Re: [Asterisk-Users] Intel 536ep as a FXO?

[Bruce Ferrell]

As someone who used to adjust hybrids for a living a number of years 
ago, I can tell you, complex impedence matching is only a part of the 
equation.

The most important part is proper gain structure.  If that's wrong no 
there is no way to control echo.  No amount of tweaking of compensation 
networks will bring one into balance... No Convolution processing can 
control it.  On old style equipment i.e. stuff built by Tellabs, the 
gain structure had to be "right" within about .5 DBm0.

Alignment meant dialing up a milliwatt test signal, measuring that 
signal at the 2 wire point and adjusting pads on the module so that the 
4 wire transmit point was at a fixed and correct level.  If memory 
serves, on an analog microwave system, 0 DBm into a module was supposed 
to be -16 DBm on the 4 wire transmit point.  The "picture" below may 
help to clarify:

===

  --->2 wire  TX|
 |  |
0DBm | /--o  -16DBm |
 |/ |
C.O. milliwatt o-x   4 wire |
  \ |
   \--o  +7DBm  |
|
  <--- RX   |
===

So... given that we know the C.O. milliwatt is 0 DBm we also know that 
the signal seen at the point marked 2 wire is the sum of 0DBm minus the 
line loss, usually around 3 to 4 DB.  When that signal passes through 
the hybrid and correctly adjusted associated attenuator it will appear 
as marked and discussed.  Conversly, +7 DBm is inserted at the 4 wire RX 
point and the associated attenuator adjusted so that sufficient signal 
is seen at the 2 wire point at 0 DBm.

The microwave system that connects to the 4 wire point has 23 DB of gain 
so that the layout above can be mirrored for a complete analog 2 wire/4 
wire/2 wire circuit with an overall loss of between 6 to 8 DB.

The old bell specifications called for minimum 12 DB longitudinal loss 
across the 4 wire points for a hybrid on a local circuit and 16 DB for 
long haul.  There were milage specifications, but I don't remember them 
anymore.  Just getting the gain structure right was usually enough to 
meet that requirement.  If not, then we got into a backend adjustment to 
impedance match the 2 wire circuit to the hybrid... Interestingly 
enough, on an in use circuit, the losses and impedances didn't tend to 
change much over a period of years.

I think this has gone on long enough... suffice to say, gain/levels are 
crucial to echo control... It you send is too hot, you WILL have echo 
and I don't care how good your card is.  These principals applied to 
channel banks that I adjusted in olden days as well... Mostly Northern 
Telecom DE4, but others as well.  We used special equipment to measure 
signal levels at the T1 point.  I have to presume E1 equipment is/was 
similar, but I have no experience there.

'nuff said

Steve Underwood wrote:
Andrew Kohlsmith wrote:

[...]

Not at all.  Any of the channel banks I've tested have better echo and 
audio quality than the X100P.  I believe it comes down to the Part68 
interface being better able to accomodate different lines but YMMV.  I 
have never had decent results with an X100P.  All of the tricks and 
hacks you see on the wiki with it are proof that it's a substandard 
card, IMO.
 

If you are trying to do cellular, satellite, VoIP  or any telephony with 
high latency and do not use echo cancellation you are on to a looser. 
Sure, the problem is worse with some interfaces in combination with 
certain lines (you can't separate the two), but echo performance will 
always be lousy without proper echo cancellation. With echo cancellation 
almost any FXO interface should work well. Every cell phone call to the 
PSTN is echo cancelled. Every cheapo or expensive VoIP interface box 
echo cancels. Hybrids of any design are really lousy, and do little more 
than stop howling. You can hand tweak some of them connected to a 
particular line and get great performance. However, they always drift, 
the lines get altered, or in some other way they get screwed up again. 
Echo cancellation is a requirement, not an option.

If the X100P's interface matches a line well it will work well. If it 
matches it badly it will work badly. Same with the channel banks, or any 
other analogue line interface. Almost all use a compromise line match. 
Adaptive line matching is rare. The echo you get is the luck of the 
draw, regardless of what FXO hardware you are using.

Regards,
Steve
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Re: [Asterisk-Users] Intel 536ep as a FXO?

[Andrew Kohlsmith]

> If you are trying to do cellular, satellite, VoIP  or any telephony with
> high latency and do not use echo cancellation you are on to a looser.

I agree.

> Sure, the problem is worse with some interfaces in combination with
> certain lines (you can't separate the two), but echo performance will
> always be lousy without proper echo cancellation. With echo cancellation
> almost any FXO interface should work well. Every cell phone call to the
...
> Echo cancellation is a requirement, not an option.

Again, I would tend to agree with you.

> If the X100P's interface matches a line well it will work well. If it
> matches it badly it will work badly. Same with the channel banks, or any
> other analogue line interface. Almost all use a compromise line match.
> Adaptive line matching is rare. The echo you get is the luck of the
> draw, regardless of what FXO hardware you are using.

This is where I am trying to make my point.  The X100P's hybrid is, IMO, 
substandard.  We don't officially support any other winmodem so I can't say 
that another one would be better.  I'd love to have more supported WinModems 
and comparison data but my needs for low density FXO solutions aren't all 
that great so it's a fight I'm not particularly interested in fighting.

I've used Carrier Access' Access Bank I and their Adit600 FXO modules and they 
both perform FAR better than the X100P, with the Adit600 winning out over the 
ABI.  Now of course there are cost differences -- I never meant to imply that 
they were on equal footing in terms of cost.  I was instead trying to point 
out that the X100P is not "rock solid hw" as indicated by Mr. Matteo to Mr. 
Mazurek, and I gave the literally dozens of posts about poor X100P 
performance *with* the echo cancellations, tip and ring reversal tips and 
various attempts to balance out imbalanced lines as proof of my position.

I haven't seen the FXO modules for the TDM400P, but I really do hope that 
their hybrid will be of much higher quality than that of a $20 rebranded 
WinModem.  Digium is charging for the TDM400P and also for the FXO modules -- 
there should be plenty of money in there for support and still have a better 
hybrid than the X100P.  I hope.

Regards,
Andrew
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[Asterisk-Users] spandsp...can't compile *

[Denis E. Pilon]

Why is it I keep getting this error when trying to get spandsp compiled
with asterisk ?

gcc -O2 -g  -Iinclude -I../include -c -o  app_rxfax.o app_rxfax.c
app_rxfax.c:45: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here
(not in a function)


Denis
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RE: [Asterisk-Users] libspandsp.so.0

[Sam Bingner]

It's worked good for me... Only had a garbled page once when it was a 15
page fax, and that was a few versions ago so I'm not sure if it would do
the same now or not

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Enger
Sent: Sunday, April 18, 2004 5:22 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] libspandsp.so.0


Put it in /etc/ld.so.conf (the path to the lib dir the file is in) then
run ldconfig.

Has anyone had any success with rxfax? Every time I have used it the tiff
file has a garbled page.


On Mon, 2004-04-19 at 11:02, Karl Brose wrote:
> ldconfig
> 
> 
> - Original Message -
> From: <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, April 18, 2004 19:24
> Subject: [Asterisk-Users] libspandsp.so.0
> 
> 
> > I successfully compiled & installed the 
> > spandsp-0.0.1k.tar.gz modules for faxing and 
> > patched the asterisk according to the readme and 
> > rebuilt and installed * but I am getting this 
> > error when attempting to start *. The 
> > libspandsp.so.0 file exists and I have coppied it 
> > to several directories recompiled and have the 
> > same results.   
> > What am I doing wrong? help please 
> >   
> >   [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: 
> > loader.c:239 ast_load_resource: libspandsp.so.0: 
> > cannot open shared object file: No such file or 
> > directory 
> > Apr 18 18:57:20 WARNING[1024]: loader.c:407 
> > load_modules: Loading module app_rxfax.so failed! 
> >  
> > 
> > 
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED] 
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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-- 
Matthew Enger
[EMAIL PROTECTED]
Mob: 0412 463 080
Direct: (03) 9747 4001
X Integration
A Netcruiser Pty Ltd business
Ph: 1300 730 997
Fax: 1300 136 720
-- 
Matthew Enger <[EMAIL PROTECTED]>
Xintegration

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smime.p7s
Description: S/MIME cryptographic signature

[Asterisk-Users] AGI Module

[Robert Jackson]

Hey all,
I'm sorry to bother you with something so trivial, but I seem to
be having an issue with the Asterisk::AGI module.  I am a relative
newbie with Perl so it could be a stupid syntax mistake that I missed.
It seems when I try to execute either the stream_file or the get_data
subs nothing is actually done.  It doesn't seem to stream the files, but
on the console it says it played the file.  If you need more information
please let me know and I will get it to you ASAP.  Here are my source
and exentsions.conf:

Extensions.conf
---
; Test agi extension
exten => 3102,1,Answer
exten => 3102,2,AGI,agi-SQLTest.agi
exten => 3102,3,Hangup


Agi-SQLTest.agi
---
#!/usr/bin/perl

use Asterisk::AGI;
use DBI;

$AGI = new Asterisk::AGI;

# Read * input info.
my %input = $AGI->ReadParse();

# Set SQL server info.
$uid = 'asterisk';
$pwd = 'pbx';
$srv = 'SMARTTCom';

# Print variables for debugging.
$AGI->verbose("AGI Environment Dump:\n",3);
foreach $i (sort keys %input) {
$AGI->verbose(" -- $i = $input{$i}\n",3);
}

# Connect to server using freetds
$AGI->verbose("Connecting to server: $srv",3);
my $dbh = DBI->connect("dbi:Sybase:server=$srv", $uid, $pwd, {PrintError
=> 0});

die "Unable for connect to server $DBI::errstr"
unless $dbh;

$AGI->verbose('Getting account number.',2);

# Test stream_file.  This is for debugging only.
# once fixed remove.
$AGI->stream_file('invalid','12345');

# Trying to get the caller's account number.  This is not working
either.
$accountNumber = $AGI->get_data('agi-enteraccountnumber','15000','5');

# If accountNumber is blank then just queue the caller into the
# 'unverified' queue.
if ('foo' . $accountNumber eq 'foo') {
$AGI->exec('Queue', 'unverified-patientq');
$AGI->verbose('No account number entered', 1);
exit(0);
}

# Run query to get verification data from server.
$AGI->verbose('Executing queury',3);
my $sth = $dbh->prepare("select SSN, [Birth Date], Zip, [Home Phone]
from SEARCHBASE where [Demo#] = $accountNumber");
$sth->execute();

if ($sth->rows == 1) {
@data = $sth->fetchrow_array();
my $ssn = $data[0];
my $dob = $data[1];
my $dob = $data[1];
my $zip = $data[2];
my $homePhone = $data[3];
$AGI->verbose("Data returned: $ssn, $dob, $zip, $homePhone",3)
} else {
$AGI->verbose("Invalid account number: $accountNumber",1);
$AGI->exec('Queue', 'unverified-patientq');
exit(0);
}

$sth->finish;
$dbh->disconnect;

Exit;

Any help would be greatly appreciated.  

Thanks,

Robert Jackson
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Re: [Asterisk-Users] libspandsp.so.0

[Matthew Enger]

Put it in /etc/ld.so.conf (the path to the lib dir the file is in) then
run ldconfig.

Has anyone had any success with rxfax? Every time I have used it the
tiff file has a garbled page.


On Mon, 2004-04-19 at 11:02, Karl Brose wrote:
> ldconfig
> 
> 
> - Original Message - 
> From: <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, April 18, 2004 19:24
> Subject: [Asterisk-Users] libspandsp.so.0
> 
> 
> > I successfully compiled & installed the 
> > spandsp-0.0.1k.tar.gz modules for faxing and 
> > patched the asterisk according to the readme and 
> > rebuilt and installed * but I am getting this 
> > error when attempting to start *. The 
> > libspandsp.so.0 file exists and I have coppied it 
> > to several directories recompiled and have the 
> > same results.   
> > What am I doing wrong? help please 
> >   
> >   [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: 
> > loader.c:239 ast_load_resource: libspandsp.so.0: 
> > cannot open shared object file: No such file or 
> > directory 
> > Apr 18 18:57:20 WARNING[1024]: loader.c:407 
> > load_modules: Loading module app_rxfax.so failed! 
> >  
> > 
> > 
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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-- 
Matthew Enger
[EMAIL PROTECTED]
Mob: 0412 463 080
Direct: (03) 9747 4001
X Integration
A Netcruiser Pty Ltd business
Ph: 1300 730 997
Fax: 1300 136 720
-- 
Matthew Enger <[EMAIL PROTECTED]>
Xintegration

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Re: [Asterisk-Users] Intel 536ep as a FXO?

[Steve Underwood]

Andrew Kohlsmith wrote:

[...]

Not at all.  Any of the channel banks I've tested have better echo and audio 
quality than the X100P.  I believe it comes down to the Part68 interface 
being better able to accomodate different lines but YMMV.  I have never had 
decent results with an X100P.  All of the tricks and hacks you see on the 
wiki with it are proof that it's a substandard card, IMO.
 

If you are trying to do cellular, satellite, VoIP  or any telephony with 
high latency and do not use echo cancellation you are on to a looser. 
Sure, the problem is worse with some interfaces in combination with 
certain lines (you can't separate the two), but echo performance will 
always be lousy without proper echo cancellation. With echo cancellation 
almost any FXO interface should work well. Every cell phone call to the 
PSTN is echo cancelled. Every cheapo or expensive VoIP interface box 
echo cancels. Hybrids of any design are really lousy, and do little more 
than stop howling. You can hand tweak some of them connected to a 
particular line and get great performance. However, they always drift, 
the lines get altered, or in some other way they get screwed up again. 
Echo cancellation is a requirement, not an option.

If the X100P's interface matches a line well it will work well. If it 
matches it badly it will work badly. Same with the channel banks, or any 
other analogue line interface. Almost all use a compromise line match. 
Adaptive line matching is rare. The echo you get is the luck of the 
draw, regardless of what FXO hardware you are using.

Regards,
Steve
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Re: [Asterisk-Users] VoIP SIP SoftPhone Recommendations

[Edmund]

I'm using Linphone. It works pefectly with *.

Edmund

JORA ROME wrote:

What SoftPhone working very well with *? S.O. is Debian Linux
Thanks for your comments.
JRR

_
MSN Amor: busca tu ½ naranja http://latam.msn.com/amor/
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RE: [Asterisk-Users] Incoming on Zap

[Juan M. Fach]

Hello Everybody
 
Is there anybody that knows a out-box Asterisk solution 
that install of the stable features available for *
 
I am refering mainly to web base administration 
features
 
I am intended to use * mainly for IP solutions 

 
Regards
 
John Fach
 

[Asterisk-Users] Does RTP traffic go through Asterisk IP PBX ?

[PTCHEN]

Hello,
 
Is there anybody knows if RTP traffic goes thru Asterisk IP 
PBX?
If it is, it must limit the capacity of Asterisk. Do you 
know the concurrent 
SIP call capacity?  
And Is there any guy modify the source 
code to prevent this?
 
Thanks!
Chunghwa Telecom BTA Tech. 
Lab.E-mail:[EMAIL PROTECTED]
 
 

Re: [Asterisk-Users] Intel 536ep as a FXO?

[Andrew Kohlsmith]

> Any other FXO card will look just like the present one. A winmodem is
> nothing more or less than an FXO card. It deals with the line
> signalling, and analogue conversion and leaves everything else up to the
> software. In the case of a modem that "everything else" is mostly modem
> DSP. In the case of an FXO it is routing and switching. The hardware is,
> however, identical.

I call bullshit and you should know better -- You can match Part68 and still 
have an absolutely horrible interface.  All Part68s aren't created equally, 
and IMO the X100P's is crap.

> I think you are the zealot. You seem to have a kind of "if it isn't
> custom made for my job it must be second rate" attitude.

Not at all.  Any of the channel banks I've tested have better echo and audio 
quality than the X100P.  I believe it comes down to the Part68 interface 
being better able to accomodate different lines but YMMV.  I have never had 
decent results with an X100P.  All of the tricks and hacks you see on the 
wiki with it are proof that it's a substandard card, IMO.

> What is wrong with it? It is a perfectly good FXO card.

See above.

> Well, a TDM400P is essentially just 4 winmodems plugged into a base board.

Well their FXS interfaces first, but I'm not going to get into a semantics war 
with you -- I am positive that the FXO modules will also perform better than 
the X100P.  I haven't had any issues with the FXS interfaces on the TDM400P 
-- the act just like any FXS channel bank I've used.

Regards,
Andrew
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[Asterisk-Users] Accommodating multiple FWD users

[Malcolm Taylor]

I have five SIP users on my * box, each of whom has his own FWD account.
Right now I have my configuration set so that the first user dials the
prefix 8 when calling to an FWD number, the second user dials the prefix 7
and so on.  This way, the FWD user he is calling sees the correct Caller ID
information.

Can anyone suggest a way in which all users could dial the prefix 8 and *
would automatically associate the correct FWD account for the outbound call?

Many thanks,

Malcolm

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Re: [Asterisk-Users] Intel 536ep as a FXO?

[Steve Underwood]

Andrew Kohlsmith wrote:

P.S. and you'll have free installation support from
digium and a rock solid hw made for asterisk.
   

I have and use the X100/101P, TDM400P, T100P, TE410P and TE405P from Digium.

Only a Digium zealot would call the X100/X101P 'rock solid hw' -- Digium is 
reselling a generic WinModem for these cards and they are simply not good 
hardware.  I love the other cards to death -- they are solid and stable and 
Just Work, but please don't blindly endorse anything that comes out with 
a Digium stamp on it -- you're only decreasing the value of Digium's name.
 

Any other FXO card will look just like the present one. A winmodem is 
nothing more or less than an FXO card. It deals with the line 
signalling, and analogue conversion and leaves everything else up to the 
software. In the case of a modem that "everything else" is mostly modem 
DSP. In the case of an FXO it is routing and switching. The hardware is, 
however, identical.

I think you are the zealot. You seem to have a kind of "if it isn't 
custom made for my job it must be second rate" attitude.

They needed a cheap FXO interface for the masses and for now, that's what we 
have.  It's certainly not a good solution, but it is *a* solution.
 

What is wrong with it? It is a perfectly good FXO card.

I am eagerly awaiting proper stocking of the IAXy and an FCC-certified FXO 
module for the TDM400P -- I think those should be Digium's flagship products, 
not a rebranded craptastic WinModem.
 

Well, a TDM400P is essentially just 4 winmodems plugged into a base board.

Regards,
Steve
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RE: [Asterisk-Users] Dynamic recording function?

[mattf]

If you will tolerate using a GUI on a computer while you are on the phone,
astguiclient has one-button click to record and stop:

http://astguiclient.sf.net/

The way we do it is simply by using manager action commands. There are
several ways of triggering them, but it's not that easy to do it by pressing
a button on the phone when you're already on a conversation.

MATT---


-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Sent: Sunday, April 18, 2004 4:26 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dynamic recording function?


Hi Folks,

Yesterday I had need to record a phone conversation. This is not something
I'd ordinarily have to do and so I have not configured my * server to do
any recordings.

When looking for example dialplan stuff I found many examples where the
calls are always recorded when the phone is picked up but none that could
be done dynamicly.

What I'd like to be able to do is press a button and have * start
recording the call from that moment and then either stop when I hang up or
stop when I press another button.

Ideas?


G7LTT/KC2ENI
Mark Phillips
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Re: [Asterisk-Users] libspandsp.so.0

[Steve Underwood]

If you used the default build and install configuration it will install 
the library in /usr/local/lib. Is that in yout library paths?

Regards,
Steve
[EMAIL PROTECTED] wrote:

I successfully compiled & installed the 
spandsp-0.0.1k.tar.gz modules for faxing and 
patched the asterisk according to the readme and 
rebuilt and installed * but I am getting this 
error when attempting to start *. The 
libspandsp.so.0 file exists and I have coppied it 
to several directories recompiled and have the 
same results.   
What am I doing wrong? help please 
 
 [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: 
loader.c:239 ast_load_resource: libspandsp.so.0: 
cannot open shared object file: No such file or 
directory 
Apr 18 18:57:20 WARNING[1024]: loader.c:407 
load_modules: Loading module app_rxfax.so failed!  
 

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Re: [Asterisk-Users] libspandsp.so.0

[Karl Brose]

ldconfig


- Original Message - 
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, April 18, 2004 19:24
Subject: [Asterisk-Users] libspandsp.so.0


> I successfully compiled & installed the 
> spandsp-0.0.1k.tar.gz modules for faxing and 
> patched the asterisk according to the readme and 
> rebuilt and installed * but I am getting this 
> error when attempting to start *. The 
> libspandsp.so.0 file exists and I have coppied it 
> to several directories recompiled and have the 
> same results.   
> What am I doing wrong? help please 
>   
>   [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: 
> loader.c:239 ast_load_resource: libspandsp.so.0: 
> cannot open shared object file: No such file or 
> directory 
> Apr 18 18:57:20 WARNING[1024]: loader.c:407 
> load_modules: Loading module app_rxfax.so failed! 
>  
> 
> 
> 
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Re: [Asterisk-Users] small linux distro to run * in old boxes

[Greg Boehnlein]

On Fri, 9 Apr 2004, Victor Perez wrote:

> Has anybody tried to install * in any of these minimalist linux distros like 
> tinylinux?
> 
> Which linux distro would you use to run * in old P2, P3 boxes?

I run Debian 3.0, testing branch on my P-133 / 16 meg Asterisk Super 
Server. ;)

Did a base install and then added some additional stuff to get 
compilations working. Total install peaks is about 98 megs. Works great. 
Been stable since installation.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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RE: [Asterisk-Users] libspandsp.so.0

[Todd Lieberman]

copy libspandsp.so.0 to /usr/lib/asterisk/modules

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, April 18, 2004 7:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] libspandsp.so.0


I successfully compiled & installed the 
spandsp-0.0.1k.tar.gz modules for faxing and 
patched the asterisk according to the readme and 
rebuilt and installed * but I am getting this 
error when attempting to start *. The 
libspandsp.so.0 file exists and I have coppied it 
to several directories recompiled and have the 
same results.   
What am I doing wrong? help please 
  
  [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: 
loader.c:239 ast_load_resource: libspandsp.so.0: 
cannot open shared object file: No such file or 
directory 
Apr 18 18:57:20 WARNING[1024]: loader.c:407 
load_modules: Loading module app_rxfax.so failed! 
 



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[Asterisk-Users] libspandsp.so.0

[asterisk]

I successfully compiled & installed the 
spandsp-0.0.1k.tar.gz modules for faxing and 
patched the asterisk according to the readme and 
rebuilt and installed * but I am getting this 
error when attempting to start *. The 
libspandsp.so.0 file exists and I have coppied it 
to several directories recompiled and have the 
same results.   
What am I doing wrong? help please 
  
  [app_rxfax.so]Apr 18 18:57:20 WARNING[1024]: 
loader.c:239 ast_load_resource: libspandsp.so.0: 
cannot open shared object file: No such file or 
directory 
Apr 18 18:57:20 WARNING[1024]: loader.c:407 
load_modules: Loading module app_rxfax.so failed! 
 



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[Asterisk-Users] SMS receiving & sending

[Philipp von Klitzing]

Hi there,

just a quick note: recently a very interesting patch made it into the 
bugtracker that those of you interested in SMS might want to have a look 
at:

http://bugs.digium.com/bug_view_page.php?bug_id=0001437

Cheers, Philipp

P.S.: I am not the author, just the messenger. :-)


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Re: [Asterisk-Users] Dynamic recording function?

[John Baker]

Try

http://www.voip-info.org/wiki-Monitor+setup+sample

John

Mark Phillips wrote:
Hi Folks,

Yesterday I had need to record a phone conversation. This is not something
I'd ordinarily have to do and so I have not configured my * server to do
any recordings.
When looking for example dialplan stuff I found many examples where the
calls are always recorded when the phone is picked up but none that could
be done dynamicly.
What I'd like to be able to do is press a button and have * start
recording the call from that moment and then either stop when I hang up or
stop when I press another button.
Ideas?

G7LTT/KC2ENI
Mark Phillips
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Re: [Asterisk-Users] x100p config

[Darren Poulson]

Oops, the link in my previous message should be:

http://www.22balmoralroad.net/modules.php?name=Sections&op=viewarticle&artid=1

Cheers,

Darren.


On Sunday 18 Apr 2004 9:29 pm, Paul Tyreman wrote:
> I don't have any wait commands in my s extention.
>
> I don't use (or need to use since they don't work in the UK) caller display
> on external calls, but I do want to keep it on intenal calls, so is there
> any way to turn it off on exernal calls only ?
>
> One more point, I rebooted my server and when I tried to resart Asterisk
> again, I got an error saying something about no card on d0001 (or something
> similar) and it refused to start.  I had to run "modprobe wcfxo" before I
> could start the server.  Is that normal, or is there something I can do so
> it automaticly decects the card when I turn the server on.
>
> Thanks again for yor help Sean.
>
> Paul.
>
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman
> Posted At: 18 April 2004 20:41
> Posted To: Asterisk-Users
> Conversation: [Asterisk-Users] x100p config
> Subject: RE: [Asterisk-Users] x100p config
>
>
> It could be one of several things.  The two things that come to mind is
> Caller ID and a Wait() statement in your dialplan.  Since the Caller ID
> information is transmitted between the first and second ring, Asterisk has
> to wait for it if Caller ID is enabled.  Other than that, is there a Wait()
> line in your S extension?
>
> Sean
> -Original Message-
> From: Paul Tyreman [mailto:[EMAIL PROTECTED]
> Sent: Sunday, April 18, 2004 2:31 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] x100p config
>
>
> Thanks for your help.  I've got it working now.
>
> Only one problem.  When users from the public network call my server, they
> hear three rings before the phones on my server start ringing.  Is that
> usual, or is it a setting that can be changed ?
>
> Thanks, Paul.
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman
> Posted At: 18 April 2004 19:48 Posted To: Asterisk-Users Subject: RE:
> [Asterisk-Users] x100p config
>
>
> Welcome to the wonderful world of Asterisk!  In the future, you might want
> to make sure that you post in plain text mode instead of HTML. There are
> quite a few people here who are great assets that won't even read if you
> post in HTML.
>
> Your problem has to do with the contexts.  In your zapata.conf file, you
> will see reference to a context for your X100P.  That is the context into
> which calls on that card will be dumped.  If you check your
> extensions.conf, you should find a matching context that will have all of
> the demo stuff in it.  You can either change the demo context to meet your
> needs, or change your zapata.conf to point to a more useful context that
> has just what you want in it.
>
> You might want to read over the info at http://www.voip-info.org. There's a
> lot of good reading there that will help you make the most of Asterisk.
>
> Sean
>
> -Original Message-
> From: Paul Tyreman [mailto:[EMAIL PROTECTED]
> Sent: Sunday, April 18, 2004 1:31 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] x100p config
>
>
> Hi,
>
> I have just installed my first X100P card, and seams to be half working.
>
> You can call the public telephone number which the card is attached to and
> hear some lady telling you about asterisk.  If I dial the extention number
> of the phone I want to call, it connects and it's all good.
>
> However, I have put this line in my extensions.conf:
> [incoming]
> exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)
>
> So it should ring phone one and phone two rather then give that that girls
> voice !  Can anyone tell me what I'm doing wrong ?
>
>
> Also, I have put this in the same extensions.conf file: [outgoing] exten =>
> _0X.,1,Dial,Zap/1/${EXTEN:1}
>
> [sip]
> include => outgoing
>
> Yet I still cannot make outgoing calls, when I dial 0 and the number I want
> to call on the public network.
>
> Any help would be great as I'm starting to pull my hair out !
>
> Thanks, Paul.
> ___
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-- 
Darren Poulson - Unix Admin
PGP Key at: http://www.22balmoralroad.net/~daz/pgp.key
If you hype something and it succeeds, you're a genius -- it wasn't a
hype.  If you hype it and it fails, then it was just a hype.
-- Neil Bogart
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Re: [Asterisk-Users] x100p config

[Darren Poulson]

 
On Sunday 18 Apr 2004 9:29 pm, Paul Tyreman wrote:
> I don't have any wait commands in my s extention.
>
> I don't use (or need to use since they don't work in the UK) caller display
> on external calls, but I do want to keep it on intenal calls, so is there
> any way to turn it off on exernal calls only ?
>
> One more point, I rebooted my server and when I tried to resart Asterisk
> again, I got an error saying something about no card on d0001 (or something
> similar) and it refused to start.  I had to run "modprobe wcfxo" before I
> could start the server.  Is that normal, or is there something I can do so
> it automaticly decects the card when I turn the server on.
>
> Thanks again for yor help Sean.
>
> Paul.
>
>

Hi,

I've been messing around today with * and noticed the delay in ringing too. As 
a previous reply mentioned it is something to do with CallerID. In your 
zapata.conf file, add 

usecallerid=no

to the channel relating to your x100 card. This will turn off the detection of 
callerID on your incoming line, which won't work in the UK with X100p cards. 
I've just chaned my zapata.conf file and there is only a single ring of 
waiting now.

On a related note, if you want caller ID, I've managed to get it working using 
a modem and I've documented it at:

http://www.flapper.net/modules.php?name=Sections&op=viewarticle&artid=1

How do you add to the wiki? There's little or no information that I can find 
regarding the often asked question of UK CID.

Cheers,

Darren.

>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman
> Posted At: 18 April 2004 20:41
> Posted To: Asterisk-Users
> Conversation: [Asterisk-Users] x100p config
> Subject: RE: [Asterisk-Users] x100p config
>
>
> It could be one of several things.  The two things that come to mind is
> Caller ID and a Wait() statement in your dialplan.  Since the Caller ID
> information is transmitted between the first and second ring, Asterisk has
> to wait for it if Caller ID is enabled.  Other than that, is there a Wait()
> line in your S extension?
>
> Sean
> -Original Message-
> From: Paul Tyreman [mailto:[EMAIL PROTECTED]
> Sent: Sunday, April 18, 2004 2:31 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] x100p config
>
>
> Thanks for your help.  I've got it working now.
>
> Only one problem.  When users from the public network call my server, they
> hear three rings before the phones on my server start ringing.  Is that
> usual, or is it a setting that can be changed ?
>
> Thanks, Paul.
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman
> Posted At: 18 April 2004 19:48 Posted To: Asterisk-Users Subject: RE:
> [Asterisk-Users] x100p config
>
>
> Welcome to the wonderful world of Asterisk!  In the future, you might want
> to make sure that you post in plain text mode instead of HTML. There are
> quite a few people here who are great assets that won't even read if you
> post in HTML.
>
> Your problem has to do with the contexts.  In your zapata.conf file, you
> will see reference to a context for your X100P.  That is the context into
> which calls on that card will be dumped.  If you check your
> extensions.conf, you should find a matching context that will have all of
> the demo stuff in it.  You can either change the demo context to meet your
> needs, or change your zapata.conf to point to a more useful context that
> has just what you want in it.
>
> You might want to read over the info at http://www.voip-info.org. There's a
> lot of good reading there that will help you make the most of Asterisk.
>
> Sean
>
> -Original Message-
> From: Paul Tyreman [mailto:[EMAIL PROTECTED]
> Sent: Sunday, April 18, 2004 1:31 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] x100p config
>
>
> Hi,
>
> I have just installed my first X100P card, and seams to be half working.
>
> You can call the public telephone number which the card is attached to and
> hear some lady telling you about asterisk.  If I dial the extention number
> of the phone I want to call, it connects and it's all good.
>
> However, I have put this line in my extensions.conf:
> [incoming]
> exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)
>
> So it should ring phone one and phone two rather then give that that girls
> voice !  Can anyone tell me what I'm doing wrong ?
>
>
> Also, I have put this in the same extensions.conf file: [outgoing] exten =>
> _0X.,1,Dial,Zap/1/${EXTEN:1}
>
> [sip]
> include => outgoing
>
> Yet I still cannot make outgoing calls, when I dial 0 and the number I want
> to call on the public network.
>
> Any help would be great as I'm starting to pull my hair out !
>
> Thanks, Paul.
> ___
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> [EMAIL PROTECTED]
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> update options visit:
>http://lists.digium.com/mailma

[Asterisk-Users] Using PCI cards with a laptop

[James H. Thompson]

I ran accross this product which allows you to connect a PCI card to a laptop with a 
PCMCIA/Cardbus
slot.

http://www.mobl.com/expansion/pci/index.html

It seems like this would allow you to use any of the Digium PCI cards with a laptop.
Has anyone tried something like this?


Jim

James H. Thompson
[EMAIL PROTECTED]

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Re: [Asterisk-Users] Intel 536ep as a FXO?

[Brancaleoni Matteo]

> 
> Only a Digium zealot would call the X100/X101P 'rock solid hw' -- Digium is 
I'm not a zealot , nor endorsed in any way by digium..
for me , with a a lot of X100P installed, is rock solid. never missed a
hit.

> They needed a cheap FXO interface for the masses and for now, that's what we 
> have.  It's certainly not a good solution, but it is *a* solution.
Is a good solution. At least the combination callItAsYouWantcard +
zaptel drivers...
> I am eagerly awaiting proper stocking of the IAXy and an FCC-certified FXO 
> module for the TDM400P -- I think those should be Digium's flagship products, 
> not a rebranded craptastic WinModem.
Hope so.

surely works better than the intel one, and I don't see any reason
in loosing (your, of course) time into making it work under zaptel.
Isn't it a craptastic WinModem also? even if made by Intel?


-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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RE: [Asterisk-Users] x100p config

[Dave Cotton]

On Sun, 2004-04-18 at 22:31, William J Mandra wrote:
> Paul,
>I just added the modprobe commands to my /etc/rc.local file to load
> the wc cards. 
>  
>Bill

If you have a Redhat/Mandrakelinux type system "make config" in
/usr/src/zaptel will put the necessary files in place then

chkconfig zaptel on 

and 

service zaptel start/stop

will do all the necessary loading and unloading in the correct order and
everything will be loaded at boot time.

"make config" in /usr/src/asterisk will do the same there also.
-- 
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] x100p config

[William J Mandra]

Paul,
   I just added the modprobe commands to my /etc/rc.local file 
to load the wc cards. 
 
   Bill

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Paul 
  TyremanSent: Sunday, April 18, 2004 ONYX 4:30 PMTo: 
  [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] x100p 
  config
  I don't have any wait commands in my s 
  extention.
   
  I don't use (or need to use since they don't work 
  in the UK) caller display on external calls, but I do want to keep it on 
  intenal calls, so is there any way to turn it off on exernal calls only 
  ?
   
  One more point, I rebooted my server and when I 
  tried to resart Asterisk again, I got an error saying something about no card 
  on d0001 (or something similar) and it refused to start.  I had to run 
  "modprobe wcfxo" before I 
  could start the server.  Is that normal, or is there something I can do 
  so it automaticly decects the card when I turn the server 
  on.
   
  Thanks again for yor help 
  Sean.
   
  Paul.
  
   
   
   
   
  -Original Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Sean 
  CheesmanPosted At: 18 April 2004 20:41Posted To: 
  Asterisk-UsersConversation: [Asterisk-Users] x100p configSubject: RE: 
  [Asterisk-Users] x100p config
   
  It could be one of several things.  The two things that come to 
  mind is Caller ID and a Wait() statement in your dialplan.  Since the 
  Caller ID information is transmitted between the first and second ring, 
  Asterisk has to wait for it if Caller ID is enabled.  Other than that, is 
  there aWait() line in your S extension?
   
  Sean-Original Message-From: Paul Tyreman 
  [mailto:[EMAIL PROTECTED] Sent: Sunday, April 18, 2004 2:31 PMTo: 
  [EMAIL PROTECTED]Subject: 
  RE: [Asterisk-Users] x100p config
   
  Thanks for your help.  I've got it working now.
   
  Only one problem.  When users from the public network call my 
  server, they hear three rings before the phones on my server start 
  ringing.  Is that usual, or is it a setting that can be changed ?
   
  Thanks, Paul.
   
   
   
  -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] 
  On Behalf Of Sean Cheesman Posted At: 18 April 2004 19:48 Posted To: 
  Asterisk-UsersSubject: RE: [Asterisk-Users] x100p config
   
  Welcome to the wonderful world of Asterisk!  In the future, you 
  might want to make sure that you post in plain text mode instead of HTML. 
  There are quite a few people here who are great assets that won't even read if 
  you post in HTML.
   
  Your problem has to do with the contexts.  In your zapata.conf file, 
  you will see reference to a context for your X100P.  That is the context 
  into which calls on that card will be dumped.  If you check your 
  extensions.conf, you should find a matching context that will have all of the 
  demo stuff in it.  You can either change the demo context to meet your 
  needs, or change your zapata.conf to point to a more useful context that has 
  just what you want in it.
   
  You might want to read over the info at http://www.voip-info.org. There's a lot of 
  good reading there that will help you make the most of Asterisk.
   
  Sean
   
  -Original Message-From: Paul Tyreman 
  [mailto:[EMAIL PROTECTED] Sent: Sunday, April 18, 2004 1:31 PMTo: 
  [EMAIL PROTECTED]Subject: 
  [Asterisk-Users] x100p config
   
  Hi,
   
  I have just installed my first X100P card, and seams to be half 
  working.
   
  You can call the public telephone number which the card is attached to 
  and hear some lady telling you about asterisk.  If I dial the extention 
  number of the phone I want to call, it connects and it's all good.
   
  However, I have put this line in my 
  extensions.conf:[incoming]exten => 
  s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)
   
  So it should ring phone one and phone two rather then give that that 
  girls voice !  Can anyone tell me what I'm doing wrong ?
   
  Also, I have put this in the same extensions.conf file: [outgoing] 
  exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
   
  [sip]include => outgoing
   
  Yet I still cannot make outgoing calls, when I dial 0 and the number I 
  want to call on the public network.
   
  Any help would be great as I'm starting to pull my hair out !
   
  Thanks, 
  Paul.___Asterisk-Users 
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Re: [Asterisk-Users] Incoming on Zap

[Paul Tyreman]

I have just installed a X100P card in my 
machine.
 
I found this webpage quite helpful:
http://users.pandora.be/Asterisk-PBX/InstallWildcard.htm
 
Hope its of some help to you,
 
Paul.
 
PS. Make sure your context is set up right, as that 
was my downfall.
PPS.See the posts I have made tonight on the same 
subject for more info.
 
 
 
-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Posted At: 18 April 2004 
21:28Posted To: Asterisk-UsersConversation: [Asterisk-Users] Incoming on 
ZapSubject: [Asterisk-Users] Incoming on Zap
 
Hello Everybody,
 
I have 1 X100P (FXO) card attached to my *, also there is a line connected 
to it, but when i dial that number it is not forwarding to anywhere, just * 
recognizes that the call is coming.
 
I want incoming call to be forward on my x-lite extension lets say 
2000,
 
can anybody tell me the settings of extensions.conf and other conf 
files.
 
thanks and best 
regards.-Neo___Asterisk-Users 
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RE: [Asterisk-Users] x100p config

[Paul Tyreman]

I don't have any wait commands in my s 
extention.
 
I don't use (or need to use since they don't work 
in the UK) caller display on external calls, but I do want to keep it on intenal 
calls, so is there any way to turn it off on exernal calls only ?
 
One more point, I rebooted my server and when I 
tried to resart Asterisk again, I got an error saying something about no card on 
d0001 (or something similar) and it refused to start.  I had to run 
"modprobe wcfxo" before I 
could start the server.  Is that normal, or is there something I can do so 
it automaticly decects the card when I turn the server on.
 
Thanks again for yor help 
Sean.
 
Paul.

 
 
 
 
-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Sean 
CheesmanPosted At: 18 April 2004 20:41Posted To: 
Asterisk-UsersConversation: [Asterisk-Users] x100p configSubject: RE: 
[Asterisk-Users] x100p config
 
It could be one of several things.  The two things that come to 
mind is Caller ID and a Wait() statement in your dialplan.  Since the 
Caller ID information is transmitted between the first and second ring, Asterisk 
has to wait for it if Caller ID is enabled.  Other than that, is there 
aWait() line in your S extension?
 
Sean-Original Message-From: Paul Tyreman 
[mailto:[EMAIL PROTECTED] Sent: Sunday, April 18, 2004 2:31 PMTo: [EMAIL PROTECTED]Subject: 
RE: [Asterisk-Users] x100p config
 
Thanks for your help.  I've got it working now.
 
Only one problem.  When users from the public network call my server, 
they hear three rings before the phones on my server start ringing.  Is 
that usual, or is it a setting that can be changed ?
 
Thanks, Paul.
 
 
 
-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] 
On Behalf Of Sean Cheesman Posted At: 18 April 2004 19:48 Posted To: 
Asterisk-UsersSubject: RE: [Asterisk-Users] x100p config
 
Welcome to the wonderful world of Asterisk!  In the future, you 
might want to make sure that you post in plain text mode instead of HTML. There 
are quite a few people here who are great assets that won't even read if you 
post in HTML.
 
Your problem has to do with the contexts.  In your zapata.conf file, 
you will see reference to a context for your X100P.  That is the context 
into which calls on that card will be dumped.  If you check your 
extensions.conf, you should find a matching context that will have all of the 
demo stuff in it.  You can either change the demo context to meet your 
needs, or change your zapata.conf to point to a more useful context that has 
just what you want in it.
 
You might want to read over the info at http://www.voip-info.org. There's a lot of 
good reading there that will help you make the most of Asterisk.
 
Sean
 
-Original Message-From: Paul Tyreman 
[mailto:[EMAIL PROTECTED] Sent: Sunday, April 18, 2004 1:31 PMTo: [EMAIL PROTECTED]Subject: 
[Asterisk-Users] x100p config
 
Hi,
 
I have just installed my first X100P card, and seams to be half 
working.
 
You can call the public telephone number which the card is attached to and 
hear some lady telling you about asterisk.  If I dial the extention number 
of the phone I want to call, it connects and it's all good.
 
However, I have put this line in my extensions.conf:[incoming]exten 
=> s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)
 
So it should ring phone one and phone two rather then give that that girls 
voice !  Can anyone tell me what I'm doing wrong ?
 
Also, I have put this in the same extensions.conf file: [outgoing] 
exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
 
[sip]include => outgoing
 
Yet I still cannot make outgoing calls, when I dial 0 and the number I want 
to call on the public network.
 
Any help would be great as I'm starting to pull my hair out !
 
Thanks, 
Paul.___Asterisk-Users 
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[Asterisk-Users] Incoming on Zap

[neo]

Hello Everybody,

I have 1 X100P (FXO) card attached to my *, also there is a line connected to 
it, but when i dial that number it is not forwarding to anywhere, just * 
recognizes that the call is coming.

I want incoming call to be forward on my x-lite extension lets say 2000,

can anybody tell me the settings of extensions.conf and other conf files.

thanks and best regards.
-Neo
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[Asterisk-Users] Dynamic recording function?

[Mark Phillips]

Hi Folks,

Yesterday I had need to record a phone conversation. This is not something
I'd ordinarily have to do and so I have not configured my * server to do
any recordings.

When looking for example dialplan stuff I found many examples where the
calls are always recorded when the phone is picked up but none that could
be done dynamicly.

What I'd like to be able to do is press a button and have * start
recording the call from that moment and then either stop when I hang up or
stop when I press another button.

Ideas?


G7LTT/KC2ENI
Mark Phillips
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Re: [Asterisk-Users] Database for extensions+vm+sip

[Fran Boon]

On Sun, 2004-04-18 at 18:51, Carlo Pires wrote:
> Is database available only for sip friends ? Is possible to put
> voicemail.conf and extensions.conf into db ?

Yes, all possible:
http://voip-info.org/wiki-Asterisk+configuration+from+database

F

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[Asterisk-Users] OpenPhone <-> Asterisk w/H.323

[Scott Stingel]

Hello-

In order to satisfy a customer requirement, I've just build H.323 under
asterisk (using the specified versions of OpenH323 & PWLib, and trying to
follow the instructions religiously), and it seems to have come up fine.
When testing with with OpenPhone (Windows version 1.8.1) and NetMeeting,
I've gotten some intermittent results however.  All my calls are from a PC
to asterisk - I don't have an outbound requirement.

If anyone has successfully made either of these combo's work, could you
please suggest some area where I may have gone wrong?

With OpenPhone:
When using MakeCall on OpenPhone, asterisk answers the call fine, even
though the OpenPhone display still shows "ringing" throughout the duration
of the call.  The audio (G7.11 uLaw) comes through the PC speaker fine.
When I issue Hangup on OpenPhone, asterisk most of the time does not get a
hang up signal, and comtinues to play until the timeout.  When dialing a
DTMF (out-of-band rfc2833) from OpenPhone, this works exactly once on
asterisk, and no further digits are seen by asterisk.

With NetMeeting:
On and off hook signalling is immediate, and seems to be fine.  Only about
one out of four calls produces audio through the PC speaker however.

So in summary, OpenPhone has reliable audio but signalling problems, and
NetMeeting has reliable signalling but flakey audio.

Any suggestions (please)?

Thanks!
 
 
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com  


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RE: [Asterisk-Users] x100p config

[Sean Cheesman]

It could be one of several things.  The two things that come to mind is
Caller ID and a Wait() statement in your dialplan.  Since the Caller ID
information is transmitted between the first and second ring, Asterisk
has to wait for it if Caller ID is enabled.  Other than that, is there a
Wait() line in your S extension?

Sean
-Original Message-
From: Paul Tyreman [mailto:[EMAIL PROTECTED] 
Sent: Sunday, April 18, 2004 2:31 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] x100p config


Thanks for your help.  I've got it working now.

Only one problem.  When users from the public network call my server,
they hear three rings before the phones on my server start ringing.  Is
that usual, or is it a setting that can be changed ?

Thanks, Paul.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Cheesman
Posted At: 18 April 2004 19:48
Posted To: Asterisk-Users
Subject: RE: [Asterisk-Users] x100p config


Welcome to the wonderful world of Asterisk!  In the future, you might
want to make sure that you post in plain text mode instead of HTML.
There are quite a few people here who are great assets that won't even
read if you post in HTML.

Your problem has to do with the contexts.  In your zapata.conf file, you
will see reference to a context for your X100P.  That is the context
into which calls on that card will be dumped.  If you check your
extensions.conf, you should find a matching context that will have all
of the demo stuff in it.  You can either change the demo context to meet
your needs, or change your zapata.conf to point to a more useful context
that has just what you want in it.

You might want to read over the info at http://www.voip-info.org.
There's a lot of good reading there that will help you make the most of
Asterisk.

Sean

-Original Message-
From: Paul Tyreman [mailto:[EMAIL PROTECTED] 
Sent: Sunday, April 18, 2004 1:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] x100p config


Hi,

I have just installed my first X100P card, and seams to be half working.

You can call the public telephone number which the card is attached to
and hear some lady telling you about asterisk.  If I dial the extention
number of the phone I want to call, it connects and it's all good.

However, I have put this line in my extensions.conf:
[incoming]
exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)

So it should ring phone one and phone two rather then give that that
girls voice !  Can anyone tell me what I'm doing wrong ?


Also, I have put this in the same extensions.conf file: [outgoing] exten
=> _0X.,1,Dial,Zap/1/${EXTEN:1}

[sip]
include => outgoing

Yet I still cannot make outgoing calls, when I dial 0 and the number I
want to call on the public network.

Any help would be great as I'm starting to pull my hair out !

Thanks, Paul.
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RE: [Asterisk-Users] x100p config

[Paul Tyreman]

Thanks for your help.  I've got it working 
now.
 
Only one problem.  When users from the public 
network call my server, they hear three rings before the phones on my server 
start ringing.  Is that usual, or is it a setting that can be changed 
?
 
Thanks, Paul.
 
 
 
-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Sean 
CheesmanPosted At: 18 April 2004 19:48Posted To: 
Asterisk-UsersSubject: RE: [Asterisk-Users] x100p config
 
Welcome to the wonderful world of Asterisk!  In the future, you 
might want to make sure that you post in plain text mode instead of HTML. There 
are quite a few people here who are great assets that won't even read if you 
post in HTML.
 
Your problem has to do with the contexts.  In your zapata.conf file, 
you will see reference to a context for your X100P.  That is the context 
into which calls on that card will be dumped.  If you check your 
extensions.conf, you should find a matching context that will have all of the 
demo stuff in it.  You can either change the demo context to meet your 
needs, or change your zapata.conf to point to a more useful context that has 
just what you want in it.
 
You might want to read over the info at http://www.voip-info.org. There's a lot of 
good reading there that will help you make the most of Asterisk.
 
Sean
 
-Original Message-From: Paul Tyreman 
[mailto:[EMAIL PROTECTED] Sent: Sunday, April 18, 2004 1:31 PMTo: [EMAIL PROTECTED]Subject: 
[Asterisk-Users] x100p config
 
Hi,
 
I have just installed my first X100P card, and seams to be half 
working.
 
You can call the public telephone number which the card is attached to and 
hear some lady telling you about asterisk.  If I dial the extention number 
of the phone I want to call, it connects and it's all good.
 
However, I have put this line in my extensions.conf:[incoming]exten 
=> s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)
 
So it should ring phone one and phone two rather then give that that girls 
voice !  Can anyone tell me what I'm doing wrong ?
 
Also, I have put this in the same extensions.conf file: [outgoing] 
exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
 
[sip]include => outgoing
 
Yet I still cannot make outgoing calls, when I dial 0 and the number I want 
to call on the public network.
 
Any help would be great as I'm starting to pull my hair out !
 
Thanks, 
Paul.___Asterisk-Users 
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[Asterisk-Users] asterisk demo (was: x100p config)

[Mark Elkins]

I think what is missing with asterisk is what I'd call a 'working' demo.
Its real cool getting the 'Welcome to asterisk' demo running...

I think that there should be a 'make basic-plan' that would generate
some well commented '.conf' files that set up a basic working systems
with..

Two phones for Support
Two phones for Sales
Two phones for Accounting
Voice mail for all.
Outgoing group.
Incoming group that does the 'press 1 for sales, 2 for support...'
etc... 

Once I figure it all out - I might try to do it myself. There are some
nice 'Starting Guide' walk-throughs - but no one has put a complete
idiots guide together yet... (for idiots like me).(Yeh - I know - WIP)

-- 
  .  . ___. .__  Posix Systems - Sth Africa
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496



signature.asc
Description: This is a digitally signed message part

RE: [Asterisk-Users] x100p config

[Sean Cheesman]

Welcome to the wonderful world of Asterisk!  In the future, you might
want to make sure that you post in plain text mode instead of HTML.
There are quite a few people here who are great assets that won't even
read if you post in HTML.

Your problem has to do with the contexts.  In your zapata.conf file, you
will see reference to a context for your X100P.  That is the context
into which calls on that card will be dumped.  If you check your
extensions.conf, you should find a matching context that will have all
of the demo stuff in it.  You can either change the demo context to meet
your needs, or change your zapata.conf to point to a more useful context
that has just what you want in it.

You might want to read over the info at http://www.voip-info.org.
There's a lot of good reading there that will help you make the most of
Asterisk.

Sean

-Original Message-
From: Paul Tyreman [mailto:[EMAIL PROTECTED] 
Sent: Sunday, April 18, 2004 1:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] x100p config


Hi,

I have just installed my first X100P card, and seams to be half working.

You can call the public telephone number which the card is attached to
and hear some lady telling you about asterisk.  If I dial the extention
number of the phone I want to call, it connects and it's all good.

However, I have put this line in my extensions.conf:
[incoming]
exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)

So it should ring phone one and phone two rather then give that that
girls voice !  Can anyone tell me what I'm doing wrong ?


Also, I have put this in the same extensions.conf file:
[outgoing]
exten => _0X.,1,Dial,Zap/1/${EXTEN:1}

[sip]
include => outgoing

Yet I still cannot make outgoing calls, when I dial 0 and the number I
want to call on the public network.

Any help would be great as I'm starting to pull my hair out !

Thanks, Paul.
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[Asterisk-Users] x100p config

[Paul Tyreman]

Hi,
 
I have just installed my first X100P card, and 
seams to be half working.
 
You can call the public telephone number which the 
card is attached to and hear some lady telling you about asterisk.  If I 
dial the extention number of the phone I want to call, it connects and it's all 
good.
 
However, I have put this line in my 
extensions.conf:
[incoming]exten => 
s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)
 
So it should ring phone 
one and phone two rather then give that that girls voice !  Can anyone tell 
me what I'm doing wrong ?
 
 
Also, I have put this 
in the same extensions.conf file:
[outgoing]exten => 
_0X.,1,Dial,Zap/1/${EXTEN:1}
 
[sip]
include => 
outgoing
 
Yet I still cannot make outgoing calls, 
when I dial 0 and the number I want to call on the public 
network.
Any help would be great as I'm starting to pull my hair out !
 
Thanks, Paul.

Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Vlok Stone]

here's addition info on sip debug


11 headers, 9 lines
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format telephone-event
Capabilities: us - 14, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: 
list_route: hop: 
set_destination: Parsing 
for address/port to send to
set_destination: set destination to 192.246.69.223, port 5060


sip show channelsPeer User/ANRCall ID  Seq (Tx/Rx) 
Lag  Jitter  Format
192.246.69.223   613 1ecd512b4bf  00103/0  0ms  ms
ULAW

192.168.1.247200094915249b0e  00102/01317  0ms  ms 
ULAW

are these normal?



On Sat, 2004-04-17 at 17:12, Olle E. Johansson wrote:
> Chris Orme wrote:
> 
> >>>exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> 
> Isn't the 'r' forcing a 'ringing' signal from start, regardless
> of what the device you are calling are signalling. If you are calling
> a SIP device, that device might return 'busy' and that's propably
> why you first hear 'ringing' and then a 'busy' signal.
> 
> I would like app_dial gurus to explain the 'r' option a bit
> more so we can document it better.
> 
> /O
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[Asterisk-Users] Database for extensions+vm+sip

[Carlo Pires]

Hello folks,

Is database available only for sip friends ? Is possible to put
voicemail.conf and extensions.conf into db ?
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[Asterisk-Users] FWD registration problems

[Larry Keyes]

Hi..I'm having trouble registering my asterisk box with FWDIt worked the
other day.  

I also have an individual Grandstream phone which registers fine right now. 

I looked at the archives and saw the thing about the maximum retries limit
to 5...but since my Grandstream phone seems to register on the first try,
I'm thinking the problem lies elsewhere.  Any ideas? 


sip show peers
Name/usernameHost Mask Port Status
fwd/291990   192.246.69.223   255.255.255.255  5060 Unmonitored
1001/1001192.168.0.162255.255.255.255  5060 OK (3 ms)
1000/1000192.168.0.160   (D)  255.255.255.255  5060 OK (3 ms)


sip show registry
Host  Username Refresh State
192.246.69.223:5060   291990   160 Request Sent

NOTICE[1125350192]: File chan_sip.c, Line 2949 (sip_reg_timeout):
Registration for '[EMAIL PROTECTED]' timed out, trying again

WARNING[1125350192]: File chan_sip.c, Line 462 (retrans_pkt): Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 103 (Request)

Best wishes, 
 
-- Larry 
 
 
-
Lawrence Keyes
Microdesign Consulting Inc.
Software Development * Network Design
Technical Leadership & Management 
[EMAIL PROTECTED] - (802) 658-4673
www.mxdesign.net
 

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RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?

[Victor Perez]

I don't know if this helps, but I started having this problem after I sent out a fax. 
My fax machine was connected to line 1 at that time. I tried changing the FAX 
detection settings but no luck.


Regards,
Victor Perez



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Musone
Sent: Sunday, April 18, 2004 10:33 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?


I'm not sure if anybody has determined the cause/fix for this problem,
but I am getting the same problem.

I turned on syslog debugging and there were some interesting results:

...

My feeling is that this is a Sipura problem. I've upgraded to the
firmare 2.02, but still no difference.


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RE: [Asterisk-Users] E100P for Bandwidth Termination

[Todd Lieberman]

If memory serves correct, you'll you have to pass the data channels to an
open T1 card and use a cross over cable to a router.  Don't quote me on this
as I have not done this myself.

TL

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Azher Amin
Sent: Saturday, April 17, 2004 1:59 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] E100P for Bandwidth Termination


Hi,

I have a query from a client that can he use the E100P card to terminate
the 2Mbps bandwidth in a linux box, thus reducing the cost of cisco
router ??

The other end is a cisco 2620 router with E1 VWIC-1MFC.

Can anyone explain if its possible with Asterisk and further any
configuration help. Applreciated.

Regards
Azher Amin
---
http://www.consulttech.com.pk




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Re: [Asterisk-Users] PRI: This number has been disconnected

[willy]

THANK YOU
- Original Message Follows -
> Hi,
> 
> it seems like you are using the 'r' option of app_dial.
> This will fake ring indication and will not pass any audio
> until the call is answered.

THANK YOU!
That was it!
As usual as was looking for a complex solution instead of
seeing the obvious.
I must admit that with the recent issues with the 'r' option
I was confused about its meaning.  I remember that some
(short) time ago, without an 'r' no ringing would be heard
at all.  This must have been a bug.  However, that is when I
added 'r' to all my Dial commands.
Thanks again,
Willy


> What does your dial extension look like?
> 
> best regards
> 
> kapejod
> -- 
> Klaus-Peter Junghanns
> 
> CEO, CTO
> Junghanns.NET GmbH
> Breite Strasse 13a - 12167 Berlin - Germany
> fon: (de) +49 30 79705390
> fon: (uk) +44 870 1244692
> fax: (de) +49 30 79705391
> iaxtel: 1-700-157-8753
> http://www.Junghanns.NET/asterisk/
> 
> 
> Am So, 2004-04-18 um 17.09 schrieb [EMAIL PROTECTED]:
> > All,
> > When calling an invalid number using, I expect to hear:
> > "dooh-deeh-daah We're sorry you have reached a number
> > which has been disconnected ..."
> > And that is indeed what I hear when I dial out from [*]
> > using analog FXO, or VoicePulse or NuPhone.  When I dial
> > that same number trough the T1 / PRI interface however,
> > I continually hear ringing, and then the call gets
> > hungup. Any ideas anyone?
> > It kinda annoys our users, since they like to *know*
> > when they dial an invalid number.
> > TIA,
> > WW
> > 
> > Willy Wouters
> > ypOne Publishing
> > 
> > ___
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> > [EMAIL PROTECTED]
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> > To UNSUBSCRIBE or update options visit:
> >   
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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Willy Wouters
ypOne Publishing

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Re: [Asterisk-Users] PRI: This number has been disconnected

[willy]

- Original Message Follows -
> All,
> When calling an invalid number using, I expect to hear:
> "dooh-deeh-daah We're sorry you have reached a number
> which has been disconnected ..."
> And that is indeed what I hear when I dial out from [*]
> using analog FXO, or VoicePulse or NuPhone.  When I dial
> that same number trough the T1 / PRI interface however, I
> continually hear ringing, and then the call gets hungup.


Further info ...
So I ran the scenario with PRI DEBUG SPAN 1 and here is what
I can see:
** Call to a Good Number **
< Message type: CALL PROCEEDING (2)
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare:
0, Exclusive Dchan: 0
 Any ideas anyone?
> It kinda annoys our users, since they like to *know* when
> they dial an invalid number.
> TIA,
> WW
> 
> Willy Wouters
> ypOne Publishing
> 
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Willy Wouters
ypOne Publishing

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RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?

[Mark Musone]

I'm not sure if anybody has determined the cause/fix for this problem,
but I am getting the same problem.

I turned on syslog debugging and there were some interesting results:

1. phone call answered on line2:

Apr 18 11:14:13 192.168.1.21 [0:0]AUD ALLOC CALL (port=16428)
Apr 18 11:14:14 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:14:14 192.168.1.21 [1:0]AUD ALLOC CALL (port=16430)
Apr 18 11:14:14 192.168.1.21 [1:0]RTP Rx Up
Apr 18 11:14:17 192.168.1.21 [1]Off Hook
Apr 18 11:14:17 192.168.1.21 [1]CID interrupted
Apr 18 11:14:17 192.168.1.21 [1:0]RTP Rx 1st PKT @16430(2)
Apr 18 11:14:17 192.168.1.21 [0:0]AUD Rel Call
Apr 18 11:14:17 192.168.1.21 CC:Ended
Apr 18 11:14:17 192.168.1.21 CC:Connected
Apr 18 11:14:18 192.168.1.21 [1:0]ENC INIT 0
Apr 18 11:14:18 192.168.1.21 [1:0]RTP Tx Up (pt=0->c0a80114:15840)
Apr 18 11:14:18 192.168.1.21 [1:0]RTCP Tx Up
Apr 18 11:14:18 192.168.1.21 [1:0]DEC INIT 0
Apr 18 11:14:22 192.168.1.21 [1]On Hook
Apr 18 11:14:22 192.168.1.21 [1:0]AUD Rel Call
Apr 18 11:14:22 192.168.1.21 DLG Terminated
Apr 18 11:14:22 192.168.1.21 Sess Terminated
Apr 18 11:14:49 192.168.1.21 DLG Terminated
Apr 18 11:14:49 192.168.1.21 Sess Terminated
Apr 18 11:14:54 192.168.1.21 CC:Clean Up


2. Phone call answered on line1 (no outgoing voice)

Apr 18 11:01:16 192.168.1.21 [0:0]AUD ALLOC CALL (port=16424)
Apr 18 11:01:16 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:01:16 192.168.1.21 [1:0]AUD ALLOC CALL (port=16426)
Apr 18 11:01:16 192.168.1.21 [1:0]RTP Rx Up
Apr 18 11:01:18 192.168.1.21 [0]Off Hook
Apr 18 11:01:18 192.168.1.21 [0]CID interrupted
Apr 18 11:01:18 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:01:18 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:01:18 192.168.1.21 CC:Connected
Apr 18 11:01:18 192.168.1.21 No Common TxCodec
Apr 18 11:01:18 192.168.1.21 [0:0]RTP Rx 1st PKT @16424(2)
Apr 18 11:01:19 192.168.1.21 [1:0]AUD Rel Call
Apr 18 11:01:19 192.168.1.21 CC:Ended
Apr 18 11:01:19 192.168.1.21 [0:0]DEC INIT 0
Apr 18 11:01:21 192.168.1.21 [0]On Hook
Apr 18 11:01:21 192.168.1.21 [0:0]AUD Rel Call
Apr 18 11:01:21 192.168.1.21 DLG Terminated
Apr 18 11:01:21 192.168.1.21 Sess Terminated
Apr 18 11:01:50 192.168.1.21 DLG Terminated
Apr 18 11:01:50 192.168.1.21 Sess Terminated
Apr 18 11:01:53 192.168.1.21 CC:Clean Up

Seems like for some reason, line1 is not agreeing on the Tx codec..i
tried playing around with the codecs configured in both the sipura
configs and asterisk, but could not find anything that worked..


Below is the interesting flash-hook fix:

Apr 18 11:17:29 192.168.1.21 [0:0]AUD ALLOC CALL (port=16432)
Apr 18 11:17:29 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:17:29 192.168.1.21 [1:0]AUD ALLOC CALL (port=16434)
Apr 18 11:17:29 192.168.1.21 [1:0]RTP Rx Up
Apr 18 11:17:32 192.168.1.21 [0]Off Hook
Apr 18 11:17:32 192.168.1.21 [0]CID interrupted
Apr 18 11:17:32 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:17:32 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:17:32 192.168.1.21 [0:0]RTP Rx 1st PKT @16432(2)
Apr 18 11:17:32 192.168.1.21 CC:Connected
Apr 18 11:17:32 192.168.1.21 No Common TxCodec
Apr 18 11:17:32 192.168.1.21 [1:0]AUD Rel Call
Apr 18 11:17:32 192.168.1.21 CC:Ended
Apr 18 11:17:32 192.168.1.21 [0:0]DEC INIT 0
Apr 18 11:17:35 192.168.1.21 [0]Hook Flash
Apr 18 11:17:35 192.168.1.21 [0:0]RTP Tx Dn
Apr 18 11:17:35 192.168.1.21 [0:0]RTP Rx Dn
Apr 18 11:17:36 192.168.1.21 CC:Hold
Apr 18 11:17:36 192.168.1.21 [0:0]RTP Tx Dn
Apr 18 11:17:36 192.168.1.21 CC:Remote Resume
Apr 18 11:17:36 192.168.1.21 [0:0]RTCP Tx Up
Apr 18 11:17:38 192.168.1.21 CC:Connected
Apr 18 11:17:38 192.168.1.21 [0:0]RTP Tx Dn
Apr 18 11:17:39 192.168.1.21 [0:0]ENC INIT 0
Apr 18 11:17:39 192.168.1.21 [0:0]RTP Tx Up (pt=0->c0a80114:10040)
Apr 18 11:17:39 192.168.1.21 [0:0]RTCP Tx Up
Apr 18 11:17:39 192.168.1.21 CC:Remote Resume
Apr 18 11:17:39 192.168.1.21 [0:0]RTCP Tx Up
Apr 18 11:17:39 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:17:39 192.168.1.21 [0:0]RTP Rx 1st PKT @16432(2)
Apr 18 11:17:39 192.168.1.21 [0:0]DEC INIT 0
Apr 18 11:17:38 192.168.1.21 [0]Hook Flash
Apr 18 11:17:46 192.168.1.21 [0]On Hook
Apr 18 11:17:47 192.168.1.21 [0:0]AUD Rel Call
Apr 18 11:17:56 192.168.1.21 [1]RegOK. NextReg in 180
Apr 18 11:17:56 192.168.1.21 [0]RegOK. NextReg in 180
Apr 18 11:18:04 192.168.1.21 DLG Terminated
Apr 18 11:18:04 192.168.1.21 Sess Terminated
Apr 18 11:18:10 192.168.1.21 DLG Terminated
Apr 18 11:18:10 192.168.1.21 Sess Terminated
Apr 18 11:18:18 192.168.1.21 CC:Clean Up


I tried doing some sniffs with ethereal and sip debug, but neither
seemed to sow any info like this (of course, I don't know the sip debug
stuff that well)

w
My feeling is that this is a Sipura problem. I've upgraded to the
firmare 2.02, but still no difference.

Besides that, my configs for line1 and line2 in the sipura configuration
are exactly the same, except for SIP port (5060 and 5061 respectively)
and the extention numbers (2201 and 2202 respectively)

My asterisk config for the sip lines are:

[2202]
type=friend
host=dynamic
context=home

Re: [Asterisk-Users] PRI: This number has been disconnected

[Klaus-Peter Junghanns]

Hi,

it seems like you are using the 'r' option of app_dial.
This will fake ring indication and will not pass any audio
until the call is answered.
What does your dial extension look like?

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


Am So, 2004-04-18 um 17.09 schrieb [EMAIL PROTECTED]:
> All,
> When calling an invalid number using, I expect to hear:
> "dooh-deeh-daah We're sorry you have reached a number which
> has been disconnected ..."
> And that is indeed what I hear when I dial out from [*]
> using analog FXO, or VoicePulse or NuPhone.  When I dial
> that same number trough the T1 / PRI interface however, I
> continually hear ringing, and then the call gets hungup.
> Any ideas anyone?
> It kinda annoys our users, since they like to *know* when
> they dial an invalid number.
> TIA,
> WW
> 
> Willy Wouters
> ypOne Publishing
> 
> ___
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Re: [Asterisk-Users] Intel 536ep as a FXO?

[Andrew Kohlsmith]

> P.S. and you'll have free installation support from
> digium and a rock solid hw made for asterisk.

I have and use the X100/101P, TDM400P, T100P, TE410P and TE405P from Digium.

Only a Digium zealot would call the X100/X101P 'rock solid hw' -- Digium is 
reselling a generic WinModem for these cards and they are simply not good 
hardware.  I love the other cards to death -- they are solid and stable and 
Just Work, but please don't blindly endorse anything that comes out with 
a Digium stamp on it -- you're only decreasing the value of Digium's name.

They needed a cheap FXO interface for the masses and for now, that's what we 
have.  It's certainly not a good solution, but it is *a* solution.

I am eagerly awaiting proper stocking of the IAXy and an FCC-certified FXO 
module for the TDM400P -- I think those should be Digium's flagship products, 
not a rebranded craptastic WinModem.

Regards,
Andrew
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[Asterisk-Users] PRI: This number has been disconnected

[willy]

All,
When calling an invalid number using, I expect to hear:
"dooh-deeh-daah We're sorry you have reached a number which
has been disconnected ..."
And that is indeed what I hear when I dial out from [*]
using analog FXO, or VoicePulse or NuPhone.  When I dial
that same number trough the T1 / PRI interface however, I
continually hear ringing, and then the call gets hungup.
Any ideas anyone?
It kinda annoys our users, since they like to *know* when
they dial an invalid number.
TIA,
WW

Willy Wouters
ypOne Publishing

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Re: [Asterisk-Users] Intel 536ep as a FXO?

[Brancaleoni Matteo]

buy a x100p for 100 bucks and support digium.

Matteo.

P.S. and you'll have free installation support from
digium and a rock solid hw made for asterisk.


Il dom, 2004-04-18 alle 16:43, Marcin Mazurek ha scritto:
> Hi,
> 
> I've seen some reports about ruuning Intel modem with 537 or MD3200
> chipset running with Zaptel drivers as a FXO port. Did anybody managed
> to set up a PCI faxmodem based on Intel536ep chipset to work with * and
> Zaptel drivers?
> Modem seemd to work just fine with Linux, but the driver says no;)
> 
> some more info:
> 
> Linux 2.4.26
> 
> mazuchna:~# cat /proc/pci | grep 536
> Communication controller: Intel Corp. 536EP Data Fax Modem (rev 0).
> 
> mazuchna:~# lsmod
> Module  Size  Used byTainted: P
> ztdynamic   6692   0  (unused)
> zaptel177280   0  [ztdynamic]
> Intel536  876524   0  (unused)
> 
> mazuchna:/lib/modules/2.4.26/misc# insmod wcfxo
> Using /lib/modules/2.4.26/misc/wcfxo.o
> /lib/modules/2.4.26/misc/wcfxo.o: init_module: No such device
> Hint: insmod errors can be caused by incorrect module parameters,
> including invalid IO or IRQ parameters.
>   You may find more information in syslog or the output from dmesg
> 
> is there anything more I can do?
> tia
> mazek
-- 
Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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Re: [Asterisk-Users] grandstream and stun

[Brancaleoni Matteo]

the only difference with NAT=yes into GS is that
enabling it the phone will send periodic (every 20secs @default)
empty UDP packets to the SIP server, keeping
the NAT hole open... so you don't have to
do a dnat rule onto the nat'ing device.

very useful :)

Matteo.

Il dom, 2004-04-18 alle 16:46, Ryan Thrash ha scritto:
> FYI, with 1.0.4.55 and NAT set to off (but with the * config set as 
> nat=yes), I'm able to bypass stun servers completely with a GS phone as 
> well.
> 
> HTH,
> Ryan
> 
> 
> On Apr 18, 2004, at 5:08 AM, Brancaleoni Matteo wrote:
> 
> > you don't need stun to make GS work under NAT
> > with *
> >
> > Just set NAT=yes into the GS, and leave the stun server addr
> > entry empty.
> > And set nat=yes into the sip.conf entry.
> >
> > Il dom, 2004-04-18 alle 11:26, Richard ha scritto:
> >> Hi,
> >>
> >> I noticed some issues with how grandstream handles
> >> stun test. GS is running version 1.0.4.50. First I
> >> reset the NAT router. Then reboot GS, get results of
> >> "restricted cone". Immediately reboot GS, get results
> >> "full cone". I tried quite a few public and commercial
> >> stun servers. Also tried different model/version of
> >> linksys routers. I always got the same issue. Winstun
> >> on the PC doesn't have this issue. Some ngrep on the
> >> stund 0.91 on Fedora linux revealed winstun had about
> >> 20 UDP packets back and forward. However GS only had
> >> less than 10.
> 
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Brancaleoni Matteo <[EMAIL PROTECTED]>
Espia - Emmegi Srl

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Re: [Asterisk-Users] grandstream and stun

[Ryan Thrash]

FYI, with 1.0.4.55 and NAT set to off (but with the * config set as 
nat=yes), I'm able to bypass stun servers completely with a GS phone as 
well.

HTH,
Ryan
On Apr 18, 2004, at 5:08 AM, Brancaleoni Matteo wrote:

you don't need stun to make GS work under NAT
with *
Just set NAT=yes into the GS, and leave the stun server addr
entry empty.
And set nat=yes into the sip.conf entry.
Il dom, 2004-04-18 alle 11:26, Richard ha scritto:
Hi,

I noticed some issues with how grandstream handles
stun test. GS is running version 1.0.4.50. First I
reset the NAT router. Then reboot GS, get results of
"restricted cone". Immediately reboot GS, get results
"full cone". I tried quite a few public and commercial
stun servers. Also tried different model/version of
linksys routers. I always got the same issue. Winstun
on the PC doesn't have this issue. Some ngrep on the
stund 0.91 on Fedora linux revealed winstun had about
20 UDP packets back and forward. However GS only had
less than 10.
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[Asterisk-Users] Intel 536ep as a FXO?

[Marcin Mazurek]

Hi,

I've seen some reports about ruuning Intel modem with 537 or MD3200
chipset running with Zaptel drivers as a FXO port. Did anybody managed
to set up a PCI faxmodem based on Intel536ep chipset to work with * and
Zaptel drivers?
Modem seemd to work just fine with Linux, but the driver says no;)

some more info:

Linux 2.4.26

mazuchna:~# cat /proc/pci | grep 536
Communication controller: Intel Corp. 536EP Data Fax Modem (rev 0).

mazuchna:~# lsmod
Module  Size  Used byTainted: P
ztdynamic   6692   0  (unused)
zaptel177280   0  [ztdynamic]
Intel536  876524   0  (unused)

mazuchna:/lib/modules/2.4.26/misc# insmod wcfxo
Using /lib/modules/2.4.26/misc/wcfxo.o
/lib/modules/2.4.26/misc/wcfxo.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters,
including invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg

is there anything more I can do?
tia
mazek

-- 
http://mazek.netsync.pl/::: nic-hdl: MM3380-RIPE
GnuPG 6687 E661 98B0 AEE6 DA8B  7F48 AEE4 776F 5688 DC89
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Re: [Asterisk-Users] VOIP Spam

[John Todd]

At 8:32 AM -0700 on 4/15/04, Tom Green wrote:
Hi,

Some people have suggested maintaining black lists and
white lists to avoid spammers and allow legitimate
callers into the network. However, the problem with
this method is that the spammer's IP address might
change due to DHCP. Today a spammer might get
aaa.bbb.ccc.ddd and lets say that I put this address
in my blacklist. To my annoyance, tomorrow a
legitimate caller might get aaa.bbb.ccc.ddd and the
spammer might get a different IP address. In the end,
I end up blocking the legitimate caller also. Any
ideas or thoughts to on this problem is appreciated.
Thanks,
Tom
I've read the rest of this thread about PKI, shared certs, etc. but I 
think that an important middle step is being missed by everyone.

I believe strongly in the concept of end-to-end connectivity as the 
"optimal" method to ensure authentication and authorization between 
two user agents (web, voip, email, whatever.)  However, it is often 
difficult to build such mechanisms that are easily used by the "end 
user."  Most end users will happily hand over the responsibility for 
protection against "spam" in any form to a central administrator, and 
I think that as a first step it is appropriate to move the smart 
stuff to a central server instead of to every user's desktop (though 
eventually there should be smart stuff on the desktop.)

To this end: why is it _mandatory_ that all VOIP endpoints accept 
calls from other endpoints?  Of course, you could filter based on 
some type of kludge-y network filters, but that is ugly and does not 
scale.  SIP (and possibly IAX; I haven't looked at it much) have the 
ability to demand credentials from the remote host.  Why don't we use 
these features?

Here is my ideal world: When a SIP INVITE (or NOTIFY, or whatever) 
hits my desk SIP phone, it should refuse the message with a "401 
Unauthorized" message.  Without correct credentials, messages simply 
aren't allowed past the threshhold of the SIP UA.  This should be a 
configurable option on my SIP UA - maybe I have some reasons to allow 
all messages from all hosts at some time.  However, most of the time 
I would want my SIP server (Asterisk, SER, whatever) to be in the 
path, and that "smart" gateway could do my blacklisting, 
authentication (PKI, etc.) and other tasks which would require more 
brains and more central administration.

No SIP device that I've ever seen has the option to deny SIP messages 
from all but authenticated hosts.  Why is that?  Seems pretty 
obvious.  It's always the other way around - SIP proxies allow or 
disallow messages according to authentication credentials (shared 
secret.)  Since I've never seen this in place, perhaps it is the case 
that I am mis-understanding how authentication can possibly work with 
SIP between a UA and a proxy?

JT
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Re: [Asterisk-Users] Asterisk in pass-thru mode

[John Todd]

At 8:13 PM +0800 on 4/15/04, Radius wrote:
Hi all,

Below is what I did to run Asterisk in pass-thru mode:

sip.conf:
[general]
disallow=all
allow=ulaw
canreinvite=yes
For each channel, canreinvite=yes is enabled. No dial command has 't' option.

However, it seems that Asterisk still stay in the media path and 
bridge the 2 end points. Am I missing something???

sip*CLI> show channels
Channel  (ContextExtensionPri )   State Appl. Data
SIP/5001-c60b  (company11   )  Up Bridged 
Call  SIP/1234-faf1
  SIP/1234-faf1  (company1   5001 1   )  Up Dial 
SIP/5001|20|r
2 active channel(s)

sip*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Lag  Jitter  Format
192.168.1.1015001257684717aa  00104/0  0ms  ms  ULAW
210.17.211.5 1234003094c2-fd  00104/00102  0ms  ms  ULAW
2 active SIP channel(s)
Thanks.
Ben
Ben -
  Yes.
http://lists.digium.com/pipermail/asterisk-users/2004-March/039663.html

JT
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Re: [Asterisk-Users] Re: Windows Drivers for Wildcard FXO Card

[Serge Mankovski]

It depends on what you are going to do with it.

If you want to use it as a modem, then you can find a driver for it. I tried 
to do that. It installs, but I could not find a wave driver for it to have 
it as a voice modem even.

This hardware is sort of useles on Windows... but on Linux it rocks!

Serge



From: Anon <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Windows Drivers for Wildcard FXO Card
Date: Sun, 18 Apr 2004 06:39:33 -0600
On Friday 16 April 2004 09:37 am, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
>
> Bill McCready <[EMAIL PROTECTED]> wrote:
> > Where may I find a Windows driver for a Wildcard FXO Card ???
>
> Why would anyone want such a thing?
Job security and profit through many service calls... like any other
Windoze product.
[If it works, don't fix it] = [no service call profit]

Anon

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Re: [Asterisk-Users] SoundPointR IP 300

[Anon]

On Friday 16 April 2004 05:07 am, Shad Mortazavi wrote:
> Dear Group,
>
> Does any one have experience using SoundPoint(r) IP 300?
>
> I have one call center on Snom 200's I'm adding a second and was looking at
> the SoundPoint, but needed some input.

Also
I have read the Asterisk-User's list for about 8 months, and have yet to see
any serious problem posted about a Polycom phone, despite knowing there 
are people on the list successfully using Polycom SoundPoint IP phones.  
To me, this indicates the Polycom SoundPoint phones are likely to be 
easy (or at least easier) to install/run, and be more reliable than other
phones.

By contrast, I have read many, many problems posted with other brands of
phones, with the Cisco in particular.

Anon

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Re: [Asterisk-Users] VOIP Spam

[Duane]

Tom Green wrote:

Some people have suggested maintaining black lists and
white lists to avoid spammers and allow legitimate
callers into the network. However, the problem with
this method is that the spammer's IP address might
change due to DHCP. Today a spammer might get
aaa.bbb.ccc.ddd and lets say that I put this address
in my blacklist. To my annoyance, tomorrow a
legitimate caller might get aaa.bbb.ccc.ddd and the
spammer might get a different IP address. In the end,
I end up blocking the legitimate caller also. Any
ideas or thoughts to on this problem is appreciated.
A couple of discussion about this have come up, and something occurred 
to me about the FCC decision about free world dialup not being 
classified as a phone service. This opens the flood gates to 
telemarketers to FWD users as they don't have to honour any form of do 
not call list the FCC issues, which then of course leads on to other 
systems like IAXTEL with all voice data over the internet rather then 
pstn network...

Also a possibly solution may have come out of the same discussions, 
technically if everyone enforces some kind of enum lookup before 
accepting calls, and the same enum lookup will return NAPTR records, so 
a slight modification to loop through all DNS records could then be 
checked against the current hostname/IP in a similar fashion to SPF 
records and mail servers...

End result is a nice neat little database of blacklisted phone numbers 
rather then IPs, you'd need some resolution service/time-out period to 
remove the black listing, but  lot harder to get new phone numbers 
then new IPs...

PS We've come up with a patch to the enum lookup to return a Caller Name 
from a TXT record, appreciate any feedback, our c skills are a little 
rusty so it's possibly not the most elegant solution...

http://bugs.digium.com/bug_view_page.php?bug_id=0001442

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 Duane
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http://e164.org - Using Enum.164 to interconnect asterisk servers
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Re: [Asterisk-Users] Re: Windows Drivers for Wildcard FXO Card

[Anon]

On Friday 16 April 2004 09:37 am, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
>
> Bill McCready <[EMAIL PROTECTED]> wrote:
> > Where may I find a Windows driver for a Wildcard FXO Card ???
>
> Why would anyone want such a thing?
Job security and profit through many service calls... like any other
Windoze product.

[If it works, don't fix it] = [no service call profit]

Anon

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[Asterisk-Users] h323 oh323 g729 please help !

[Serge]

Hello list,
 

I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. 
etc I need: G711 from 
old phones must be convert to G729 via asterisk and send to provider I have this 
problem: 

oh323 (last version): - 

asterisk work with this driver ok for old phones, if I only faststart=no . 
But problem with codec , asterisk can speak with provider ( G729 ) only if I 
disable all other codec ! ( bug ? ) , but I need minimum 2 - g711 and 
g729. 
h323 -- 

all work ok , but only for new phones ! like cisco ATA .., with this driver 
old phones don't may speak with asterisk ! So, and last bug.. when I enable 2 codec in both version, I 
need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set 
codec by destination? ( like SIP )
I try use 2 cnannels at the same time, but asterisk 
down with segmentation fault...
Thanks,Serge.

Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Mark Elkins]

On Sat, 2004-04-17 at 15:58, Chris Orme wrote:

> My dialplan is for the outgoing SIP call is:
> 
> exten => _00.,1,AbsoluteTimeout(3600)
> exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
> exten => _00.,3,Answer
> exten => _00.,4,Hangup
> exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
> exten => _00.,104,Answer
> exten => _00.,105,Hangup
> 
> (if call can go through on TRUNK1 send it out, if TRUNK1 is out of
> capacity and therefore busy then try trunk 2 before giving up) if that is
> busy (therefore it is likely the number really is busy then grab the
> caller and hang them up (and they then hear 'busy').  


Um - I'm probably missing the point entirely - but why are your trunks
not in a group and why are you not then using the group to dial out on?

(not posted to Asterisk - just you)
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Re: [Asterisk-Users] grandstream and stun

[Mark Elkins]

On Sun, 2004-04-18 at 11:26, Richard wrote:
> Hi,
> 
> I noticed some issues with how grandstream handles
> stun test. GS is running version 1.0.4.50.

The latest release of software for Grandstream (dunno if its the same
for all phone??? - but for Product Model: BT100)
is: 
Software Version: Program:1.0.4.55 Bootloader:1.0.0.14 HTML:1.0.0.24

One way to upgrade is set the phone's TFTP up to load from grandstream.
(Line reads...)
TFTP Server: 4.3.153.50 (for remote software upgrade and configuration)

Then reboot. I had to reboot twice in order to update everything.

I notice that the Grandstream phone tries to download from the TFTP
server two interestingly names files...
cfg000b82006e69  -and- cft.txt
The first is cfg + Mac address of my phone.
I'd guess this is could be my phone's config? Anyone know the format
of the file - and how to make a phone dump its config?
Anyone know the format of the cfg.txt file?
Is there a definitive document on this anywere? (I've Searched Google)

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Re: [Asterisk-Users] grandstream and stun

[Brancaleoni Matteo]

you don't need stun to make GS work under NAT
with *

Just set NAT=yes into the GS, and leave the stun server addr
entry empty.
And set nat=yes into the sip.conf entry.

That's all

Matteo.

Il dom, 2004-04-18 alle 11:26, Richard ha scritto:
> Hi,
> 
> I noticed some issues with how grandstream handles
> stun test. GS is running version 1.0.4.50. First I
> reset the NAT router. Then reboot GS, get results of
> "restricted cone". Immediately reboot GS, get results
> "full cone". I tried quite a few public and commercial
> stun servers. Also tried different model/version of
> linksys routers. I always got the same issue. Winstun
> on the PC doesn't have this issue. Some ngrep on the
> stund 0.91 on Fedora linux revealed winstun had about
> 20 UDP packets back and forward. However GS only had
> less than 10.
> 
> Did anyone notice the same problem?
> 
> Thanks,
> Richard
> 
> 
> 
>   
>   
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Espia - Emmegi Srl

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[Asterisk-Users] grandstream and stun

[Richard]

Hi,

I noticed some issues with how grandstream handles
stun test. GS is running version 1.0.4.50. First I
reset the NAT router. Then reboot GS, get results of
"restricted cone". Immediately reboot GS, get results
"full cone". I tried quite a few public and commercial
stun servers. Also tried different model/version of
linksys routers. I always got the same issue. Winstun
on the PC doesn't have this issue. Some ngrep on the
stund 0.91 on Fedora linux revealed winstun had about
20 UDP packets back and forward. However GS only had
less than 10.

Did anyone notice the same problem?

Thanks,
Richard





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Re: [Asterisk-Users] DECT PC interface for cordless phones?

[Rainer Jochem]

On Sun, Apr 18, 2004 at 04:24:58PM +0800, [EMAIL PROTECTED] wrote:
> Hello.
> 
> Is anyone aware of a manufacturer of DECT cordless phone interface cards to act
> as the DECT base station that would be addressable channels under * ?

Kirk (http://www.kirktelecom.dk) has a DECT base station with an
Ethernet which speaks SCCP and H.323. So you should be able to connect
it to *. (http://www.kirktelecom.dk/company/suk207.asp)


Another interesting thing is from Dosch&Amand
(http://www.dasystems.de/index_en.php) who actually offer an PCI card 
called "COM-ON-AIR® PCI"
(http://www.dasystems.de/en/pdfs/produkte/com_on_air_pci_e.pdf)
but I'm not sure if this thing can be used as a base station also.
Perhaps you're lucky and they give you the specs and you can implement
a chan_dect for this card ;)


Greetings, 
 Rainer

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[Asterisk-Users] DECT PC interface for cordless phones?

[pjn]

Hello.
 
Is anyone aware of a manufacturer of DECT cordless 
phone interface cards to act
as the DECT base station that would be addressable 
channels under * ?
 
Regards
 
Petar