I'm running 1.2.9.
> 
> Thanks.
> 
> JR

JR--

Good news and bad news.

To be truthful, I personally have doubts that 1.2 will ever be fixed as
far as transfers are concerned.  The changes that will be necessary to
correct the situation will result in non-backwards compatible changes to
the behavior of CDR's. It'll simply muck up everyone who has built CDR
systems to date.

That said, I'm beginning a "final solution" to the problems in 1.4 and
trunk; I see what needs to be done, and how to do it. I think I have a
grip on what to do with Local channels, and masqueraded channels.

But the devil's in the details, and I'll find out how good my guesses
are when I start testing. I'll try to make sure that channel goodies
don't get smudged off the CDR's, like acctcode, etc.

So, hang on, and I'll post notices as to my progress.

murf

-- 
Steve Murphy <[EMAIL PROTECTED]>
Digium

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Re: [asterisk-users] help on asterisk sipp

[Andre Wangler]

Hello!
If ou mean SIPp, the testing tool for the SIP protocol or kind of a call 
generator for Asterisk PBX, have a look at

http://sipp.sourceforge.net/doc/reference.html

cheers

  - Original Message - 
  From: khawla khawla 
  To: asterisk-users@lists.digium.com 
  Sent: Monday, May 28, 2007 3:09 PM
  Subject: [asterisk-users] help on asterisk sipp


   
   Good morning
  I was wondering whether you could help me. I installed sipp on my Asterisk 
server but I don't really understand how does it fonction! Has someone ever 
tried it?
  If you can explain to me the principle, I would be extremely grateful.
  Thank you very much in advance.



--
  Lancez des recherches en toute sécurité depuis n'importe quelle page Web. 
Téléchargez GRATUITEMENT Windows Live Toolbar aujourd'hui ! Essayez-le 
maintenant ! 


--


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[asterisk-users] help on asterisk sipp

[khawla khawla]

  Good morningI was wondering whether you could help me. I
installed sipp on my Asterisk server but I don't really understand how
does it fonction! Has someone ever tried it?If you can explain to me the 
principle, I would be extremely grateful.Thank you very much in advance.
_
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Re: [asterisk-users] vmoutcall

[Tom Rymes]

On May 24, 2007, at 3:28 PM, Doug Lytle wrote:


Paul Aviles wrote:

Hello guys,

I have been looking for a way to call a cell phone after someone  
has left a



This can easily be done with database lookups and .call files

to accomplishing this? Most analog pbx's have this feature and I  
am amazed

Asterisk does not natively.


It can be done natively; within the dial plan.

Doug


An aside, this can also be done using the externnotify option in the  
voicemail.conf file. That option allows you to specify an external  
script that will run when VoiceMailMain() exits. Watch out, because  
(as I just found out yesterday, the hard way), this script is run  
both when a voicemail message is left and when a user logs out of  
voicemail after checking their messages.


From there, create a script in your favorite language (a simple  
shell script ought to work for your purposes) and have it create a  
call file to call out to the user and drop them into a context that  
plays an announcement ("You have new voicemail") and then asks them  
to confirm with a keypress ("Dial 2 for voicemail"). At that point,  
set the IVR up so dialing 2 drops them into Voicemail() and you are  
good to go.


Tom
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Re: [asterisk-users] Asterisk Time Card

[Nitesh Divecha]

What does codec has to do...? I am using G729a

Cheers,
Nitesh



ram wrote:



On 5/26/07, *Nitesh Divecha* <[EMAIL PROTECTED] 
> wrote:


Thanks Shanon and everyones input...

Finally, got the application working as planned with PHPAGI...

Now the only draw back is the voice... I am using text2wav to
prompt all
the questions, but the voice is creepy...

Is their any easier way to replace the text2wav voice with proper
recorded female voice?

Please advice...

 
 
what codec are you using
 
ram
 



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[asterisk-users] Limit outgoing call for sip peer

[Everton Goularth]

Hi All,

I need to limit outgoing calls in my sip peers...
I tried to use "call-limit=1" in these peers in the sip.conf, but it 
didn't work...


Here is my peer configuration in the sip.conf:

[sip.broadvoice.com]
accountcode=broadvoice
type=peer
dynamic=yes
username=MYUSERNAME
fromuser=MYUSERNAME
authname=MYUSERNAME
user=MYUSERNAME
secret=
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
outboundproxy=proxy.nyc.broadvoice.com
insecure=port
disallow=all
allow=gsm
allow=g729
dtmfmode=rfc2833
dtmf=inband
canreinvite=no
context=from-broadvoice
ntp_ip=ntp.broadvoice.com
call-limt=1

Here is my configuration in the extensions.conf:

exten => _9XX[2-5].,1,dial(SIP/01155${EXTEN:[EMAIL PROTECTED],40)
exten => _9XX[2-5].,2,congestion()
exten => _9XX[2-5].,102,busy()


How can I limit calls in my sip peers??


Everton Goularth
Uberlandia - MG - Brazil


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[asterisk-users] include=>context in REALTIME!!!!!!

[Cheikhou DIAW]

hi list,
im setting up a realtime , and while entering datas in my extensions_table
one question came in my mind
how should insert the different includes ligns (includes=>context)in my
database  knowing that
im not going to use the extensions.conf file anymore
should i add thoses lines in the pbx_config table ???

any clue???


thanks for sharing

BR

--
Cheikhou DIAW
intern iTG Software Engineering India Pvt.Ltd
331-332, Nemi Sagar Colony,
Queens Road, Vaishali Nagar,
Jaipur, 302021
Rajasthan, India
Tel: +91-141-2357582
Tel: +91-141-2357583
Mobile:+91 9828825224
msn: [EMAIL PROTECTED]
e-mail:[EMAIL PROTECTED]
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[asterisk-users] Queues with announce

[Andrea Spadaccini]

Hello *,
do queues allow me to set an announce like the A() option of the Dial() cmd?

The announce that I've found is a message that is heard by the caller. I'd like
to send a message to the member of the queue that picks up the call.

Thanks in advance,

-- 
Dott. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
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RE: [asterisk-users] Zonbu

[matteo brancaleoni]

Hi,

On Sun, 2007-05-27 at 17:35 -0400, Nabeel Jafferali wrote:
> Looks like a rebadged Patton 6075 to me:
> 
> http://www.patton.com/products/pe_products.asp?category=337

also patton rebrands that unit.
At Cebit there was plenty of these boxes from .tw manufacturers.

Matteo

-- 
Matteo Brancaleoni
R&D Director
Tel  :+39.02.70633354
Voip :sip:[EMAIL PROTECTED]

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[asterisk-users] Meet me

[Khaled Chehab]

I am using asterisk 1.4.4 now and facing a problem with meetme,the code  I
was using with asterisk 1.2 is not functioning with 1.4 ,my code is 

conf => 222| at meetme.conf 

at meet_me_additional 

 

like this 

exten => 21,1,MeetMe(21,dq)

exten => 21,2,Playback(beep)

 

or this 

exten => 222,1,GotoIfTime(*|mon-sun|08-08|may-may?223,1)

exten => 222,n,Playback(vm-goodbye)

exten => 222,n,Hangup

exten => STARTMEETME,1,MeetMe(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN})

exten => STARTMEETME,n,Hangup

exten => h,1,Hangup

 exten => 223,1,Set(MEETME_ROOMNUM=222)

 exten => 223,n,GotoIf($[${DIALSTATUS} = ANSWER]?READPIN)

 exten => 223,n,Answer

exten => 223,n,Wait(1)

 exten => 223,n(READPIN),Read(PIN,enter-conf-pin-number,,)

 exten => 223,n,GotoIf($[foo${PIN} = foo]?USER)

 exten => 223,n,GotoIf($[${PIN} = ]?ADMIN)

 exten => 223,n,Playback(conf-invalidpin)

 exten => 223,n,Goto(READPIN)

 exten => 223,n(ADMIN),Set(MEETME_OPTS=aAwciMs)

 exten => 223,n,Goto(STARTMEETME,1)

 exten => 223,n(USER),Set(MEETME_OPTS=ciMs)

 exten => 223,n,Goto(STARTMEETME,1)

 

 

please guide me 

 




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Re: [asterisk-users] Asterisk Time Card

[ram]

On 5/26/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:


Thanks Shanon and everyones input...

Finally, got the application working as planned with PHPAGI...

Now the only draw back is the voice... I am using text2wav to prompt all
the questions, but the voice is creepy...

Is their any easier way to replace the text2wav voice with proper
recorded female voice?

Please advice...




what codec are you using

ram
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[asterisk-users] Progress passing problem.

[Adam Rybak]

Hi,

   i have Asterisk 1.2.7.1 and outgoing trunk connected via SIP (this is Cisco
AS5350)and user is connected via sip too.

When user calling out (via AS5350) he receives progress tone generated by
voip-phone not that passing from telco line.

I turned on debug and see that the AS send: 183 Session Progreess but to user is
sent Ringing, not progress.

I have progressinband=never in sip.conf so shouold be transferred.

Where can be a problem?

Regards,
Adam Rybak
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[asterisk-users] ZAPTEL problem

[ram]

Hi

I have 100XP Digium clone card

Installed in my pc

and compiled zaptel and asterisk again after installing the card

but after i  rebooted

i can load zaptel and wcfxo modprobe with out any problem

but when i intiated ztcfg -

i get the following error


Zaptel Configuration
==


1 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)



-

dmesg errors

Zapata Telephony Interface Unloaded
Zapata Telephony Interface Registered on major 196
Zaptel Version: SVN-branch-1.2-r1725 Echo Canceller: KB1
Failed to initailize DAA, giving up...
wcfxo: probe of :00:10.0 failed with error -5

--

lspci

00:0c.1 SCSI storage controller: Adaptec AIC-7896U2/7897U2
00:10.0 Communication controller: Motorola Unknown device 5608



any suggestions

ram
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