date:20060211

On Sat, Feb 11, 2006 at 08:29:31AM +, Peter Bowyer wrote:
 On 10/02/06, Jordan Novak [EMAIL PROTECTED] wrote:
 
  I am using Asterisk to send Voicemail out as Email. I am running into a
  problem I believe to be caused by the exchange server requiring SMTP
  authentication. I cannot get the sys admin's to turn it off. Does anyone
  know enough about sendmail to help me. 

Have you RTFMed? 

http://sendmail.org/ has, under the section Primary Sources for
Information:

please read the FAQ[1], as well as Compiling[2] and Configuration[3]
before asking any questions.

[1] http://sendmail.org/faq/
[2] http://sendmail.org/compiling.html
[3] http://sendmail.org/m4/readme.html

Under the Configuration link, the table of contents refers you to a page
about SMTP authentication:
http://sendmail.org/m4/smtp_auth.html

The relevant parts of it are the parts where sendmail is a SMTP client
to another SMTP server (the MS-Exchange server, in this case).

  I am assuming that the default
  mail client is sendmail. It will also send to other non-SMTP
  authenticated servers. Your help is much appreciated.

BTW: you really don't have to use sendmail. You can use just about any
other mailer that provider a /usr/sbin/sendmail program . postfix, exim
and qmail will also do. I personally prefer postfix. Generally stick to
the default one of your distro if you don't have a good reason to change
it, as it will probably be the most maitained.

 
 Install MSMTP as your local MTA (replacing sendmail). Configure
 Asterisk to use the local MTA, and configure MSMTP to forward to the
 Exchange server with authentication.
 
 http://msmtp.sourceforge.net/

The problem with msmtp and similar programs (ssmtp, nullmailer) is that
they don't queue. Thus if there was a temporary problem at the network
or the recieving side, the message is lost.

And frankly, you may not want every message from the crond to end over
remotely.

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

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[Asterisk-Users] FYI: new firmware for 7905/12 - RPID support

2006-02-11 Thread Pavel Jezek


maybe usefull for displaying CALLED party name when dialing
I'm remember, that this feature was planned to add to asterisk, any 
progress?

PJ




New and Changed Information

Release 8.0(0) includes the following new and enhanced features:

•Remote-Party ID support has been added for incoming INVITE and UPDATE 
requests and 18x and 200 responses.

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Re: [Asterisk-Users] No Voice when canreinvite=no

2006-02-11 Thread Jonathan Feally


Upgrade to 1.2.4 - bug in 1.2.2 - see www.asterisk.org front page.

-Jon

Kamran Ahmad wrote:


Hi all

I am using Asterisk 1.2.2 on frdora core 4. i have two
sip UA. if i put canreinvite=yes voice Ok on both
sides. and if i change canreinvite=no there is no
voice (media through asterisk) 


one thing more if i try to use playback application
for playing some sound file it is also working (like
exten = 500,1,Playback(demo-abouttotry) this is
working).

here is sip.conf

//sip.conf//

[general]
context=default
bindport=5060
bindaddr=0.0.0.0 
srvlookup=yes

allow=all
nat=no 


[6000]
type=peer
host=dynamic
context=default
canreinvite=yes
allow=all

[1000]
type=peer
host=dynamic
secret=1000
canreinvite=yes
allow=all


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[Asterisk-Users] Looking for Asterisk Platform for DID

2006-02-11 Thread john

ello,

Our company is looking for an Asterisk platform to
offer DID service to our existing clients. We are
located in Dammam Saudi Arabia. Our local client base
is 97% business from small stores to large businesses
and production companies. We also have a global client
base but are more targeting our business clients.

We currently offer many services such as pc2phone,
callback, prepaid calling cards and IP PHONE desktop
solution. From a few meetings and conversations with
our clients we found DID (Direct Inward Dialing) may
be a great service to offer.

Contact me offlist.

John



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Re: [Asterisk-Users] TE411P Really Bad Echo


On Sat, 11 Feb 2006, Rob Lith wrote:

It claims to have carrier-grade algorithms - don't glibly translate that
to carrier grade hardware, it's a PCI card...


What echo canceller hardware do you recommend for an asterisk PC?

-Dan
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[Asterisk-Users] Qwest disconnect supervision?

Anyone managed to get Qwest to enable disconnect supervision on analogue 
lines?


-Dan
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[Asterisk-Users] Problem with CLI output on [EMAIL PROTECTED]

Hello Gurus, here's my problem:

I downloaded and installed [EMAIL PROTECTED] (the version sporting Asterisk
1.2.4) and now I've got the following problem on the CLI output console
(Alt+F9): Most text is fine, except something that looks to me like
parameters. I don't really know how to explain this so I'm going to give an
example:

It show text like:
sample
Goto (loca_gateway,cifra,500)
Executing XX(XX, X) in new stack
Playing 'digits/1' (language 'en')
Executing XX(, X) in new stack
/sample

In my sample all the  text is actually something impossible to reproduce
with a keyboard. It looks like code-page hell. It looks like random chars
with code  128!

Everything else looks fine.
Is there a way to fix this? Where do I look?

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[Asterisk-Users] Help with dialplan

I've got a Mobile-to-PBX gateway installed and I want the ability to dial
from my mobile phone into my PBX and next dial a land-line from the PBX so I
can make cheep mobile-to-land-line calls while on the go.

I've contemplated using the WaitExten application but it only seems to wait
for ONE digit! Is there a way to put the calling mobile phone into a context
and wait for a full-length extension?

Thanks!

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[Asterisk-Users] Dialing part of the extension

I know this one must be easy but I'm an newbye so please help.
In my extensions.conf I want to have a line like:

Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)

Ie: I want to dial all the XXX-es, but not the 9;
How do I do that? What do I write in place of ${SOMETHING}? Navigating the
wiki didn't provide any usefull advice...

Thanks.

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RE : [Asterisk-Users] Dialing part of the extension

2006-02-11 Thread Olivier.taylor

Exten = 9XX,1,Dial(Zap/4/${EXTEN:1},40,r)

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Cosmin
Prund
Envoyé : samedi 11 février 2006 12:19
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Dialing part of the extension


I know this one must be easy but I'm an newbye so please help. In my
extensions.conf I want to have a line like:

Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)

Ie: I want to dial all the XXX-es, but not the 9;
How do I do that? What do I write in place of ${SOMETHING}? Navigating
the wiki didn't provide any usefull advice...

Thanks.

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Re: [Asterisk-Users] asterisk logger - urgent!!!

2006-02-11 Thread Bob Goddard

On Saturday 11 Feb 2006 00:10, Kevin P. Fleming wrote:
 Warren Burstein wrote:
  How about if it would set a global variable before each disk write so
  the SIGFSZ handler would know which file caused it?

 Ha!

 Signals are asynchronous. This global variable would to be
 lock-protected, would require copying (possibly long) paths for every
 write, and would not necessarily be correct when the signal arrived.

 Sorry, this is not a solution. There is no solution, other than paying
 attention to your server and making sure that files don't get
 ridiculously large.

Well, you seem to have totally ignore what I said.

Using fopen/fputs to ONLY append to a file, is quite frankly, stupid.
Change it to open/write and you will be able to trap via the write
return code and errno.


B


-- 
http://www.mailtrap.org.uk/
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RE: [Asterisk-Users] Dialing part of the extension

Thanks, it works!

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Olivier.taylor
 Sent: Saturday, February 11, 2006 1:28 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE : [Asterisk-Users] Dialing part of the extension
 
 Exten = 9XX,1,Dial(Zap/4/${EXTEN:1},40,r)
 
 Olivier
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de Cosmin
 Prund
 Envoyé : samedi 11 février 2006 12:19
 À : asterisk-users@lists.digium.com
 Objet : [Asterisk-Users] Dialing part of the extension
 
 
 I know this one must be easy but I'm an newbye so please help. In my
 extensions.conf I want to have a line like:
 
 Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)
 
 Ie: I want to dial all the XXX-es, but not the 9;
 How do I do that? What do I write in place of ${SOMETHING}? Navigating
 the wiki didn't provide any usefull advice...
 
 Thanks.
 
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[Asterisk-Users] configure TE205P on [EMAIL PROTECTED]

2006-02-11 Thread nik600

hi

i'm trying to configure a TE205P on [EMAIL PROTECTED]

i've edited /etc/sysconfig/zaptel adding this line:

MODULES=$MODULES wct2xxp

now, when the system is loading, i can see that the wct2xxp module is
loaded correctly

but if i try the command:
 /usr/local/sbin/genzaptelconf

i get:

STOPPING ASTERISK

STOPPING FOP SERVER
Generating  '/etc/zaptel.conf'
Generating  '/etc/asterisk/zapata-auto.conf'


STOPPING ASTERISK

STOPPING FOP SERVER
Unloading zaptel hardware drivers:.
Removing zaptel module:  ERROR: Module zaptel is in use by wct4xxp
   [FAILED]
Loading zaptel framework:  [  OK  ]
Waiting for zap to come online...OK
Loading zaptel hardware modules: wct2xxpRunning ztcfg: [  OK  ]

SETTING FILE PERMISSIONS
Permissions OK

STARTING ASTERISK
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.

-
Asterisk could not start!
Use 'tail /var/log/asterisk/full' to find out why.
-
Unable to connect to remote asterisk (does
/var/run/asterisk/asterisk.ctl exist?)

the file asterisk.ctl esists...
[EMAIL PROTECTED] /]# ls -la /var/run/asterisk/asterisk.ctl
srwxr-xr-x  1 asterisk asterisk 0 Feb 11 07:06 /var/run/asterisk/asterisk.ctl

and this is what is reported in the logs:
[EMAIL PROTECTED] /]# tail /var/log/asterisk/full
Feb 11 07:06:05 VERBOSE[4808] logger.c:  [chan_zap.so]Feb 11 07:06:05
VERBOSE[4808] logger.c:  [chan_zap.so] = (Zapata Telephony w/PRI)
Feb 11 07:06:05 VERBOSE[4808] logger.c:   == Parsing
'/etc/asterisk/zapata.conf': Feb 11 07:06:05 VERBOSE[4808] logger.c:  
== Parsing '/etc/asterisk/zapata.conf': Found
Feb 11 07:06:05 VERBOSE[4808] logger.c:   == Parsing
'/etc/asterisk/zapata-auto.conf': Feb 11 07:06:05 VERBOSE[4808]
logger.c:   == Parsing '/etc/asterisk/zapata-auto.conf': Found
Feb 11 07:06:05 VERBOSE[4808] logger.c:   == Parsing
'/etc/asterisk/zapata_additional.conf': Feb 11 07:06:05 VERBOSE[4808]
logger.c:   == Parsing '/etc/asterisk/zapata_additional.conf': Found
Feb 11 07:06:05 WARNING[4808] chan_zap.c: Unable to specify channel 1:
No such device or address
Feb 11 07:06:05 ERROR[4808] chan_zap.c: Unable to open channel 1: No
such device or address
here = 0, tmp-channel = 1, channel = 1
Feb 11 07:06:05 ERROR[4808] chan_zap.c: Unable to register channel '1-23'
Feb 11 07:06:05 WARNING[4808] loader.c: chan_zap.so: load_module
failed, returning -1
Feb 11 07:06:05 WARNING[4808] loader.c: Loading module chan_zap.so failed!
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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-11 Thread bbench

 On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
 I've been playing with astcc, but while
 'billseconds' stays empty, 'billcost' has
 strange behavior - either stays ampty
 or takes ONCE the Connect fee(if I put one)
 and keeps it that way no matter how long
 the call is ...( if no Connect fee -stays empty).
 i.e.
 [inbound]
 exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
 exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
 exten = 1122334455,3,Hangup
 
 Michiel van Baak wrote:
 DeadAGI is for hungup channels, not for active channels.
 That might be a problem.
 
 Try this:
 exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
 
 On Monday 06 February 2006 09:25, JP Carballo wrote:
 ASTCC works fine here. The duration and billseconds fields in my cdrs as
 well as ASTCC's cdr are filled.
 I don't use the connect fee field though and all are set to 0.
 
 Would you share with me how'd you do billing on a DID
 (if you do), and through what Technology?
 Anything that goes Local here is ANSWEREDTIME zero.

On Saturday 11 February 2006 06:32, Darren Wiebe wrote:
 Are you running a relatively recent version of ASTCC?  Say within the
 last 6 months.  The answeredtime = 0 bug was supposed to have been fixed
 by http://bugs.digium.com/view.php?id=4300  Unless something has changed
 in Asterisk that affects this

Thanks Daren,
Yes, my version of astcc is the most recent one.
Asterisk-1.2.4
I have found you patch 0004300 from 16 May 2005.
Probably it's time to reverse it back since something has changed
in Asterisk that affects this... as you said.
My observation is:
If I keep:
$dialstr = Local/[EMAIL PROTECTED]{path}|30|HL/n( . ($maxtime * 60 * 1000) . 
:6:3);
Either the billseconds is empty(when dial out through Local), either there is 
aZOMBIE when dialing in. 
I put back the dialstring to:
Local/$phone/$res-{path}|30|HL/n( . ($maxtime * 60 * 1000) . 
:6:3);
The only difference that it looks only for is a default context.

extensions.conf
[inbound]
; 10 digits DID = _XX = cardnumber
; 
exten = _XX ,1,Answer()
exten = _XX ,n,Set(DB(RCID/${CALLERIDNUM})=${CALLERIDNUM})
exten = _XX ,n,Set(realcid=${DB(RCID/${CALLERIDNUM})})
exten = _XX ,n,Noop(${REALCID})
;exten = _XX ,n,Set(TIMEOUT(digit)=4)
exten = _XX ,n,Set(CALLERID(number)=${EXTEN})
exten = _XX ,n,Set(CALLERID(name)= ${REALCID})
;exten = t,3,Goto(h|1)
;exten = _XX 2,Goto(s|1)
;exten = s,1,Wait,1 ; is this preventing HUP?
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${CALLERIDNUM},4) 
; must be h,1 as per Michiel van Baak note(above).
exten = h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten = 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include = internal
[personal]
exten = t,1,Hangup
include = inbound

Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but looong problem - Timeout about 10 
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.

Sorry for the long explanation.
What do you think? Is there something suspicious in
that solution?
Thanks,
benchev

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Re: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Rob Lith

TE406P/411P and if you need to go dedicated to hanlde all possible look at an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO CANCELLER (1U Version) ± $1295
RobOn 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sat, 11 Feb 2006, Rob Lith wrote: It claims to have carrier-grade algorithms - don't glibly translate that
 to carrier grade hardware, it's a PCI card...What echo canceller hardware do you recommend for an asterisk PC?-Dan___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Stagg Shelton





Yes, right now we are only using span 1 on the quad span card with
plans to pull in another T1 PRI when we get this echo problem solved.
The echo is only experienced when the call terminates to traditional
analog circuits both local and long distance. Calls to cell phones,
and other known digital circuits do not exhibit the symptoms. I've
been sent a few articles off list which discussed the reasons why this
may occur, imperfect impedance where the digital circuit is switched to
the analog circuit.

In testing I've set echocancel=no made calls, and echocancel=yes and
made calls with no real audible difference. I haven't done a zap show
channel while on a call with echo, but I plan to this weekend. I was
informed that the number of taps would be shown. I used a stop watch
last night to try to get the delay, and it was about 1/2 to 1 second
delay and was continual so long as I was talking. It wasn't affected
by differing acoustical variations. I tested this using handset,
headset, and speakerphone. Disabling VPM and recompiling zaptel or
removing the VPM off the board completely is the only thing that has
any effect on the echo. Hopefully the zap show channel will provide to
me another data point to help me determine if the HW module is active.

More to come...

Stagg Shelton
www.oneringnetworks.com


Cory Andrews wrote:

  
  
  
  
  Stagg - I know you get a full 128ms
tail of echo can on the Sangoma. I believe that on the TE411P, the
128ms tail is shared by all (4) spans, and as you add additional spans
up to maximum of 4, the echo can tail amount decreases accordingly. If
you are running 4 spans, you have 32ms of echo can tail on each span,
not the full 128ms.
  
  Now that I'm reading back through
your post thread, it looks like you were only running 1 span on the
TE411P, so you should have been getting the full 128ms of echo can
tail. You may not find an improvement with the Sangoma. I have never
experience, nor heard of a 1-2 full second delay. Trial and error is
likely your course of action. 
  
  Cory J Andrews

VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY
  
-
Original Message - 
From:
Stagg Shelton 
To:
Asterisk Users Mailing
List - Non-Commercial Discussion 
Sent:
Friday, February 10, 2006 11:34 PM
Subject:
Re: [Asterisk-Users] TE411P Really Bad Echo


It was Digium's opinion that perhaps the card had a VPM. We got a
replacement TE411P, I implemented it tonight and still the exact same
echo problem. At this point I feel like I can rule out failed
hardware. 

I contacted Digium support and now they are telling me it's something
with my carrier, and I should call them. I called Bellsouth, and they
ran a full stress test on the circuit taking me offline for about 30
minutes. 

The end result is that the circuit test passed with no errors.
Bellsouth says it's not in their network, Digium says its not their
card, and I have a te411p with VPM disabled in the wct4xx kernel module
because something doesn't work the way it should. My customer is
wanting to know about sangoma cards with the echo cancellation, and at
this point I'm nervous to recommend any hardware. I'm going to look
into the sangoma that you suggested. Are there any other kinds of
products that I could look into either Passive or Active.

Thanks 

Stagg Shelton
www.oneringnetworks.com


Matt wrote:

  try sangoma carrier grade 104d hardware EC card. we're using it ourself.

Best Regards

Matt
- Original Message - 
From: "Anthony Rodgers" [EMAIL PROTECTED]
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@lists.digium.com
Sent: Tuesday, February 07, 2006 12:57 PM
Subject: Re: [Asterisk-Users] TE411P Really Bad Echo


  
  
For what it's worth, we have been going through very similar issues
with a TE411P - with Digium support, we have basically gone as far as
we can with the HW EC, and are now using MG2 with much better results.

We have a Ditech EC box on order.

Regards,
-- 
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Feb 7, 2006, at 7:36 AM, Matthew Fredrickson wrote:



  On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote:

  
  
I just implemented a system using a TE411P hardware echo cancellation
card. Per Digium, I setup zaptel.conf, and zapata.conf the same way

  
  as
  
  
I always have. To my surprise calls out to the PSTN had a terrible
echo. 1 - 2 second delay, and quite clear. The echo was so bad that

  
  I
  
  
had to remove the hardware echo cancellation module from the card.

  
  We
  
  
are only using the 1st span of

RE: [Asterisk-Users] TE411P Really Bad Echo ORION

2006-02-11 Thread Darren Wright









The Orion echo canceller is just ok.     The
Tellabs units work just as well if you dont mind 10 mins of soldering.



I have the orion running with an adit 600
and a TE110P.   Echo cancel is fairly good, but I have loads of problems with
DTMF digits.



-Darren

    













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Rob Lith
Sent: Saturday, February 11, 2006
8:10 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
TE411P Really Bad Echo





TE406P/411P and if you
need to go dedicated to hanlde all possible look at an external dedicated
canceller like www.oriontelecom.com
VCL-E1 ECHO CANCELLER (1U Version) ± $1295

Rob



On 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


On Sat, 11 Feb 2006, Rob Lith wrote:
 It claims to have carrier-grade algorithms - don't glibly
translate that 
 to carrier grade hardware, it's a PCI card...

What echo canceller hardware do you recommend for an asterisk PC?

-Dan
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[Asterisk-Users] Can I configure the console to ring on one sound card and the headset on another sound card?

2006-02-11 Thread Anthony Azzopardi

Can I configure the console to ring on one sound card and the headset on 
another sound card?

Best regards,
Anthony.

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Re: [Asterisk-Users] Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *

2006-02-11 Thread Johnathan Corgan

Andres wrote:

 There is no username in the above To header.  Check your DIAL command
 because something is wrong here.  Thats why you get a 404.  The SPA
 can't match the username.

Yes.  I had not reverted to an early enough commit on the configuration
files and the usernames were still missing in sip.conf.

Thanks.

-Johnathan
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Re: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Rob Lith

StaggI don't think it's a matter of trying echocancel on and off, it is a matter of tuning your system to your local PSTN - this is a combination of trying the different echocan alogrithims (i.e. MG2), the echotrainign etc and setting your txgain - too loud outgoing audio will result in echo.
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718
RobOn 2/11/06, Stagg Shelton [EMAIL PROTECTED] wrote:

Yes, right now we are only using span 1 on the quad span card with
plans to pull in another T1 PRI when we get this echo problem solved.
The echo is only experienced when the call terminates to traditional
analog circuits both local and long distance. Calls to cell phones,
and other known digital circuits do not exhibit the symptoms. I've
been sent a few articles off list which discussed the reasons why this
may occur, imperfect impedance where the digital circuit is switched to
the analog circuit.

In testing I've set echocancel=no made calls, and echocancel=yes and
made calls with no real audible difference. I haven't done a zap show
channel while on a call with echo, but I plan to this weekend. I was
informed that the number of taps would be shown. I used a stop watch
last night to try to get the delay, and it was about 1/2 to 1 second
delay and was continual so long as I was talking. It wasn't affected
by differing acoustical variations. I tested this using handset,
headset, and speakerphone. Disabling VPM and recompiling zaptel or
removing the VPM off the board completely is the only thing that has
any effect on the echo. Hopefully the zap show channel will provide to
me another data point to help me determine if the HW module is active.

More to come...

Stagg Shelton
www.oneringnetworks.com

Cory Andrews wrote:

Stagg - I know you get a full 128ms
tail of echo can on the Sangoma. I believe that on the TE411P, the
128ms tail is shared by all (4) spans, and as you add additional spans
up to maximum of 4, the echo can tail amount decreases accordingly. If
you are running 4 spans, you have 32ms of echo can tail on each span,
not the full 128ms.

Now that I'm reading back through
your post thread, it looks like you were only running 1 span on the
TE411P, so you should have been getting the full 128ms of echo can
tail. You may not find an improvement with the Sangoma. I have never
experience, nor heard of a 1-2 full second delay. Trial and error is
likely your course of action.

Cory J Andrews

VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
AIM - B2CORY

-
Original Message -

From:
Stagg Shelton
To:
Asterisk Users Mailing
List - Non-Commercial Discussion
Sent:
Friday, February 10, 2006 11:34 PM
Subject:
Re: [Asterisk-Users] TE411P Really Bad Echo

It was Digium's opinion that perhaps the card had a VPM. We got a
replacement TE411P, I implemented it tonight and still the exact same
echo problem. At this point I feel like I can rule out failed
hardware.

I contacted Digium support and now they are telling me it's something
with my carrier, and I should call them. I called Bellsouth, and they
ran a full stress test on the circuit taking me offline for about 30
minutes.

The end result is that the circuit test passed with no errors.
Bellsouth says it's not in their network, Digium says its not their
card, and I have a te411p with VPM disabled in the wct4xx kernel module
because something doesn't work the way it should. My customer is
wanting to know about sangoma cards with the echo cancellation, and at
this point I'm nervous to recommend any hardware. I'm going to look
into the sangoma that you suggested. Are there any other kinds of
products that I could look into either Passive or Active.

Thanks

Stagg Shelton
www.oneringnetworks.com

Matt wrote:

try sangoma carrier grade 104d hardware EC card. we're using it ourself.Best RegardsMatt- Original Message - From: Anthony Rodgers
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comSent: Tuesday, February 07, 2006 12:57 PMSubject: Re: [Asterisk-Users] TE411P Really Bad Echo

For what it's worth, we have been going through very similar issueswith a TE411P - with Digium support, we have basically gone as far aswe can with the HW EC, and are now using MG2 with much better results.
We have a Ditech EC box on order.Regards,-- Anthony RodgersBusiness Systems AnalystDistrict of North VancouverWeb:
http://www.dnv.orgRSS Feed: http://www.dnv.org/rss.aspOn Feb 7, 2006, at 7:36 AM, Matthew Fredrickson wrote:

On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote:

I just implemented a system using a TE411P hardware echo cancellationcard. Per Digium, I setup zaptel.conf, and zapata.conf the same

Re: [Asterisk-Users] TE411P Really Bad Echo ORION


Darren Wright wrote:


The Orion echo canceller is just ok. The Tellabs units work just as 
well if you don’t mind 10 mins of soldering.


I have the orion running with an adit 600 and a TE110P. Echo cancel is 
fairly good, but I have loads of problems with DTMF digits.



And, are usually found for under $100US on ebaY.

Doug


Ben Franklin quote:

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


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Re: [Asterisk-Users] Error running iaxcomm

On Sat, Feb 11, 2006 at 10:17:37AM +0500, ast guy wrote:
 Hi,
  I have downloaded iaxcomm version iaxcomm-lin-1.0rc3, when I try to
 execute it it gives following error.
 
 # ./iaxcomm
 Error wxWindows Fatal Error : Couldn't Initialize IAX Client .
 
 any idea what's going wrong ?

No. But this is the mailing list for Asterisk, not for
iaxclient/iaxcomm. Try their mailing list.

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

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Re: [Asterisk-Users] asterisk logger - urgent!!!

2006-02-11 Thread Kevin P. Fleming

Bob Goddard wrote:

 Using fopen/fputs to ONLY append to a file, is quite frankly, stupid.
 Change it to open/write and you will be able to trap via the write
 return code and errno.

Patches to fix bugs are most welcome. Given that these files are written
using fprintf (because they are using format strings and long lists of
arguments), using write will require allocating memory, building the
output there and then calling write(). Seems like an awful lot of work
to avoid a simple system administration failure.
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Re: [Asterisk-Users] OH323 Peer

2006-02-11 Thread Abdul Lateef

Hi,

i treid this 
OH323/ipgateway:port
and working well for me. But i need to add some more
featurres, like some of my H323 GW supporting only
G.7231 codec and some one G.729 and others feature
like rtptimeout etc

So if i am direct dialing without these feautres, the
GW are not able to handel my calls.

Any more suggestion..?







Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com

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Re: [Asterisk-Users] TE411P Really Bad Echo

I thought that if the VPM was detected then you didn't have any control as to 
which algorithm was used.  

I was under the impression that the algorithms were only used for the software 
echo cancellation.  

At this point I'll give anything a try.

Stagg Shelton
www.oneringnetworks.com

-Original Message-

From:  Rob Lith [EMAIL PROTECTED]
Subj:  Re: [Asterisk-Users] TE411P Really Bad Echo
Date:  Sat Feb 11, 2006 10:17 am
Size:  4K
To:  Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com

Stagg

I don't think it's a matter of trying echocancel on and off, it is a matter
of tuning your system to your local PSTN - this is a combination of trying
the different echocan alogrithims (i.e. MG2), the echotrainign etc and
setting your txgain - too loud outgoing audio will result in echo.
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718

Rob

On 2/11/06, Stagg Shelton [EMAIL PROTECTED] wrote:

 Yes, right now we are only using span 1 on the quad span card with plans
 to pull in another T1 PRI when we get this echo problem solved.  The echo is
 only experienced when the call terminates to traditional analog circuits
 both local and long distance.  Calls to cell phones, and other known digital
 circuits do not exhibit the symptoms.  I've been sent a few articles off
 list which discussed the reasons why this may occur, imperfect impedance
 where the digital circuit is switched to the analog circuit.

 In testing I've set echocancel=no made calls, and echocancel=yes and made
 calls with no real audible difference.  I haven't done a zap show channel
 while on a call with echo, but I plan to this weekend.  I was informed that
 the number of taps would be shown.  I used a stop watch last night to try to
 get the delay, and it was about 1/2 to 1 second delay and was continual so
 long as I was talking.  It wasn't affected by differing acoustical
 variations.  I tested this using handset, headset, and speakerphone.
 Disabling VPM and recompiling zaptel or removing the VPM off the board
 completely is the only thing that has any effect on the echo.  Hopefully the
 zap show channel will provide to me another data point to help me determine
 if the HW module is active.

 More to come...

 Stagg Shelton
 www.oneringnetworks.com


 Cory Andrews wrote:

 Stagg - I know you get a full 128ms tail of echo can on the Sangoma.  I
 believe that on the TE411P, the 128ms tail is shared by all (4) spans, and
 as you add additional spans up to maximum of 4, the echo can tail amount
 decreases accordingly.  If you are running 4 spans, you have 32ms of echo
 can tail on each span, not the full 128ms.

 Now that I'm reading back through your post thread, it looks like you were
 only running 1 span on the TE411P, so you should have been getting the full
 128ms of echo can tail.  You may not find an improvement with the Sangoma.
 I have never experience, nor heard of a 1-2 full second delay.  Trial and
 error is likely your course of action.

 Cory J Andrews
 
 VOIPSupply.com
 454 Sonwil Drive
 Buffalo, NY 14225
 ++
 voice - 716.630.1555 X22
 email - [EMAIL PROTECTED]
 AIM - B2CORY

 - Original Message -
 *From:* Stagg Shelton [EMAIL PROTECTED]
 *To:* Asterisk Users Mailing List - Non-Commercial 
 Discussionasterisk-users@lists.digium.com
 *Sent:* Friday, February 10, 2006 11:34 PM
 *Subject:* Re: [Asterisk-Users] TE411P Really Bad Echo

  It was Digium's opinion that perhaps the card had a VPM.  We got a
 replacement TE411P, I implemented it tonight and still the exact same echo
 problem.  At this point I feel like I can rule out failed hardware.

 I contacted Digium support and now they are telling me it's something with
 my carrier, and I should call them.  I called Bellsouth, and they ran a full
 stress test on the circuit taking me offline for about 30 minutes.

 The end result is that the circuit test passed with no errors.  Bellsouth
 says it's not in their network, Digium says its not their card, and I have a
 te411p with VPM disabled in the wct4xx kernel module because something
 doesn't work the way it should.  My customer is wanting to know about
 sangoma cards with the echo cancellation, and at this point I'm nervous to
 recommend any hardware.  I'm going to look into the sangoma that you
 suggested.  Are there any other kinds of products that I could look into
 either Passive or Active.

 Thanks

 Stagg Shelton
 www.oneringnetworks.com


 Matt wrote:

 try sangoma carrier grade 104d hardware EC card. we're using it ourself.

 Best Regards

 Matt
 - Original Message -
 From: Anthony Rodgers [EMAIL PROTECTED] [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com asterisk-users@lists.digium.com
 Sent: Tuesday, February 07, 2006 12:57 PM
 Subject: Re: [Asterisk-Users] TE411P Really Bad Echo


For what it's worth, we have been going

Re: [Asterisk-Users] more cpu intensive echo cancellers ?

2006-02-11 Thread Matthew Fredrickson



On Feb 10, 2006, at 10:25 PM, Steve Underwood wrote:


Matthew Fredrickson wrote:



On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:



Found it, going to go test it right now :) thanks!
So far reports have been positive on the echo, but its a slow day ;)
We're using cisco 7960 phones, they're pricy, but they work great and
sound good, if it wasn't for the echo issue, I would have been able 
to

roll the whole setup out already.
Actually that's not quite true, I still have to make the 7914 addon
module work with the 7960 phone, but that's not a show stopper.

Either way, so far big thumbs up for the MG2 echo can, and if any
developers read this, feel free to add a compile flag to make it more
cpu intensive ;) and do more canceling.



Does latest MG2 behave better than KB1 on your analog lines?  I heard 
in the past that in some cases (primarily with analog lines) that KB1 
worked better.  Also, have you tried the echotraining=800  (in 
zapata.conf) tweak as well?


A lot of the variability is probably due to thr lack of a DC blocker 
at the front of the echo canceller. As far as I remember, none of the 
cancellers in * has a DC blocker.




Where can one find out more information on writing a DC blocker?  I 
google'd around a bit, but couldn't find a definitive overview of what 
one was, and how to write one.  Thanks!


---
Matthew Fredrickson

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Re: [Asterisk-Users] What ATA should I buy?

2006-02-11 Thread Tele Cost Price Reducer

Hi Sam Tam,
i would be interested in these ATA that you can offer.
please provide me with more details about this option.

thank you very much,

Mickey Lazar
On 2/9/06, Sam Tam [EMAIL PROTECTED] wrote:
We have got some ATA for only $55 if you are interested?Sam-Original Message-From: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Michael
SampsonSent: Thursday, February 09, 2006 11:01 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] What ATA should I buy?I've used the spa-1001 and the spa-2001 for faxes. Works good over a
local area network. thevoipconnection sells those for about 60 bucks though.Michael SampsonInformation Systems ManagerCustomer Contact Services[EMAIL PROTECTED]
952-936-4000Tomislav Parčina wrote:I have running * without any Digium (or any other) hardware. Now I need toconnect analog FAX machine to it. I think that cheapest and easiest way is
to buy ATA. Please correct me if I'm wrong.Now, which ATA should I buy? Local dealer sells those four. I can buysomething else (if there is any reason for it), but I prefer something ofthis.
One more question, can I plug two lines in any of those ATA-s?Sipura SPA-2100 SIP-ATA160$Sipura SPA-1001 SIP-ATA125$ALL7902 IP SIP ATA Adapter / Router106$
Grandstream HandyTone ATA486 142$Thank you for any suggestions.P.S.If this is second time you see this message, then sorry for resending, butI didn't see it on list...
--Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)393447e-mail: tparcina#lama.hrhttp://www.lama.hr
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Re: [Asterisk-Users] more cpu intensive echo cancellers ?

2006-02-11 Thread Steve Underwood


Matthew Fredrickson wrote:



On Feb 10, 2006, at 10:25 PM, Steve Underwood wrote:


Matthew Fredrickson wrote:



On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:



Found it, going to go test it right now :) thanks!
So far reports have been positive on the echo, but its a slow day ;)
We're using cisco 7960 phones, they're pricy, but they work great and
sound good, if it wasn't for the echo issue, I would have been able to
roll the whole setup out already.
Actually that's not quite true, I still have to make the 7914 addon
module work with the 7960 phone, but that's not a show stopper.

Either way, so far big thumbs up for the MG2 echo can, and if any
developers read this, feel free to add a compile flag to make it more
cpu intensive ;) and do more canceling.



Does latest MG2 behave better than KB1 on your analog lines?  I 
heard in the past that in some cases (primarily with analog lines) 
that KB1 worked better.  Also, have you tried the echotraining=800  
(in zapata.conf) tweak as well?



A lot of the variability is probably due to thr lack of a DC blocker 
at the front of the echo canceller. As far as I remember, none of the 
cancellers in * has a DC blocker.




Where can one find out more information on writing a DC blocker?  I 
google'd around a bit, but couldn't find a definitive overview of what 
one was, and how to write one.  Thanks!


DC in the signal through the echo canceller represents a signal the 
canceller's adaption can never eliminate. It fights; it fails; it many 
get very upset trying. DC needs to be eliminated before cancellation. 
A-law/u-law ports are not supposed to give you any DC, but some do. The 
following will estimate and remove DC from the signal. Prime 
dc_estimate with zero.


int 16_t dc_removal(int32_t dc_estimate, int16_t sample)
{
   dc_estimate += int32_t) sample  15) - dc_bias)  9);
   sample -= (dc_estimate  15);
   return sample;
}

Its a first order noise shaped single pole IIR. '9' is the damping 
factor. If you make it bigger, the low frequency response will improve, 
but the estimate will take longer to settle after step changes. This may 
affect initial convergence if a DC hiccup occurs as the line is picked 
up. 9 should be a good starting point to try.


Regards,
Steve

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Re: [Asterisk-Users] What ATA should I buy?

2006-02-11 Thread Paul

The -biz list is more appropriate for this.

Tele Cost Price Reducer wrote:

 Hi Sam Tam,
 i would be interested in these ATA that you can offer.
 please provide me with more details about this option.
  
 thank you very much,
  
 Mickey Lazar

  
 On 2/9/06, *Sam Tam* [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:

 We have got some ATA for only $55 if you are interested?

 Sam

 -Original Message-
 From: [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]] On Behalf Of
 Michael
 Sampson
 Sent: Thursday, February 09, 2006 11:01 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] What ATA should I buy?

 I've used the spa-1001 and the spa-2001 for faxes. Works good over a
 local area network. thevoipconnection sells those for about 60
 bucks though.

 Michael Sampson
 Information Systems Manager
 Customer Contact Services
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 952-936-4000



 Tomislav Parčina wrote:

 I have running * without any Digium (or any other) hardware. Now
 I need to
 connect analog FAX machine to it. I think that cheapest and
 easiest way is
 to buy ATA. Please correct me if I'm wrong.
 
 Now, which ATA should I buy? Local dealer sells those four. I can buy
 something else (if there is any reason for it), but I prefer
 something of
 this.
 
 One more question, can I plug two lines in any of those ATA-s?
 
 Sipura SPA-2100 SIP-ATA160$
 Sipura SPA-1001 SIP-ATA125$
 ALL7902 IP SIP ATA Adapter / Router106$
 Grandstream HandyTone ATA486   142$
 
 
 Thank you for any suggestions.
 
 
 P.S.
 If this is second time you see this message, then sorry for
 resending, but
 I didn't see it on list...
 
 
 --
 Tomislav Parčina
 Lama Computers Split
 Stinice 12, 21000 Split
 Tel.: +385(21)393447
 e-mail: tparcina#lama.hr
 http://www.lama.hr http://www.lama.hr
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[Asterisk-Users] Asterisk 1.2.4 and IAX MOH

Has anybody has issues with the new Native MOH and IAX trunking when 
placing a call on hold?


My scenario,

Call is placed on a Definity G3 via PRI to Asterisk.  Gets trunked over 
to another Asterisk system via IAX2.  Call is answered by operator and 
placed on hold.  At that point, audio is very broken and feedback 
pulsing is heard.  Bad enough that the caller hangs up.  If the call is 
answered before the caller hangs up, audio is broken for the first 10 
seconds, then clears.


In testing this weekend, I've moved inter-office on hold music back to 
mpg123 and the problem went away.


Any suggestion on this one?

Thanks,

Doug

--
Ben Franklin quote:

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


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Re: [Asterisk-Users] Asterisk 1.2.4 and IAX MOH



Doug Lytle wrote:
Has anybody has issues with the new Native MOH and IAX trunking when 
placing a call on hold?


Actually, I meant to say, the call is parked.

Doug

--
Ben Franklin quote:

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


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Re: [Asterisk-Users] Sendmail with exchange

2006-02-11 Thread Sean Cook

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

The only way you would need authenticated SMTP is for relaying.  My
suggestion would be to not set up sendmail to use a smart host but have
it act as an internet mail server.  It will lookup the mx records and
make the sending determinations based on the domain it is sending to.

The exchange server should accept (with out authentication) anything
that it is addressed to a locally hosted domain.

Sean

kevin ling wrote:
 Hi,
 
 Can you make some test to send voicemail to other mail account? (e.g,
 @yahoo.com, @hotmail.com...). If it's work. I think not a SMTP authetication
 problem. Or you can check the asterisk maillog first.
 
 Kevin 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak
 Sent: Saturday, February 11, 2006 5:42 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Sendmail with exchange
 
  
 I am using Asterisk to send Voicemail out as Email. I am running into a
 problem I believe to be caused by the exchange server requiring SMTP
 authentication. I cannot get the sys admin's to turn it off. Does anyone
 know enough about sendmail to help me. I am assuming that the default mail
 client is sendmail. It will also send to other non-SMTP authenticated
 servers. Your help is much appreciated.
 
 Jordan Novak
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Re: [Asterisk-Users] Expression GotoIf - bug or personal misunderstanding?


At 12:51 AM 02/10/2006, you wrote:

-- Executing GotoIf(Zap/29-1, 1  0?4:3) in new stack
-- Goto (macro-stdexten,s-NOANSWER,4)


Should look like:


GotoIf( $[1  0]?4:3 )


Ira



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Re: [Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)



At 07:08 AM 02/10/2006, you wrote:
I mean, can I write
the following two lines in only one line?

exten= 12345/100,1,Hangup
exten= 12345/200,1,Hangup
I would think this would work:
exten= 12345/[1-2]00,1,Hangup

Ira




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Re: [Asterisk-Users] meetme application

2006-02-11 Thread Alexander Chemeris

Miguel,

On 2/11/06, Miguel [EMAIL PROTECTED] wrote:
 pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension
 i did a normal make, make install, did i miss something?
You need zaptel headers installed to build MeetMe application. And you
need zaptel devices (or ztdummy) to run MeetMe.
See voip-info.org for more information.


Regards,
Alexander Chemeris
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Re: [Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)


At 07:08 AM 02/10/2006, you wrote:

I mean, can I write the following two lines in only one line?

exten= 12345/100,1,Hangup
exten= 12345/200,1,Hangup


Oops, forgot the underscore!

I would think this would work:

exten= _12345/[1-2]00,1,Hangup


Ira



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[Asterisk-Users] Codec issue with my iaxy

2006-02-11 Thread Mark Ratering

I just bought a new IAXy box and am only achieving one way calling.  
Both iax.conf and the IAXy support ulaw and gsm.  When I try to call, 
however i get this error:


Feb 11 15:20:32 NOTICE[7963]: channel.c:1893 ast_read: Dropping 
incompatible voice frame on IAX2/iaxy-2 of format ilbc since our native 
format has changed to ulaw


I added support for ilbc on the IAXy and in iax.conf and am still 
getting this issue.  Any help would be greatly appriciated.

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Re: [Asterisk-Users] Dialing part of the extension


At 03:18 AM 02/11/2006, you wrote:

Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)

Ie: I want to dial all the XXX-es, but not the 9;


Exten = 9XX,1,Dial(Zap/4/${EXTEN:1},40,r)

Ira


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Re: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Mike Clark


Stagg Shelton wrote:

It was Digium's opinion that perhaps the card had a VPM.  We got a 
replacement TE411P, I implemented it tonight and still the exact same 
echo problem.  At this point I feel like I can rule out failed hardware. 

I contacted Digium support and now they are telling me it's something 
with my carrier, and I should call them.  I called Bellsouth, and they 
ran a full stress test on the circuit taking me offline for about 30 
minutes. 

The end result is that the circuit test passed with no errors.  
Bellsouth says it's not in their network, Digium says its not their 
card, and I have a te411p with VPM disabled in the wct4xx kernel 
module because something doesn't work the way it should.  My customer 
is wanting to know about sangoma cards with the echo cancellation, and 
at this point I'm nervous to recommend any hardware.  I'm going to 
look into the sangoma that you suggested.  Are there any other kinds 
of products that I could look into either Passive or Active.


Thanks

Stagg Shelton
www.oneringnetworks.com

If you only need a single span, think about using a single span Digium 
or Sangoma  card without echo cancellation and then use an external 
hardware echo canceller such as Tellabs or Orion Telecom. We have a 
customer using a Sangoma A101 that had some lingering echo, so we 
purchased the Orion Telecom desktop T1 echo canceller. That cleared 
things up and we've had no complaints in the almost two weeks since it 
has been installed. We've got another one on the way for a second site.


Mike Clark

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Re: [Asterisk-Users] IP Authorization

2006-02-11 Thread Darren Wiebe


It's part of ASTPP.  It is in astpp -head ready for testing.

Darren Wiebe
[EMAIL PROTECTED]

Sam Tam wrote:


When will it be ready ?

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: Saturday, February 11, 2006 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP Authorization

I'm doing it similar to what the posted showed today.  Then I'm calling 
an agi script (Maybe not the nicest way) that checks to see if the IP is 
allowed and sets the accountcode for the call.


Darren

Sam Tam wrote:

 


Can you be more detail about the setup?

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E
Johansson
Sent: Friday, February 10, 2006 4:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP Authorization

Sam Tam wrote:


   


I think this is a question that has been discussed before.
But you see nowadays most carriers will provide thing like SIP using IP 
authorization rather than username and password and I am now wondering 
whether Asterisk can do something like that or not?


  

 


In the voip channels as well as in manager you can set ACLs for the
connections you define.

/O
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing  Calling Cards
www.aleph-com.net/astpp

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[Asterisk-Users] MOH broke with 1.2.4 .. ?

2006-02-11 Thread Tim Connolly

/etc/asterisk/musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s

-- Executing Answer(Zap/1-1, ) in new stack
-- Executing MusicOnHold(Zap/1-1, ) in new stack
-- Started music on hold, class 'default', on channel 'Zap/1-1'
-- Stopped music on hold on Zap/1-1



I've got three mp3 files that worked fine on the latest cvs-head
version. With the upgrade to 1.2.4, I get no audio whatsoever. Any
suggestions? I cranked up verbose to 255 with no extra info.. Same with
debug.

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[Asterisk-Users] Busy signalling for mobile callers ?

2006-02-11 Thread Tristan Graham { Skymarket Limited }


Hi folks,

Got an oddity that a user has raised to do with busy signalling and 
inparticular when calling from a mobile phone. It seems that the 
behaviour when calling * is slightly different to the norm i.e. If I 
call an engaged landline number directly from my mobile then the mobile 
gives engaged tone plus displays number busy then hangs up. If however 
I call * and pass the call simply to Busy then I get either the 
regular engaged tones (priindication=inband) or three short tones 
followed by hangup (priindication=outofband). I thought this might be 
down to iSDN being in the path but I have tried a similar test using a 
PBX on an iSDN 30 and that seems to act normally.


Can anyone verify this behavour and tell me if it is normal or bug worthy ?

This is tested on asterisk 1.2.1, caller and callee in UK.

Tristan.
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Re: [Asterisk-Users] QSIG error -- can somebody explain?

2006-02-11 Thread Wolfgang Zweimueller

Johann Steinwendtner [EMAIL PROTECTED] writes:

 I can only guess, but I think I can remember that the creflen needs
 to be 2 octets for qsig. Check what the Alcatel switch sends in the 
 setup message to *.

Thanks, I will have a look at that.

 Anyway, why do use QSIG ? Does name display work on the * implementation  ?

It is not because of name display but of an issue with call routing
on this PBX. We have a running setup with Euro-ISDN. If we can switch
over to Q.SIG there would be a benefit for the customer.

 Best regards

 Hans

 P.S.: Schoene Gruesse an Kurt Krenn

Wir gemacht!

cu,
Wolfgang
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Re: [Asterisk-Users] configure TE205P on [EMAIL PROTECTED]

2006-02-11 Thread pdhales

The genzaptelconf doesn't work with E1/T1 cards in my experience.
You will have to configure it by hand.

PsulH

- Original Message - 
From: nik600 [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 11, 2006 11:09 PM
Subject: [Asterisk-Users] configure TE205P on [EMAIL PROTECTED]


 hi

 i'm trying to configure a TE205P on [EMAIL PROTECTED]

 i've edited /etc/sysconfig/zaptel adding this line:

 MODULES=$MODULES wct2xxp

 now, when the system is loading, i can see that the wct2xxp module is
 loaded correctly

 but if i try the command:
  /usr/local/sbin/genzaptelconf

 i get:

 STOPPING ASTERISK

 STOPPING FOP SERVER
 Generating  '/etc/zaptel.conf'
 Generating  '/etc/asterisk/zapata-auto.conf'


 STOPPING ASTERISK

 STOPPING FOP SERVER
 Unloading zaptel hardware drivers:.
 Removing zaptel module:  ERROR: Module zaptel is in use by wct4xxp
[FAILED]
 Loading zaptel framework:  [  OK  ]
 Waiting for zap to come online...OK
 Loading zaptel hardware modules: wct2xxpRunning ztcfg: [  OK  ]

 SETTING FILE PERMISSIONS
 Permissions OK

 STARTING ASTERISK
 Asterisk ended with exit status 1
 Asterisk died with code 1.
 Automatically restarting Asterisk.
 Asterisk ended with exit status 1
 Asterisk died with code 1.
 Automatically restarting Asterisk.

 -
 Asterisk could not start!
 Use 'tail /var/log/asterisk/full' to find out why.
 -
 Unable to connect to remote asterisk (does
 /var/run/asterisk/asterisk.ctl exist?)

 the file asterisk.ctl esists...
 [EMAIL PROTECTED] /]# ls -la /var/run/asterisk/asterisk.ctl
 srwxr-xr-x  1 asterisk asterisk 0 Feb 11 07:06
/var/run/asterisk/asterisk.ctl

 and this is what is reported in the logs:
 [EMAIL PROTECTED] /]# tail /var/log/asterisk/full
 Feb 11 07:06:05 VERBOSE[4808] logger.c:  [chan_zap.so]Feb 11 07:06:05
 VERBOSE[4808] logger.c:  [chan_zap.so] = (Zapata Telephony w/PRI)
 Feb 11 07:06:05 VERBOSE[4808] logger.c:   == Parsing
 '/etc/asterisk/zapata.conf': Feb 11 07:06:05 VERBOSE[4808] logger.c:
 == Parsing '/etc/asterisk/zapata.conf': Found
 Feb 11 07:06:05 VERBOSE[4808] logger.c:   == Parsing
 '/etc/asterisk/zapata-auto.conf': Feb 11 07:06:05 VERBOSE[4808]
 logger.c:   == Parsing '/etc/asterisk/zapata-auto.conf': Found
 Feb 11 07:06:05 VERBOSE[4808] logger.c:   == Parsing
 '/etc/asterisk/zapata_additional.conf': Feb 11 07:06:05 VERBOSE[4808]
 logger.c:   == Parsing '/etc/asterisk/zapata_additional.conf': Found
 Feb 11 07:06:05 WARNING[4808] chan_zap.c: Unable to specify channel 1:
 No such device or address
 Feb 11 07:06:05 ERROR[4808] chan_zap.c: Unable to open channel 1: No
 such device or address
 here = 0, tmp-channel = 1, channel = 1
 Feb 11 07:06:05 ERROR[4808] chan_zap.c: Unable to register channel '1-23'
 Feb 11 07:06:05 WARNING[4808] loader.c: chan_zap.so: load_module
 failed, returning -1
 Feb 11 07:06:05 WARNING[4808] loader.c: Loading module chan_zap.so failed!
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RE: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread chip

I can definitely vouch for Sangomas cards with hardware echo
cancel.  Ive been installing Asterisk boxes for about 6
months now using Digium TDM cards and Sipura SPA-3000s in
small installations.  This past month I installed in a small
office with 3 pots lines.  The echo was very bad and of
course the phone company (SBC) claims the lines pass all
tests.  I exhausted all the echo cancel combinations in
zaptel and still had echo and bad noise during double talk. 
I installed a Sangoma A200 with hardware echo cancel and you
would never know there was a problem. Its the best sounding
connection Ive heard through Asterisk.

Chip Schweiss


-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 10, 2006 10:35 PM
To: asterisk-users
Subject: Re: [Asterisk-Users] TE411P Really Bad Echo

It was Digium's opinion that perhaps the card had a VPM.  We
got a replacement TE411P, I implemented it tonight and still
the exact same echo problem.  At this point I feel like I can
rule out failed hardware.  

I contacted Digium support and now they are telling me it's
something with my carrier, and I should call them.  I called
Bellsouth, and they ran a full stress test on the circuit
taking me offline for about 30 minutes.  

The end result is that the circuit test passed with no
errors.  Bellsouth says it's not in their network, Digium
says its not their card, and I have a te411p with VPM
disabled in the wct4xx kernel module because something
doesn't work the way it should.  My customer is wanting to
know about sangoma cards with the echo cancellation, and at
this point I'm nervous to recommend any hardware.  I'm going
to look into the sangoma that you suggested.  Are there any
other kinds of products that I could look into either Passive
or Active.

Thanks 

Stagg Shelton
www.oneringnetworks.com


Matt wrote: 
try sangoma carrier grade 104d hardware EC card. we're using
it ourself.

Best Regards

Matt
- Original Message - 
From: Anthony Rodgers [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 07, 2006 12:57 PM
Subject: Re: [Asterisk-Users] TE411P Really Bad Echo


  
For what it's worth, we have been going through very similar
issues
with a TE411P - with Digium support, we have basically gone
as far as
we can with the HW EC, and are now using MG2 with much better
results.

We have a Ditech EC box on order.

Regards,
-- 
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Feb 7, 2006, at 7:36 AM, Matthew Fredrickson wrote:


On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote:

  
I just implemented a system using a TE411P hardware echo
cancellation
card. Per Digium, I setup zaptel.conf, and zapata.conf the
same way

as
  
I always have. To my surprise calls out to the PSTN had a
terrible
echo. 1 - 2 second delay, and quite clear. The echo was so
bad that

I
  
had to remove the hardware echo cancellation module from the
card.

We
  
are only using the 1st span of this card right now, and we have a
tdm400p with 4 fxs modules installed as well.

If anyone has experience with this card, can you tell me if I am
missing
something.


1 to 2 seconds?! That's ridiculously huge. I don't think
you'll find
a echo canceler anywhere that can fix your echo problem. If
it gets
better with the VPM disabled, then definitely contact Digium
tech-support about it.

Matthew Fredrickson

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Re: [Asterisk-Users] Sendmail with exchange

2006-02-11 Thread Michiel van Baak

On 13:44, Sat 11 Feb 06, Sean Cook wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 The only way you would need authenticated SMTP is for relaying.  My
 suggestion would be to not set up sendmail to use a smart host but have
 it act as an internet mail server.  It will lookup the mx records and
 make the sending determinations based on the domain it is sending to.

Actually this is only true when your ip is a static one that
you can list as provider ip.
A lot of blacklists put all the cable and dsl enduser ip's
somewhere under dynamic or domestic use
A lot of mailservers will block this.

Sorry for being totally unrelated to asterisk, but this has
been a big issue for several of my clients asterisk boxes.

 
 The exchange server should accept (with out authentication) anything
 that it is addressed to a locally hosted domain.

When it's internal this should work. Otherwise, see my point
above

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

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

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RE: [Asterisk-Users] Dialing part of the extension

2006-02-11 Thread Michael Collins

FYI,
If you want to learn more about why ${EXTEN:1} works, check out the
Asterisk TFOT book, chapters 4 and 5.  Page 95 of chapter 5 deals
specifically with the ${EXTEN} variable and the syntax of adding :1
(or :2, :3, etc.) - good stuff to know.

Check it out: http://www.speakup.nl/en/opensource/asterisktfot/

-MC


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ira
Sent: Saturday, February 11, 2006 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialing part of the extension

At 03:18 AM 02/11/2006, you wrote:
Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)

Ie: I want to dial all the XXX-es, but not the 9;

Exten = 9XX,1,Dial(Zap/4/${EXTEN:1},40,r)

Ira


-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.375 / Virus Database: 267.15.6/257 - Release Date:
02/10/2006


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RE: [Asterisk-Users] asterisk logger - urgent!!!

2006-02-11 Thread Michael Collins

Perhaps there's a happy medium: sprintf()?

I am curious to know if putting the output into a char array with
sprintf() (to preserve the output formatting) and then writing it with
write().  How much additional overhead would this take?  Hard to know
without trying it.

Is anyone in a position to write some test code that would do this and
report back the results?

-MC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Saturday, February 11, 2006 7:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk logger - urgent!!!

Bob Goddard wrote:

 Using fopen/fputs to ONLY append to a file, is quite frankly, stupid.
 Change it to open/write and you will be able to trap via the write
 return code and errno.

Patches to fix bugs are most welcome. Given that these files are written
using fprintf (because they are using format strings and long lists of
arguments), using write will require allocating memory, building the
output there and then calling write(). Seems like an awful lot of work
to avoid a simple system administration failure.
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Re: [Asterisk-Users] MOH broke with 1.2.4 .. ?


Tim Connolly wrote:

/etc/asterisk/musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s

-- Executing Answer(Zap/1-1, ) in new stack
-- Executing MusicOnHold(Zap/1-1, ) in new stack
-- Started music on hold, class 'default', on channel 'Zap/1-1'
-- Stopped music on hold on Zap/1-1
  

Try:

[default]
mode=mp3
directory=/var/lib/asterisk/mohmp3

[
--
Ben Franklin quote:

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


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Re: [Asterisk-Users] Sendmail with exchange

On Sat, Feb 11, 2006 at 12:30:51PM +0200, Tzafrir Cohen wrote:
 On Sat, Feb 11, 2006 at 08:29:31AM +, Peter Bowyer wrote:

  Install MSMTP as your local MTA (replacing sendmail). Configure
  Asterisk to use the local MTA, and configure MSMTP to forward to the
  Exchange server with authentication.
  
  http://msmtp.sourceforge.net/
 
 The problem with msmtp and similar programs (ssmtp, nullmailer) is that
 they don't queue. Thus if there was a temporary problem at the network
 or the recieving side, the message is lost.

But now when I think about it, why won't asterisk queue the mail? The
message itself is stored in the mailbox. So Asterisk only needs to
remember where it is stored.

Basically:

If the sendmail command returns an error, The voicemail app knowss it
need to be queued. So it remembers the path to the message and the
details of the message in a queue. 

Every once in a while there is an attempt to re-end messages in that
queue.

If someone checks the messages in the mailbox, any waiting messages
should be invalidated.


Anybody feels like trying to see if this is close to implementable?

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

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Re: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Matt

I vouch for Sangoma's too.

We use sangoma 104d EC card, echoes gone, works well. As to te411p, we have
not tried yet, we don't know.

Best Regards

Matt

- Original Message - 
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, February 11, 2006 2:53 PM
Subject: RE: [Asterisk-Users] TE411P Really Bad Echo


 I can definitely vouch for Sangoma's cards with hardware echo
 cancel.  I've been installing Asterisk boxes for about 6
 months now using Digium TDM cards and Sipura SPA-3000s in
 small installations.  This past month I installed in a small
 office with 3 pots lines.  The echo was very bad and of
 course the phone company (SBC) claims the lines pass all
 tests.  I exhausted all the echo cancel combinations in
 zaptel and still had echo and bad noise during double talk.
 I installed a Sangoma A200 with hardware echo cancel and you
 would never know there was a problem. It's the best sounding
 connection I've heard through Asterisk.

 Chip Schweiss


 -Original Message-
 From: asterisk [mailto:[EMAIL PROTECTED]
 Sent: Friday, February 10, 2006 10:35 PM
 To: asterisk-users
 Subject: Re: [Asterisk-Users] TE411P Really Bad Echo

 It was Digium's opinion that perhaps the card had a VPM.  We
 got a replacement TE411P, I implemented it tonight and still
 the exact same echo problem.  At this point I feel like I can
 rule out failed hardware.

 I contacted Digium support and now they are telling me it's
 something with my carrier, and I should call them.  I called
 Bellsouth, and they ran a full stress test on the circuit
 taking me offline for about 30 minutes.

 The end result is that the circuit test passed with no
 errors.  Bellsouth says it's not in their network, Digium
 says its not their card, and I have a te411p with VPM
 disabled in the wct4xx kernel module because something
 doesn't work the way it should.  My customer is wanting to
 know about sangoma cards with the echo cancellation, and at
 this point I'm nervous to recommend any hardware.  I'm going
 to look into the sangoma that you suggested.  Are there any
 other kinds of products that I could look into either Passive
 or Active.

 Thanks

 Stagg Shelton
 www.oneringnetworks.com


 Matt wrote:
 try sangoma carrier grade 104d hardware EC card. we're using
 it ourself.

 Best Regards

 Matt
 - Original Message - 
 From: Anthony Rodgers [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, February 07, 2006 12:57 PM
 Subject: Re: [Asterisk-Users] TE411P Really Bad Echo



 For what it's worth, we have been going through very similar
 issues
 with a TE411P - with Digium support, we have basically gone
 as far as
 we can with the HW EC, and are now using MG2 with much better
 results.

 We have a Ditech EC box on order.

 Regards,
 -- 
 Anthony Rodgers
 Business Systems Analyst
 District of North Vancouver
 Web: http://www.dnv.org
 RSS Feed: http://www.dnv.org/rss.asp


 On Feb 7, 2006, at 7:36 AM, Matthew Fredrickson wrote:


 On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote:


 I just implemented a system using a TE411P hardware echo
 cancellation
 card. Per Digium, I setup zaptel.conf, and zapata.conf the
 same way

 as

 I always have. To my surprise calls out to the PSTN had a
 terrible
 echo. 1 - 2 second delay, and quite clear. The echo was so
 bad that

 I

 had to remove the hardware echo cancellation module from the
 card.

 We

 are only using the 1st span of this card right now, and we have a
 tdm400p with 4 fxs modules installed as well.

 If anyone has experience with this card, can you tell me if I am
 missing
 something.


 1 to 2 seconds?! That's ridiculously huge. I don't think
 you'll find
 a echo canceler anywhere that can fix your echo problem. If
 it gets
 better with the VPM disabled, then definitely contact Digium
 tech-support about it.

 Matthew Fredrickson

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Re: [Asterisk-Users] Dialing part of the extension

On Sat, Feb 11, 2006 at 03:26:26PM -0800, Michael Collins wrote:
 FYI,
 If you want to learn more about why ${EXTEN:1} works, check out the
 Asterisk TFOT book, chapters 4 and 5.  Page 95 of chapter 5 deals
 specifically with the ${EXTEN} variable and the syntax of adding :1
 (or :2, :3, etc.) - good stuff to know.
 
 Check it out: http://www.speakup.nl/en/opensource/asterisktfot/

Or try http://www.voip-info.org/wiki/view/Asterisk+variables

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

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Re: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Michiel van Baak

On 16:48, Sat 11 Feb 06, Matt wrote:
 I vouch for Sangoma's too.
 
 We use sangoma 104d EC card, echoes gone, works well. 

I second that, the sangoma cards are awesome.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

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

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Re: [Asterisk-Users] asterisk logger - urgent!!!

2006-02-11 Thread Kevin P. Fleming

Michael Collins wrote:

 I am curious to know if putting the output into a char array with
 sprintf() (to preserve the output formatting) and then writing it with
 write().  How much additional overhead would this take?  Hard to know
 without trying it.

Please... if you don't have the skills to test things, don't make
suggestions like this.

This is _exactly_ what I was referring to earlier. How do you know what
size array is going to be large enough? Do you just allocate an enormous
one on the stack for every call to ast_log, or do you malloc() one
instead (which has serious performance issues)? You can't just 'guess'
what will be big enough, which then leads to running through the
argument list twice... There will be a non-zero cost for doing this.

Can you honestly say that _any_ of this performance penalty, no matter
how small, for every user of Asterisk on every system everywhere, is
worth the cost when the entire problem that caused this thread can be
avoided by just paying attention to your server? No log files, CDR files
or anything else in the discussion here can grow to 2GB in anything less
than a few days under normal circumstances, if not far, far longer than
that. If you can't be bothered to run logrotate or some sort of watching
process on your server, I don't think we should be force everyone else
to have lower performance just so you won't be inconvenienced :-)

(and by 'you' I don't mean you specifically, Michael, I am referring to
those in this thread who think we should change the code used to write
to every file we write to in Asterisk)
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[Asterisk-Users] Dell server

2006-02-11 Thread Paolo Supino


Hi

Not exactly a asterisk specific question and what more I'm a newby. I
apologize.
The story: I was given the task of transitioning my company's PBX (20
people) from a normal old digital PBX to something newer. I chose to use
Asterisk. For the project i was given a Dell 850 for the task. My
initial intention is to connect asterisk to the current PBX via a T1
connection between them (there is an unused T1 port on the current PBX)
and slowly transition extensions from the old PBX to asterisk. The
server has the following expansion slots: 1 64bit/133MHz PCI-X and 1 PCI
express x8 slot. By all means this isn't my final box  for asterisk and
final solution, just an interim solution to solve some bottlenecks that
we have with the current
The questions:
1. Has anyone used a Dell 850 for a small PBX?
2. Will the Digium single span T1 or Sangoma A101 work with these
expansion slots?



TIA
Paolo


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[Asterisk-Users] Problem with Wait() and chan_capi-cm?

2006-02-11 Thread Florian Heer


Hi!

I am playing around with Asterisk and have a problem :-)
(Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4)
I have a sip-phone at my desk and an ISDN-phone (independent of the 
Asterisk-server) in my living room, when I'm not at my desk, the 
sip-phone is switched off. I would like to be able to accept calls at 
both phones (when available) and have Voicemail kick in if I don't 
answer. The 'normal' extension would be something like this:


exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)

Works fine as long as the sip-phone is available, if it is not, it is 
flagged congested/busy, so the next extension would be 102, if I wanted 
VoiceMail to kick in in that case, this works:

exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)
exten = 12345,102,VoiceMail(su12345)

But that is not, what I had in mind, I would like to have 30 seconds to 
get to the phone, so in theory, this should do the trick:

exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)
exten = 12345,102,Wait(30)
exten = 12345,103,VoiceMail(su12345)

But Asterisk can not take over the line after the wait.

To test, if the Wait was the problem, I created this:
exten = 12345,1,Wait(10)
exten = 12345,2,Answer()
exten = 12345,3,Milliwatt()

And still: Asterisk can't take over the ISDN line. The console output says:
 == ISDN1: Incoming call '12345' - '12345'
   -- Executing Wait(CAPI/ISDN1/12345-19, 10) in new stack
   -- Executing Answer(CAPI/ISDN1/12345-19, ) in new stack
 == ISDN1: Answering for 12345
   -- Executing Milliwatt(CAPI/ISDN1/12345-19, ) in new stack
   CAPI INFO 0x34d1: Invalid call reference value
 == Spawn extension (capi-in, 12345, 3) exited non-zero on 
'CAPI/ISDN1/12345-19'

 == ISDN1: CAPI Hangingup

If I try that in a pure sip-context, it works as I thought it would.

Now: do I do something wrong? Is there a problem with the Wait() 
application? Or is that more likely a bug in chan_capi-cm?


Regards, Florian.
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RE: [Asterisk-Users] Problem with Wait() and chan_capi-cm?

2006-02-11 Thread gw

Try build 8015.  I know its odd, but this is just like the problem I am
having... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Heer
Sent: Saturday, February 11, 2006 9:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem with Wait() and chan_capi-cm?

Hi!

I am playing around with Asterisk and have a problem :-)
(Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a
sip-phone at my desk and an ISDN-phone (independent of the
Asterisk-server) in my living room, when I'm not at my desk, the
sip-phone is switched off. I would like to be able to accept calls at
both phones (when available) and have Voicemail kick in if I don't
answer. The 'normal' extension would be something like this:

exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)

Works fine as long as the sip-phone is available, if it is not, it is
flagged congested/busy, so the next extension would be 102, if I wanted
VoiceMail to kick in in that case, this works:
exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)
exten = 12345,102,VoiceMail(su12345)

But that is not, what I had in mind, I would like to have 30 seconds to
get to the phone, so in theory, this should do the trick:
exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)
exten = 12345,102,Wait(30)
exten = 12345,103,VoiceMail(su12345)

But Asterisk can not take over the line after the wait.

To test, if the Wait was the problem, I created this:
exten = 12345,1,Wait(10)
exten = 12345,2,Answer()
exten = 12345,3,Milliwatt()

And still: Asterisk can't take over the ISDN line. The console output
says:
  == ISDN1: Incoming call '12345' - '12345'
-- Executing Wait(CAPI/ISDN1/12345-19, 10) in new stack
-- Executing Answer(CAPI/ISDN1/12345-19, ) in new stack
  == ISDN1: Answering for 12345
-- Executing Milliwatt(CAPI/ISDN1/12345-19, ) in new stack
CAPI INFO 0x34d1: Invalid call reference value
  == Spawn extension (capi-in, 12345, 3) exited non-zero on
'CAPI/ISDN1/12345-19'
  == ISDN1: CAPI Hangingup

If I try that in a pure sip-context, it works as I thought it would.

Now: do I do something wrong? Is there a problem with the Wait()
application? Or is that more likely a bug in chan_capi-cm?

Regards, Florian.
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Re: [Asterisk-Users] Problem with Wait() and chan_capi-cm?

2006-02-11 Thread Florian Heer


[EMAIL PROTECTED] wrote:


Try build 8015.  I know its odd, but this is just like the problem I am
having... 
 


Uhm... sorry if I seem a bit uninformed, but how do I get that version?

Regards, Florian.
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[Asterisk-Users] bad sound frequency

2006-02-11 Thread Nitin Gupta

Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor.
Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency.
i.e a sentece which shoudl finish in 4 secs finishes in much lesser time. Where can be the problem? and configuration issue? 

Thanks,
Nitin

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Re: [Asterisk-Users] TE411P Really Bad Echo


On Sat, 11 Feb 2006, Rob Lith wrote:

TE406P/411P and if you need to go dedicated to hanlde all possible look at
an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO
CANCELLER (1U Version) ? $1295


Is the orion echo canceller a higher quality EC than tellabs?

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Re: [Asterisk-Users] TE411P Really Bad Echo

Could be, but I don't see why one would spend more just becuase the
answer is yes. The tellabs will do the job perfectly (at least in my
experience) and can be picked up for less than $100.00 on eBay. They
have proven to last for 20 years (the older models being sold on eBay
were manufactured in the 80s). I see no reason to spend the money.

On 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 On Sat, 11 Feb 2006, Rob Lith wrote:
  TE406P/411P and if you need to go dedicated to hanlde all possible look at
  an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO
  CANCELLER (1U Version) ± $1295

 Is the orion echo canceller a higher quality EC than tellabs?

 -Dan

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Re: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Steve Underwood

I don't know about the Tellabs cancellers in particular, but I think any 
echo canceller built in the 80s will be a fairly poor performer. Many 
improvements in EC occurred in the early 90s, in response to the 
problems of earlier cancellers. Also, most older cancellers only cancel 
fairly short echo tails, as the compute needed for longer cancellation 
was expensive.


On the other hand, they may be adequate for your needs, and are cheap.

Steve


C F wrote:


Could be, but I don't see why one would spend more just becuase the
answer is yes. The tellabs will do the job perfectly (at least in my
experience) and can be picked up for less than $100.00 on eBay. They
have proven to last for 20 years (the older models being sold on eBay
were manufactured in the 80s). I see no reason to spend the money.

On 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 


On Sat, 11 Feb 2006, Rob Lith wrote:
   


TE406P/411P and if you need to go dedicated to hanlde all possible look at
an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO
CANCELLER (1U Version) ± $1295
 


Is the orion echo canceller a higher quality EC than tellabs?

-Dan
   



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[Asterisk-Users] What to know for installing ARI

2006-02-11 Thread Zach A

Hi everybody,

I have an Asterisk box and I want to install just ARI on it for
monitoring the calls. Installing [EMAIL PROTECTED] utilizes too much resources 
and
memory and also takes away freedom of configuration asterisk. I like
using asterisk on its CLI. But just for recorded calls I need to use
ARI. What I need to do for that. As I understand I need to install
Apache and MySQL on the same machine. What else I need to do. Is there
any step by step guide about it, or can somebody help me on this?

Thanks,

Zach

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Re: [Asterisk-Users] TE411P Really Bad Echo

The most common Tellabs EC are not the ones from the 80s, I was just
pointing out that the quality was good, since they still last now 20
years later.

On 2/11/06, Steve Underwood [EMAIL PROTECTED] wrote:
 I don't know about the Tellabs cancellers in particular, but I think any
 echo canceller built in the 80s will be a fairly poor performer. Many
 improvements in EC occurred in the early 90s, in response to the
 problems of earlier cancellers. Also, most older cancellers only cancel
 fairly short echo tails, as the compute needed for longer cancellation
 was expensive.

 On the other hand, they may be adequate for your needs, and are cheap.

 Steve


 C F wrote:

 Could be, but I don't see why one would spend more just becuase the
 answer is yes. The tellabs will do the job perfectly (at least in my
 experience) and can be picked up for less than $100.00 on eBay. They
 have proven to last for 20 years (the older models being sold on eBay
 were manufactured in the 80s). I see no reason to spend the money.
 
 On 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 
 
 On Sat, 11 Feb 2006, Rob Lith wrote:
 
 
 TE406P/411P and if you need to go dedicated to hanlde all possible look at
 an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO
 CANCELLER (1U Version) ± $1295
 
 
 Is the orion echo canceller a higher quality EC than tellabs?
 
 -Dan
 
 

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Re: [Asterisk-Users] TE411P Really Bad Echo


On Sun, 12 Feb 2006, Steve Underwood wrote:
I don't know about the Tellabs cancellers in particular, but I think any echo 
canceller built in the 80s will be a fairly poor performer. Many improvements 
in EC occurred in the early 90s, in response to the problems of earlier 
cancellers. Also, most older cancellers only cancel fairly short echo tails, 
as the compute needed for longer cancellation was expensive.


On the other hand, they may be adequate for your needs, and are cheap.


the tellabs you find on ebay were not built in the 80s. they have 64ms and 
128ms echo tails.


-Dan
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RE: [Asterisk-Users] TE411P Really Bad Echo

2006-02-11 Thread Darren Wright

Eh.   Not for $1000 more, and I've got both in production.  Customer service 
was an issue.  

-Darren


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
 Sent: Saturday, February 11, 2006 10:19 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] TE411P Really Bad Echo
 
 On Sat, 11 Feb 2006, Rob Lith wrote:
  TE406P/411P and if you need to go dedicated to hanlde all possible look
 at
  an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO
  CANCELLER (1U Version) ± $1295
 
 Is the orion echo canceller a higher quality EC than tellabs?
 
 -Dan

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Re: [Asterisk-Users] TE411P Really Bad Echo

On 2/12/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 On Sun, 12 Feb 2006, Steve Underwood wrote:
  I don't know about the Tellabs cancellers in particular, but I think any 
  echo
  canceller built in the 80s will be a fairly poor performer. Many 
  improvements
  in EC occurred in the early 90s, in response to the problems of earlier
  cancellers. Also, most older cancellers only cancel fairly short echo tails,
  as the compute needed for longer cancellation was expensive.
 
  On the other hand, they may be adequate for your needs, and are cheap.

 the tellabs you find on ebay were not built in the 80s. they have 64ms and
 128ms echo tails.


How do you know which ones I'm talking about? read up:
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers
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Re: [Asterisk-Users] TE411P Really Bad Echo