Re: [Asterisk-Users] ASTCC not billing

[Juan Luis Moyano]

On Sab, 25 de Junio de 2005, 5:20 pm, Darren Wiebe dijo:
 Do you have the notransfer and reinvite lines set properly?  I had this
 same problem with ASTCC but found that if I removed asterisk including
 the source and did a clean reinstall it worked suddenly.

 Darren

Darren, how is the proper way of setting notransfer and canreinvite lines
on IAX. TIA.

-- 
Juan Luis Moyano
[EMAIL PROTECTED]

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[Asterisk-Users] LiveVoip is Bankrupt

[Andres]

So it looks like Livevoip went Bankrupt 



---

There is a Federal Court Order in place and has been since Friday early a.m. 
ALL Suppliers are now under a Court Order that prevents them from terminating 
any and all services to LiveVoip LLC. If they take such any action they will be 
in direct
violation of a U.S. Federal Court Order. If you have any questions you may 
contact our lawyer - Customers and Creditors are now under a U.S. Court Ordered 
Stay NOT to have any contact with LiveVoip LLC Management.

LiveVoip LLC has Ceased Operations and Filed for Bankruptcy. This action was 
taken after the company was unable to resolve issues with carriers over 
billing, mass credit card fraud, suppliers not delivering on what they had been 
paid for among other things.
A Stay Order is in effect at this time and all questions must be directed to 
our company lawyer. Creditors will be hearing from the Courts in due course.

LiveVoip LLC is no Closed.

United States Federal Bankruptcy Court District Montana
Case: 05-62057 LiveVoip LLC

Company Lawyer: Robert Kampfer Esq. 406.727.954 
The LiveVoip network is offline. An Update will be issued on our main website. The trouble ticket server is also having its own problems. Please watch our main for site for complete details.



LiveVoip LLC
---


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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Yair Hakak]

well, i can't say i'm surprised. any company whose approach to
customers is you are all scum trying to cheat us, don't ask
questions, and we'll help you when we feel like it isn't going to be
around for a long time.



On 6/26/05, Andres [EMAIL PROTECTED] wrote:
 So it looks like Livevoip went Bankrupt
 
 
 ---
 
 There is a Federal Court Order in place and has been since Friday early a.m. 
 ALL Suppliers are now under a Court Order that prevents them from terminating 
 any and all services to LiveVoip LLC. If they take such any action they will 
 be in direct
 violation of a U.S. Federal Court Order. If you have any questions you may 
 contact our lawyer - Customers and Creditors are now under a U.S. Court 
 Ordered Stay NOT to have any contact with LiveVoip LLC Management.
 
 LiveVoip LLC has Ceased Operations and Filed for Bankruptcy. This action was 
 taken after the company was unable to resolve issues with carriers over 
 billing, mass credit card fraud, suppliers not delivering on what they had 
 been paid for among other things.
 A Stay Order is in effect at this time and all questions must be directed to 
 our company lawyer. Creditors will be hearing from the Courts in due course.
 
 LiveVoip LLC is no Closed.
 
 United States Federal Bankruptcy Court District Montana
 Case: 05-62057 LiveVoip LLC
 
 Company Lawyer: Robert Kampfer Esq. 406.727.954
 The LiveVoip network is offline. An Update will be issued on our main 
 website. The trouble ticket server is also having its own problems. Please 
 watch our main for site for complete details.
 
 
 LiveVoip LLC
 ---
 
 
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Re: [Asterisk-Users] Livevoip

[Luki]

See other thread about Bankruptcy. Interestingly enough, termination
still works flawlessly... and better than ever before :). And, get
this, they don't charge mobile rates to mobile phones in Germany... go
figure...

--Luki

On 6/25/05, Moody [EMAIL PROTECTED] wrote:
 I have a UK Livevoip DID that is down, and has been for several days.
 
 I'm looking to replace my London DID, low usage but need at least 2
 channels and a local London number.
 
 Please email me off list if you can provide this.
 
 J
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Brian Capouch]

Yair Hakak wrote:

well, i can't say i'm surprised. any company whose approach to
customers is you are all scum trying to cheat us, don't ask
questions, and we'll help you when we feel like it isn't going to be
around for a long time.



I agree totally.  After seeing some of the issues people were having 
with their customer support (or better, flying off the handle at their 
customers) I decided to stay clear of them.


Survival of the fittest . . .

B.
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Re: [Asterisk-Users] Livevoip

[Obelix]

Quoting Darren Wiebe [EMAIL PROTECTED]:

They never truly got their act together. I remember checking my CDR and
realising that they were charging my 0800 numbers in 1/100 of a cent instead of
cents. It is a pity their DTMF tones were not working for me. At least I would
have gained something from the payments I made to them for those numbers.

I don't  think they were ever a technically sound operation.


 Is there anybody else here that still has anything with Livevoip?  They
 are down and apparently have no idea when they will be back up.  Has
 anybody talked to them?  I wouldn't care at all if it was not that I
 have 2 DIDs that I've been unable to transfer away. :-(

 Darren Wiebe
 [EMAIL PROTECTED]
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Re: [Asterisk-Users] Looking for link.exe to compile G729 codec

[Obelix]

Quoting Tony Hoyle [EMAIL PROTECTED]:

I think I installed the framework some time ago. I will hunt for the install
location and see if I will find it.

Thanks


 Obelix wrote:
  I want to compile the G729 codec to try it out with firefly.
  I don't have Visual C++ 6 compiler. Is there a way I can obtain the
 link.exe
  alone for use with cygwin, or a substitute program?
 
  I don't look forward to installing the whole Visual C++ just for the
 link.exe
 
 The .net framework SDK apparently has it... just ignore all the .net
 bits and use nmake/cl/link from it.

 Tony
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[Asterisk-Users] handle wrong extensions in Dialplam

[Mahmoud Badran]

Hello;
i am trying to make a dial plan that can handle any wrong extensions dialled from the local sip phone for example so that if i dialled the right extension it rings but if i dialled wrong or existing extension it redirect him to the Main menu for example?

thanks in advance


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Re: [Asterisk-Users] * fax reliability between ISDN PRI and FXS ports

[[EMAIL PROTECTED]]

I plug my fax machine into a Grandstream ATA286 (No T.38).  Works fine for me 
and has done for the last year.  I can send and receive 40 page faxes without 
issue.  The ATA286 is located on the same LAN as the Asterisk. 

Craig


 Andres Maduro [EMAIL PROTECTED] wrote: 
 Hi, 
 
 
 
I am building for a customer an * solution that will use 2 Digium cards.
 
 
 
1 x TE110P (T1 ISDN PRI)
 
1 x TDM40B (4 analog ports, 2 for faxes, 2 for extensions)
 
  
 
The system will be connected to the PSTN through the T1 ISDN PRI interface.
 
All customer extensions will be SIP phones except 2 fax machines that will
 
use the analog FXS ports on the TDM40B card.
 
 
 
I have been investigating on this list and found that faxing is not reliable
 
between Zaptel cards and that Digium does not support nor recommend fax over
 
the TDMXXB interfaces.
 
 
 
Is this true ?  Will fax not be reliable enough for my customer ?
 

 
Will it be better to use Audiocodes digital and analog gateway (ATA) with
 
T.38 capabilities ?  
 
 
 
Thanks in advance, 
 
Andres Maduro
 
 
 


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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Bashir Ullah - www.Lamsre.Com]

I found a large IAX supported provider beside Voipjet. Now ...?

Bashir

I still i have good balance with them, I dont know what will be happend.
and my canadian DID .
- Original Message - 
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, June 26, 2005 1:09 AM
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt


 Yair Hakak wrote:
  well, i can't say i'm surprised. any company whose approach to
  customers is you are all scum trying to cheat us, don't ask
  questions, and we'll help you when we feel like it isn't going to be
  around for a long time.
 

 I agree totally.  After seeing some of the issues people were having
 with their customer support (or better, flying off the handle at their
 customers) I decided to stay clear of them.

 Survival of the fittest . . .

 B.
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Re: [Asterisk-Users] chan_capi-cm-0.5.1 and CLIR

[asterisk_on_oelf]

Thanks Armin. This was a good hint. In this part of the debug you can see the
callerID provided from my internal phone. You are right, that the 
included name

was the problem.
Now I added a line SetCallerID(12345) before Dial(CAPI...) to my 
extensions.conf

and everything works fine.
(12345 has to be a valid MSN, or you have the same problem as before)

Quoting Armin Schindler [EMAIL PROTECTED]:

CallingPartyNumber  = 00 a0 22Jens22 2c 3c123453e



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Re: [Asterisk-Users] handle wrong extensions in Dialplam

[C. Hatton Humphrey]

  i am trying to make a dial plan that can handle any wrong extensions
 dialled from the local sip phone for example so that if i dialled the right
 extension it rings but if i dialled wrong or existing extension it redirect
 him to the Main menu for example?

You might see if you can put the i extension to work for you in your
local dialplan.

Hatton
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[Asterisk-Users] cdr and billing

[Mahmoud Badran]

Hello ;
how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled


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Re: [Asterisk-Users] Livevoip

[Rich Adamson]

 Is there anybody else here that still has anything with Livevoip?  They 
 are down and apparently have no idea when they will be back up.  Has 
 anybody talked to them?  I wouldn't care at all if it was not that I 
 have 2 DIDs that I've been unable to transfer away. :-(

They are still down and won't be back up anytime soon. Their web site
says:

LiveVoip LLC has Ceased Operations and Filed for Bankruptcy. This 
action was taken after the company was unable to resolve issues with 
carriers over billing, mass credit card fraud, suppliers not 
delivering on what they had been paid for among other things.


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[Asterisk-Users] Asterisk RealTime Voicemail

[harry gaillac]

Hello,

I read 
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail
however when i run show voicemail users app voicemail
return users in voicemail.conf

Why?

Harry






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RE: [Asterisk-Users] cdr and billing

[jurczak]

Why dont you try nocdr



-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mahmoud Badran
Sent: Sunday, June 26, 2005 2:32
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] cdr and
billing



Hello ;
how can i enable billing only while using specific trunk (ex:zap) but internal
sip calls will not be counted specifically how to make all outbound is counted
i am using asterisk mysql cdr enabled 






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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Rich Adamson]

I've had pretty good luck with www.teliax.com



 I found a large IAX supported provider beside Voipjet. Now ...?
 
 Bashir
 
 I still i have good balance with them, I dont know what will be happend.
 and my canadian DID .
 - Original Message - 
 From: Brian Capouch [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Sunday, June 26, 2005 1:09 AM
 Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt
 
 
  Yair Hakak wrote:
   well, i can't say i'm surprised. any company whose approach to
   customers is you are all scum trying to cheat us, don't ask
   questions, and we'll help you when we feel like it isn't going to be
   around for a long time.
  
 
  I agree totally.  After seeing some of the issues people were having
  with their customer support (or better, flying off the handle at their
  customers) I decided to stay clear of them.
 
  Survival of the fittest . . .
 
  B.
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---End of Original Message-


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Re: [Asterisk-Users] * fax reliability between ISDN PRI and FXS ports

[Rich Adamson]

 I am building for a customer an * solution that will use 2 Digium cards.
  
 1 x TE110P (T1 ISDN PRI)
 1 x TDM40B (4 analog ports, 2 for faxes, 2 for extensions)
   
 The system will be connected to the PSTN through the T1 ISDN PRI interface.  
 All customer 
extensions will be SIP phones except 2 fax
 machines that will use the analog FXS ports on the TDM40B card.
  
 I have been investigating on this list and found that faxing is not reliable 
 between Zaptel 
cards and that Digium does not support nor
 recommend fax over the TDMXXB interfaces.
  
 Is this true ?  Will fax not be reliable enough for my customer ?

True... the TDM card (or its drivers) has an issue with missed frames
that seriously impacts its ability to handle _any_ modem-type calls.

 Will it be better to use Audiocodes digital and analog gateway (ATA) 
 with T.38 capabilities ? 

A good ATA should work fine, but you should probably test it before
exposing your customer to potential problems. Others have used ATA's
with analog fax machines.


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RE: [Asterisk-Users] cdr and billing

[Mahmoud Badran]

thanks alot for help but problem is; consider this scenario an internal sip phone calls the IVR which shouldnt be billed then he dial an extension from the ivr that redirects him to outbound line that makes the call have some time counting in the ivr and other time counting during the outbound call so how can i bill him on the outbound only??





On Sun, 2005-06-26 at 15:23 +0300, jurczak wrote:


Well, in your dialplan, in the place where you are calling SIP (internal phones) you should put nocdr in the first priority

So if you would have



Exten = _4.,1,NoCdr

Exten = _4.,2,Dial(SIP/${EXTEN})



Assuming that your SIP begins with 4.

With this you wont have any CDR for your internal calls.



-Original Message-
From: Mahmoud Badran [mailto:[EMAIL PROTECTED] 
Sent: Sunday, June 26, 2005 3:17 PM
To: jurczak
Subject: RE: [Asterisk-Users] cdr and billing



come on!!
whats wrong with ya?


On Sun, 2005-06-26 at 15:01 +0300, jurczak wrote:



Why dont you try nocdr



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mahmoud Badran
Sent: Sunday, June 26, 2005 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] cdr and billing



Hello ;
how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled 







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[Asterisk-Users] Bug in Mailman version 2.1.5

[Khubeka JM]

Bug in Mailman version 2.1.5

We're sorry, we hit a bug!

If you would like to help us identify the problem,
please email a copy of this page to the webmaster for
this site with a description of what happened. Thanks!
Traceback:

Traceback (most recent call last):
  File /var/mailman/scripts/driver, line 87, in
run_main
main()
  File /var/mailman/Mailman/Cgi/confirm.py, line
116, in main
subscription_confirm(mlist, doc, cookie, cgidata)
  File /var/mailman/Mailman/Cgi/confirm.py, line
345, in subscription_confirm
userdesc = mlist.pend_confirm(cookie,
expunge=False)[1]
TypeError: unsubscriptable object



Python information:

VariableValue
sys.version 2.3.5 (#1, May 25 2005, 15:49:15) [GCC
3.4.3 20041125 (Gentoo Linux 3.4.3-r1, ssp-3.4.3-0,
pie-8.7.7)]
sys.executable  /usr/bin/python
sys.prefix  /usr
sys.exec_prefix /usr
sys.path/usr
sys.platformlinux2

Environment variables:

VariableValue
HTTP_REFERER 
http://lists.digium.com/mailman/confirm/asterisk-users/f68f025a805c6fd7a439027c9b780828bcc5d5a7
SERVER_SOFTWARE Apache
SCRIPT_NAME /mailman/confirm
SERVER_SIGNATURE
REQUEST_METHOD  POST
PATH_INFO   /asterisk-users
SERVER_PROTOCOL HTTP/1.1
QUERY_STRING
CONTENT_LENGTH  137
HTTP_ACCEPT_CHARSET ISO-8859-1,utf-8;q=0.7,*;q=0.7
HTTP_USER_AGENT Mozilla/5.0 (X11; U; Linux i686;
en-US; rv:1.7.8) Gecko/20050511 Firefox/1.0.4
HTTP_CONNECTION keep-alive
SERVER_NAME lists.digium.com
REMOTE_ADDR 198.54.202.226
HTTP_VIA1.1 ndf-cache2 (NetCache NetApp/5.5R6D36),
1.1 rba-cache1 (NetCache NetApp/5.5R6D27)
PATH_TRANSLATED /var/mailman/html/asterisk-users
SERVER_PORT 80
SERVER_ADDR 69.16.138.164
DOCUMENT_ROOT   /var/mailman/html
PYTHONPATH  /var/mailman
SCRIPT_FILENAME /var/mailman/cgi-bin/confirm
SERVER_ADMIN[EMAIL PROTECTED]
HTTP_HOST   lists.digium.com
MAIL_CONFIG /etc/postfix2
REQUEST_URI /mailman/confirm/asterisk-users
HTTP_ACCEPT 
text/xml,application/xml,application/xhtml+xml,text/html;q=0.9,text/plain;q=0.8,image/png,*/*;q=0.5
GATEWAY_INTERFACE   CGI/1.1
HTTP_X_FORWARDED_FOR165.146.60.222, 196.25.253.14
REMOTE_PORT 49973
HTTP_ACCEPT_LANGUAGEen-us,en;q=0.5
CONTENT_TYPEapplication/x-www-form-urlencoded
HTTP_ACCEPT_ENCODINGgzip,deflate
UNIQUE_ID   Ln22x0UQiqQAAAd3QfEl 

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Re: [Asterisk-Users] cdr and billing

[Emanuele Pucciarelli]

Mahmoud Badran wrote:
 thanks alot for help but problem is; consider this scenario an internal
 sip phone calls the IVR which shouldnt be billed then he dial an
 extension from the ivr that redirects him to outbound line that makes
 the call have some time counting in the ivr and other time counting
 during the outbound call so how can i bill him on the outbound only??

ForkCDR and ResetCDR will probably help you!

-- 
Emanuele
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RE: [Asterisk-Users] Asterisk and Cisco CallManager Integration

[Walid Azab]

I have previously tried the  Asterisk/OH323/PWLIB/GNUGK combination and had
problems compiling OH323. I will try again from a clean installation. On the
other hand, can you send me any useful links or guides that you already
used. This can make our trial and error efforts much less.

Walid

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Oliver
Sent: Sunday, June 26, 2005 2:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration

We have successfully connect * .9x  1.0.x with CCM 3.3.x and up using both
gatekeeper and no gatekeeper..  Using SIP usually with CCM 4.0 and up..
With CCM 3.3.x, there is a limitation where the gateway H323 in your case
cannot use IP addresses, so the Asterisk box has to have correct DNS entries
to resolbve your asterisk ox..  Then just use regular route patterns and
direct it to asterisk..

That works well.  You may also want to make sure your compatibility matrix
between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities cause more
issues than I care to talk about.  The GNUGk web site has the best matrix to
follow..

Thanks,

GReg



On Sat, 2005-06-25 at 10:39 -0500, [EMAIL PROTECTED] wrote:
 Use a gatekeeper and have both boxes register with the gatekeeper.  
 That way you can specify what numbers go where.  From everything I 
 have tested, * will NOT register with CCM.  When I added in a 
 gatekeeper and had both sides register with it, everything works.
 
 Walid Azab wrote:
  Hello,
   
  I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED] latest version. I have earlier tried getting 
  Asterisk to register with CCM via H323 and failed. Back then, I 
  learned that this is a known bug in Asterisk. Also people who tried 
  doing that had also succeeded in getting calls to go through only 
  one direction like from CCM to Asterisk. I am not that expert so excuse
my ignorance with this subject.
  So please if anyone has any useful information or is sure that this 
  can now work please send me whatever you have on that.
   
  I simply want Asterisk users to get their dial tones through CCM.
   
  Thanks and I appreciate your assistance.
   
  Walid
   
   
  
  
  
  
  
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Re: [Asterisk-Users] * 1.0.8: no more reacting to callerid?

[Michael J. Tubby G8TIC]

I've put in the patch by hand, thus:

   if (start  (end = 
strrchr(appl, ')'))) {
   *start = 
*end = '\0';
   data = start 
+ 1;
   process_quotes_and_slashes(data, 
',', '|');
   } else if 
(stringp!=NULL  *stringp=='') {

   stringp++;
   data = 
strsep(stringp, \);

   stringp++;
   } else {
   if (stringp)
   data 
= strsep(stringp, ,);

   else
   data 
= ;

   }
#if 0

   pbx_substitute_variables_helper(NULL, 
ext, realext, sizeof(realext)-1);
   cidmatch = 
strchr(ext, '/');

   if (cidmatch) {
   *cidmatch = 
'\0';

   cidmatch++;
   }
   stringp=ext;
   strsep(stringp, 
/);

#endif

#if 1
   pbx_substitute_variables_helper(NULL, 
ext, realext, sizeof(realext)-1);

   stringp = realext;
   ext = 
strsep(stringp, /);

   cidmatch = stringp;
#endif

   if (!data)
   data=;
   while(*appl  
(*appl  33)) appl++;

   if (ipri) {
   if 
(!strcmp(realext, _.))
   ast_log(LOG_WARNING, 
The use of '_.' for an extension is strongly discouraged and c
   if 
(ast_add_extension2(con, 0, realext, ipri, cidmatch, appl, strdup(data), 
FREE, registrar)
   ast_log(LOG_WARNING, 
Unable to register extension at line %d\n, v-lineno);

   }
   }
   free(tc);



which is what I think you intended and it still doesn't work for me (yes, I 
did stop and restart Asterisk)...  I'm in the UK using a cheap X100P clone 
and V.23 Caller ID which used to work 100% under 1.0.7 in my extensions.conf 
I have a context, thus:


;
; from-pstn : incoming calls from the FXO card from PSTN
;
[from-pstn]
exten = s,1,Answer
exten = s,2,NoOp(CallerIDnum=${CALLERIDNUM} CallerID=${CALLERID})
exten = s/0,3,Goto(no-callerid,s,1)

; to dedicated lines on 7960s and the 7912s
exten = 
s,3,Dial(SIP/9001SIP/9002SIP/2003SIP/2004SIP/2005SIP/2006IAX2/thorcom/8102001,20,rt)

exten = s,4,Voicemail(u2001)
exten = s,5,HangUp

; okay, they withheld their caller id - play out a we dont do withheld 
callers and dump them to voicemail

[no-callerid]
exten = s,1,Playback(withheld-callerid)
exten = s,2,Voicemail(su2001)
exten = s,3,Hangup


Now when I get a call from a withheld this happens:

Connected to Asterisk 1.0.8 currently running on gate (pid = 18645)
Verbosity is at least 3
   -- Remote UNIX connection
   -- Starting simple switch on 'Zap/1-1'
   -- Executing Answer(Zap/1-1, ) in new stack
   -- Executing NoOp(Zap/1-1, CallerIDnum=0 CallerID=Number Witheld 
0) in new stack
   -- Executing Dial(Zap/1-1, 
SIP/9001SIP/9002SIP/2004SIP/2005SIP/2006IAX2/thorcom/8102001|20|rt) 
in new stack

   -- Called 9001
   -- Called 9002
   -- Called ... etc. etc.

and all my phones ring :o(


Mike



- Original Message - 
From: Kevin P. Fleming [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Saturday, June 25, 2005 7:33 PM
Subject: Re: [Asterisk-Users] * 1.0.8: no more reacting to callerid?



Daryl Jones wrote:

It's not just you. 

[Asterisk-Users] Building DUNDi?

[Bret Wortman]

I just built asterisk (and -addons, zaptel, libpri and -sounds) from the 
latest CVS but pbx_dundi.so didn't get built.  I grepped the source tree 
as checked out and I couldn't find any reference to dundi anywhere!


What did I miss?

Thanks!


Bret Wortman

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[Asterisk-Users] help regarding h323.conf

[SYED ADEEL ALI]

i m using several IAX  SIP softphones Now I've got an IP phone(Netphone) that supports H.323 protocol plz tell me how should i configure it to work with asterisk i m comfortable with sip.conf  iax.conf but what should i do to use h323.conf??? do i ned to install something or should i just copy h323.conf(the sample file) to /etc/asterisk directory ???FREE pop-up blocking with the new MSN Toolbar MSN Toolbar Get it now!

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[Asterisk-Users] failure in writing in pattern (routes)

[wassim darwish]

hi,i tried to write in pattern in routes to usa
destination 1* but i want to specify the number of
digits so i tried 1NXXNXX but it dose'nt worked so
please help me.  

__
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Re: [Asterisk-Users] * 1.0.8: no more reacting to callerid?

[Kevin P. Fleming]

Michael J. Tubby G8TIC wrote:

which is what I think you intended and it still doesn't work for me 
(yes, I did stop and restart Asterisk)...  I'm in the UK using a cheap 
X100P clone and V.23 Caller ID which used to work 100% under 1.0.7 in my 
extensions.conf I have a context, thus:


Yes, it does appear as though you applied the patch adequately... and we 
have reports from other users that it did fix the problem for them.

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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Mark Musone]

funny thing is just like their previous attitude with them blaming all
their customers, now they are blaming their bankruptcy on their
suppliers, clecs, and credit card fraud...

i hope one day they wake up and look in the mirror and see who the
real problem is...


p.s. i _love_ teliax. been using them for about 3 months now, after
canceling my broadvoice account for over a year and a half..i'll never
go back..


-Mark




On 6/26/05, Rich Adamson [EMAIL PROTECTED] wrote:
 I've had pretty good luck with www.teliax.com
 
 
 
  I found a large IAX supported provider beside Voipjet. Now ...?
 
  Bashir
 
  I still i have good balance with them, I dont know what will be happend.
  and my canadian DID .
  - Original Message -
  From: Brian Capouch [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Sunday, June 26, 2005 1:09 AM
  Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt
 
 
   Yair Hakak wrote:
well, i can't say i'm surprised. any company whose approach to
customers is you are all scum trying to cheat us, don't ask
questions, and we'll help you when we feel like it isn't going to be
around for a long time.
   
  
   I agree totally.  After seeing some of the issues people were having
   with their customer support (or better, flying off the handle at their
   customers) I decided to stay clear of them.
  
   Survival of the fittest . . .
  
   B.
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 ---End of Original Message-
 
 
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Re: [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement

[Armin Schindler]

On Sat, 25 Jun 2005, Stefan Gofferje wrote:
 Armin Schindler schrieb:
 
  I have added busy()/congestion() support to CVS HEAD now, can you please
  
  test if it works for you?
  
  
  
 Works perfectly well! Also CallingPres(32) does work! The only thing I wonder
 about is a delay. 

I know there is a delay. Currently the calling party is 'alerted' every time 
by default and this is not correct for calls which shall not be accepted.

 Is CLIP no screening (submitting a user provided number) supported by
 chan_capi_cm?

No, not yet.

Armin
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Re: [Asterisk-Users] Livevoip

[Darren Wiebe]

Well, I see they just posted a bankruptcy notice.  H.

Darren Wiebe
[EMAIL PROTECTED]

Moody wrote:


I have a UK Livevoip DID that is down, and has been for several days.

I'm looking to replace my London DID, low usage but need at least 2
channels and a local London number.

Please email me off list if you can provide this. 


J


On 6/26/05, Darren Wiebe [EMAIL PROTECTED] wrote:
 


Is there anybody else here that still has anything with Livevoip?  They
are down and apparently have no idea when they will be back up.  Has
anybody talked to them?  I wouldn't care at all if it was not that I
have 2 DIDs that I've been unable to transfer away. :-(

Darren Wiebe
[EMAIL PROTECTED]
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Darren Wiebe]

Yeah, I chuckled a little bit when I read the notice.  It has absolutely 
nothing to do with any stupid things they might have done, customers 
they chased away etc.  At least they only had $30.00 of mine. :-)


Darren

Mark Musone wrote:


funny thing is just like their previous attitude with them blaming all
their customers, now they are blaming their bankruptcy on their
suppliers, clecs, and credit card fraud...

i hope one day they wake up and look in the mirror and see who the
real problem is...


p.s. i _love_ teliax. been using them for about 3 months now, after
canceling my broadvoice account for over a year and a half..i'll never
go back..


-Mark




On 6/26/05, Rich Adamson [EMAIL PROTECTED] wrote:
 


I've had pretty good luck with www.teliax.com



   


I found a large IAX supported provider beside Voipjet. Now ...?

Bashir

I still i have good balance with them, I dont know what will be happend.
and my canadian DID .
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, June 26, 2005 1:09 AM
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt


 


Yair Hakak wrote:
   


well, i can't say i'm surprised. any company whose approach to
customers is you are all scum trying to cheat us, don't ask
questions, and we'll help you when we feel like it isn't going to be
around for a long time.

 


I agree totally.  After seeing some of the issues people were having
with their customer support (or better, flying off the handle at their
customers) I decided to stay clear of them.

Survival of the fittest . . .

B.
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---End of Original Message-


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Re: [Asterisk-Users] ASTCC not billing

[Darren Wiebe]

On IAX you set notransfer=yes and on SIP you set canreinvite=no

Darren

Juan Luis Moyano wrote:


On Sab, 25 de Junio de 2005, 5:20 pm, Darren Wiebe dijo:
 


Do you have the notransfer and reinvite lines set properly?  I had this
same problem with ASTCC but found that if I removed asterisk including
the source and did a clean reinstall it worked suddenly.

Darren

   


Darren, how is the proper way of setting notransfer and canreinvite lines
on IAX. TIA.

 



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RE: [Asterisk-Users] Livevoip 800 Choppy Audio

[John Kington]

Wiley, thanks for pointing me to NuFone for tollfree DID. I was planning to 
report

on results between LiveVOIP and NuFone. The apparent bankruptsy of LiveVOIP
means that my choppy audio will probably never be resolved. I set up both 
DID to
go through DISA and I could then use the echo test application. Everytime I 
tested

LiveVOIP, the audio was choppy. I have not experienced any choppiness with
NuFone but the echo seemed to take longer to get back to me compared to
LiveVOIP.
I now get a message that my call can not be completed when I call the LiveVOIP
DID and I see that I can not register my asterisk to them. I am glad I did 
not have

big dollars invested in them.
Regards,
John


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Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement

[Stefan Gofferje]

Armin Schindler schrieb:


On Sat, 25 Jun 2005, Stefan Gofferje wrote:
 


Armin Schindler schrieb:

   


I have added busy()/congestion() support to CVS HEAD now, can you please

test if it works for you?



 


Works perfectly well! Also CallingPres(32) does work! The only thing I wonder
about is a delay. 
   



I know there is a delay. Currently the calling party is 'alerted' every time 
by default and this is not correct for calls which shall not be accepted.
 


I noted a number of bugs and a feature-request at SF :-).

Regards,
Stefan

--
(o_   Stefan Gofferje  | Linux Systems Specialist
//\   Reg'd Linux User #247167 | Network Security Specialist
V_/_  Heckler  Koch - the original point and click interface

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Re: [Asterisk-Users] Looking for link.exe to compile G729 codec

[Tony Hoyle]

Obelix wrote:

Quoting Tony Hoyle [EMAIL PROTECTED]:

I think I installed the framework some time ago. I will hunt for the install
location and see if I will find it.


The framework is not the same as the framework SDK.

http://www.microsoft.com/downloads/details.aspx?FamilyId=9B3A2CA6-3647-4070-9F41-A333C6B9181Ddisplaylang=en

Tony
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RE: [Asterisk-Users] 2 servers via PRI

[Michael Di Martino]

-Original Message-
From: Altus Snyman [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 16, 2005 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 2 servers via PRI

Good day all
How do i set a connection between 2 asterisk servers via PRI
In Bri I would set one to NT and TE
How shoud the zapata.conf and zaptel.conf look
And how should the cable be?
All I got on the web was to set one to pri_net...this cant be all?
And the cable 
 pin1 -- pin4 pin2 -- pin5 pin3 -- pin6 pin4 -- pin1 pin5
-- pin2 pin6 -- pin3 pin5 -- pin8 pin8 -- pin7
Please Help and advice
Thanks Altus

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I have one asterisk server connected to 2 Norstar mics systems via two
PRI lines. Here is how I did it

My Zaptel.conf
loadzone = us
defaultzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no

#
# PRI's
#
span=1,0,0,esf,b8zs
#clear=1-24
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
#clear=25-48
bchan=25-47
dchan=48

my Zapata.conf

[channels]
usecallerid=yes
hidecallerid=no
usecallingpres=yes
callerid=asreceived
echocancel = yes   ; You can set this to 16, 32, 64, or 128, or 256
tweak to your needs.  Try 64.  Yes=128.
echotraining = 400 ; Ast trains to the beginning of the call, num is in
millisec.  0-4000.  Try 800.
echocancelwhenbridged = yes

context = internal
switchtype = dms100
signalling = pri_net
group = 1
channel = 1-23

context = internal
switchtype = dms100
signalling = pri_net
group = 2
channel = 25-47

my extensions.conf  (partial)

;Allows access to 2000 3000 Nortel extensions
exten = _2XXX,1,Dial(${TELX-MICS2}/${EXTEN:${TELX-MICS2-MSD}})
exten = _2XXX,2,Congestion
exten = _3XXX,1,Dial(${TELX-MICS1}/${EXTEN:${TELX-MICS1-MSD}})
exten = _3XXX,2,Congestion

I hope this help[s

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Re: [Asterisk-Users] Asterisk RealTime Voicemail

[Matthew Boehm]

 Hello,
 
 I read 
 http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail
 however when i run show voicemail users app voicemail
 return users in voicemail.conf
 
 Why?
 
 Harry

Because the function handle_show_voicemail_users does not query the realtime
database.  show voicemail users will only return those statically
configured in your voicemail.conf.

What you are seeing is expected behavior. I have 0 entries in my
voicemail.conf and over 75 in RealTime and I get no results when I give that
command.

-Matthew


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[Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!

[Khubeka JM]

Note: forwarded message attached.




__ 
Discover Yahoo! 
Have fun online with music videos, cool games, IM and more. Check it out! 
http://discover.yahoo.com/online.html---BeginMessage---
Cher Areski,

Je vis en Afrique du Sud et je suis developpeur depuis
11 ans. J'ai pu installer votre application: AreskiCC
sous RedHat 9. Cependant lorsque je passe l'ecran
d'authentification je ne vois aucun element de menu(a
index2.php).

 Il faut dire que les instructions d'installation sont
souvent incorrectes ou imprecises. J'ai du a plusieurs
reprises examiner le contenu des fichiers du programme
pour deviner ce qu'il y avait lieu de faire.

Je voit que index2.php teste si l'utilisateur a le
droit  d'access a un element de menu avant d'afficher
l'element. Cependant root/myroot ou admin/mypassword
n'ont aucun droit ou ne marchent pas!

Je soupconne que votre application fonctionne tres
bien, mais que vous RENDEZ VONLONTAIREMENT
L'INSTALLATION DIFFICILLE. Pour que tout le monde soit
oblige de vous contacter et de vous payer: C'EST UNE
PRATIQUE TRES COURANTE DANS LE MONDE GPL, mais C'EST
MALHONNETE!

Bien des choses,


Jeam Marie K

__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
---End Message---
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RE: [Asterisk-Users] Zaptel Disconnect Tone

[Dan Morin]

No one has any idea? Even a NO it cant
be done would be appreciated.



Thanks in advance.

Dan









From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dan Morin
Sent: Monday, June 20, 2005 7:24
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Zaptel
Disconnect Tone





Does anyone know if it is possible to use the following
disconnect tone setting with an x100p card?



Disconnect Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];4(.25/.25/1+2)



This tone was written for a Sipura SPA-3000 for a Panasonic
KX-TD1232. The Panasonic does not support disconnect supervision, so this
tone is the only thing that will detect a disconnect. It is not a
standard fast busy or offhook tone.



Please see this post for more info: http://www.voip-info.org/tiki-index.php?page=Asterisk-Panasonic1232vm



Any help would be greatly appreciated. Thanks in
advance.



Dan






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[Asterisk-Users] CDR: source completed with sip domain

[Rosario Pingaro]

I have Asterisk configured to be the gateway for 
sip users. 

CDR are stored using the mysql module.

But in the cdr's source filed is present only the 
user and not the domain.
I'd like to get displayed all the infos in 
this way: [EMAIL PROTECTED] and not only 
user.

In what way I can add the domain?

Thanks
Rosario


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[Asterisk-Users] DID in 513 Cincinnati

[John Kington]

Does anyone have a recommendation for a DID local to Cincinnati (513)? I am
looking for a pay as you go solution for incoming calls with light usage. I 
would

prefer IAX but can use SIP solution.
Regards,
John


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RE: [Asterisk-Users] CDR: source completed with sip domain

[jurczak]

From what I have seen,
cdr does not add the domain, maybe you could use the userfield, otherwise you
should change the source from asterisk.



-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rosario Pingaro
Sent: Sunday,
 June 26, 2005 6:59
 PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] CDR:
source completed with sip domain 





I have Asterisk configured to be the
gateway for sip users. 











CDR are stored using the mysql
module.











But in the cdr's source filed is
present only the user and not the domain.





I'd like to get displayed all
the infos in this way: [EMAIL PROTECTED] and
not only user.











In what way I can add the domain?











Thanks





Rosario




















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Re: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!

[Peter Bowyer]

I'm sure Areski would be delighted to refund your purchase price in
full if the application doesn't meet your needs.

On 26/06/05, Khubeka JM [EMAIL PROTECTED] wrote:
 
 Note: forwarded message attached.
 
 
 __
 Discover Yahoo!
 Have fun online with music videos, cool games, IM and more. Check it out!
 http://discover.yahoo.com/online.html
 
 
 -- Forwarded message --
 From: Khubeka JM [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Date: Sun, 26 Jun 2005 05:14:44 -0700 (PDT)
 Subject: JE TROUVE QUE VOUS N'ETES PAS HONETE!
 Cher Areski,
 
 Je vis en Afrique du Sud et je suis developpeur depuis
 11 ans. J'ai pu installer votre application: AreskiCC
 sous RedHat 9. Cependant lorsque je passe l'ecran
 d'authentification je ne vois aucun element de menu(a
 index2.php).
 
 Il faut dire que les instructions d'installation sont
 souvent incorrectes ou imprecises. J'ai du a plusieurs
 reprises examiner le contenu des fichiers du programme
 pour deviner ce qu'il y avait lieu de faire.
 
 Je voit que index2.php teste si l'utilisateur a le
 droit  d'access a un element de menu avant d'afficher
 l'element. Cependant root/myroot ou admin/mypassword
 n'ont aucun droit ou ne marchent pas!
 
 Je soupconne que votre application fonctionne tres
 bien, mais que vous RENDEZ VONLONTAIREMENT
 L'INSTALLATION DIFFICILLE. Pour que tout le monde soit
 oblige de vous contacter et de vous payer: C'EST UNE
 PRATIQUE TRES COURANTE DANS LE MONDE GPL, mais C'EST
 MALHONNETE!
 
 Bien des choses,
 
 Jeam Marie K
 
 __
 Do You Yahoo!?
 Tired of spam?  Yahoo! Mail has the best spam protection around
 http://mail.yahoo.com
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
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-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *-Euroisdn Italy

[Robert Rozman]

- Original Message - 
From: Emanuele Pucciarelli [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Friday, June 24, 2005 11:12 PM
Subject: Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- 
*-Euroisdn Italy




Robert Rozman wrote:

I wanted to do this (it's principle I always follow) , but we even
haven't received offer to pay for the stuff (we applied twice for offer
of two cards), so bought where we actually could buy something...


A customer of mine has had the same problem with the Italian dealer:
they behaved as though they didn't want to sell :(



I had this experience with original company No answer for 14 days...

So I got a little precausious, how would SW-drivers support look like, if 
someone even doesn't want to sell HW...


Regards,

Rob. 


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[Asterisk-Users] need help for configuring voicemail with db

[harry gaillac]

Hello,

I read wiki however I can't install voicemail.conf i
wish 
to stored voicemail conf and voice messages via odbc
in mySQL db .

Thanks for help 
Harry






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Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *-Euroisdn Italy

[Emanuele Pucciarelli]

Robert Rozman wrote:

 I had this experience with original company No answer for 14 days...
 
 So I got a little precausious, how would SW-drivers support look like,
 if someone even doesn't want to sell HW...

Well, at least they wrote them :)  Anyway, a bri [intense] debug is in
order to help you on the dropped calls problem :)

-- 
Emanuele (from Videm ;) )
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[Asterisk-Users] teliax [Was: LiveVoip is Bankrupt]

[Adam Megacz]

Rich Adamson [EMAIL PROTECTED] writes:
 I've had pretty good luck with www.teliax.com

I like them too, except for support.  I have THREE tickets open with
them that are ten days old and haven't received even a cursory we're
looking into it response.  It's absurd.

Also, for some reason you can't call American Express customer support
(800 number) through them -- the call simply doesn't connect.  I've
also had that problem with one other 800 number.

  - a

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RE: [Asterisk-Users] indexing tables for dialing

[Ipek Zivane]

Hi Jay,

Thanks for your response.

Using the fisrt apporach seems to work just for one number extension and one 
phone number. It is not clear for me how ca I manege to use one number 
extension for at least three equivalent phone numbers.


Ypek




From: Jay Milk [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Subject: RE: [Asterisk-Users] indexing tables for dialing
Date: Thu, 23 Jun 2005 09:01:05 -0500

Two approaches come to mind -- 1) Using DBPut/DBGet to associate a fixed
amount of phone-numbers with a given extension and dial, all from
extensions.conf, or 2) Using a small mySQL table and a short agi script
to accomplish the same thing.  The former solution has the advantage
that it's rather easy to implement and won't require any additional
components; the latter is more flexible and could allow maintenance of
the forward numbers by, say, a website.

 -Original Message-
 From: Ipek Zivane [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, June 22, 2005 6:50 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] indexing tables for dialing


 Hello

 I would like to know how can I manage to implement a table
 which translates
 an extension number into a phone number. Let see an example:

 If I dial an extension like 3021, Asterisk  has  to Dial an
 agent (our
 employees) located at San Francisco  using the following
 telephone number:
 415 541 . If it does not work we can also use his/her
 mobile number.

 We need to manage more than 180 agents nationwide so I would
 like to use a
 table or data base to translate a large number of agent's telephones.

 The table looks like this:

 EXTPHONE1PHONE2   PHONE3

 3021  4155   415Y   510X
 2130  415Z510L
 3060  510X   XXX
 
 .
 .
    XXX XXX

 Thanks in advance for your help.

 Ypek

 _
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RE: [Asterisk-Users] Horrible MeetMe performance

[Andy Rosen]

Did you ever have any luck with improving the MeetMe performance? We're
running into the same problems

Andy

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Lixfeld
Sent: Thursday, December 09, 2004 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Horrible MeetMe performance

Hey folks,

Using FreeBSD 5.2.1 and I've got the current zaptel driver installed 
from ports (0.8_1) and current ports asterisk (1.0.1).  I've set 
options HZ=1000 in my kernel config, recompiled and rebooted and as far 
as I can tell, I've done everything right but when I try to use the 
conference, the audio is very delayed, choppy and segmented -- totally 
unusable.

At the suggestion of someone on #asterisk, I cvsup'd * against digium 
and used that instead of ports, but that didn't seem to help either.

FYI:  When I said above when I try to use the conference I meant 
using two non-voip phones, specifically a cell phone and a land line.  
I'd dial the number for my asterix box which is in itself a b channel 
on a PRI answered by a T100P on a friend's * box and sent via IAX over 
to my * box.  Not sure if that matters, but I figure I'd mention it 
anyway.

Anyone have any ideas here?

# meetme.conf
[rooms]
conf = 97531,24680

# extensions.conf
[conf]
exten = 1,1,Answer
exten = 1,2,Wait(1)
exten = 1,3,Authenticate(5447847)
exten = 1,4,MeetMe(97531,Mas,24680)
exten = 1,5,Playback(vm-goodbye)
exten = 1,6,Hangup()
exten = 2,1,MeetMe(97531,Ms,24680)

[EMAIL PROTECTED]://~ ]$ kldstat
Id Refs AddressSize Name
  15 0xc040 5e16d8   kernel
  24 0xc231e000 2f000zaptel.ko
  31 0xc234f000 6000 wcfxo.ko
  41 0xc2355000 a000 wcfxs.ko
  51 0xc235f000 2000 ztdummy.ko
[EMAIL PROTECTED]://~ ]$

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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Orlando Guitián]

If anybody is interested, we offer a VoIP solution.  we have manufactured 
our own equipment and network from the ground up.  The service has been 
selling successfully selling for over one year (both domestic and 
internation).  If interested, let know and i will send you pricing and 
information, [EMAIL PROTECTED]



From: Rich Adamson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt
Date: Sun, 26 Jun 2005 07:01:11 -0600

I've had pretty good luck with www.teliax.com



 I found a large IAX supported provider beside Voipjet. Now 
...?


 Bashir

 I still i have good balance with them, I dont know what will be happend.
 and my canadian DID .
 - Original Message -
 From: Brian Capouch [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Sunday, June 26, 2005 1:09 AM
 Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt


  Yair Hakak wrote:
   well, i can't say i'm surprised. any company whose approach to
   customers is you are all scum trying to cheat us, don't ask
   questions, and we'll help you when we feel like it isn't going to 
be

   around for a long time.
  
 
  I agree totally.  After seeing some of the issues people were having
  with their customer support (or better, flying off the handle at 
their

  customers) I decided to stay clear of them.
 
  Survival of the fittest . . .
 
  B.
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---End of Original Message-


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Re: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!

[Areski K]

Well, did you get me close to a heart attack..! Please, this is Sunday, the
most peaceful day of the week! :)
Had you simply sent me an e-mail off-list, I would have been pleased to
help. Just like I answer each week some 30 e-mails from users asking for
help. For free...

First, let me assure you that the soft works perfectly and that installation
instructions are more than complete!
Hundreds of people on this list use it and are really happy with it. I
receive all the time e-mails of thanks  lots of good feedback.


Regarding your issues, let's have a look at the wiki page
REQUIREMENTS:

   * Apache
   * PHP  php-pgsql
   * postgresql
   * use phpagi included (http://phpagi.sourceforge.net)
   * php.ini : register_global = On


Did you put the register_global = On ? Hm, did you ??
I guess you didn't  ;p
Okey, shame on me for the register, but I will change this in the next version
(sorry about my lazy programming :-)


If you are stuck somewhere else, I advise you to look at the 'Idiots Guide',
where the installation process is really well described step by step
(unfortunately, it's more for the CentOS distro)
http://voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+Th
e+idiots+guide



From this point on, your next steps should be:
1) get a good cup of tea
2) try to relax a bit
3) change your php.ini (blame me a little, only a little)
4) reload httpd server
5) try  enjoy
6) an apology would be much appreciated, but I can live without


Have a nice Sunday, too!

Yours sincerely, 
Areski (the worst man ever!!!)




On 6/26/05, Khubeka JM [EMAIL PROTECTED] wrote:
 
 Note: forwarded message attached.
 
 
 
 
 __
 Discover Yahoo!
 Have fun online with music videos, cool games, IM and more. Check it out!
 http://discover.yahoo.com/online.html
 
 
 -- Forwarded message --
 From: Khubeka JM [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Date: Sun, 26 Jun 2005 05:14:44 -0700 (PDT)
 Subject: JE TROUVE QUE VOUS N'ETES PAS HONETE!
 Cher Areski,
 
 Je vis en Afrique du Sud et je suis developpeur depuis
 11 ans. J'ai pu installer votre application: AreskiCC
 sous RedHat 9. Cependant lorsque je passe l'ecran
 d'authentification je ne vois aucun element de menu(a
 index2.php).
 
  Il faut dire que les instructions d'installation sont
 souvent incorrectes ou imprecises. J'ai du a plusieurs
 reprises examiner le contenu des fichiers du programme
 pour deviner ce qu'il y avait lieu de faire.
 
 Je voit que index2.php teste si l'utilisateur a le
 droit  d'access a un element de menu avant d'afficher
 l'element. Cependant root/myroot ou admin/mypassword
 n'ont aucun droit ou ne marchent pas!
 
 Je soupconne que votre application fonctionne tres
 bien, mais que vous RENDEZ VONLONTAIREMENT
 L'INSTALLATION DIFFICILLE. Pour que tout le monde soit
 oblige de vous contacter et de vous payer: C'EST UNE
 PRATIQUE TRES COURANTE DANS LE MONDE GPL, mais C'EST
 MALHONNETE!
 
 Bien des choses,
 
 
 Jeam Marie K
 
 __
 Do You Yahoo!?
 Tired of spam?  Yahoo! Mail has the best spam protection around
 http://mail.yahoo.com
 
 
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[Asterisk-Users] TDM400P (TDM02B) ceased to work...

[Francois Meehan]

Hi all,

I am runing Asterisk on Centos 4. This morning I have updated the system
using yum, a whole bunch of stuff was upgraded.

Since, when I try to start zaptel, I have the following error:
Waiting for zap to come online ...OK
Loading zaptel hardware modules:
Running ztcfg:  ZT_CHANCONFIG failed on channel 1: No such device or
address (6)
   [FAILED]

Did recompile Asterisk with version 1.0.8, reboot a couple of time, still
the same error, the card seem dead...

Any ideas?

Francois


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end of a journey. - Cynthia Ozick
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Andrew Kohlsmith]

On Sunday 26 June 2005 14:32, Orlando Guitián wrote:
 If anybody is interested, we offer a VoIP solution.  we have manufactured
 our own equipment and network from the ground up.  The service has been
 selling successfully selling for over one year (both domestic and
 internation).  If interested, let know and i will send you pricing and
 information, [EMAIL PROTECTED]

I'm sorry but anyone selling their service for over a year without bothering 
to mention their company name and indeed, using an msn account already has me 
sufficiently suspicious to decide against giving them any money.

-A.
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Re: [Asterisk-Users] ASTCC not billing

[Ade Agbero]

I set "notransfer=yes" and on SIP you set "canreinvite=no", but ASTCC is still not billing.

Iformated and reinstalled [EMAIL PROTECTED] andgot the latest CVS of Astcc, butASTCC is still not billing.

What version of [EMAIL PROTECTED] can be confirmed working with Astcc.
Darren Wiebe [EMAIL PROTECTED] wrote:
On IAX you set "notransfer=yes" and on SIP you set "canreinvite=no"DarrenJuan Luis Moyano wrote:On Sab, 25 de Junio de 2005, 5:20 pm, Darren Wiebe dijo: Do you have the notransfer and reinvite lines set properly? I had thissame problem with ASTCC but found that if I removed asterisk includingthe source and did a clean reinstall it worked suddenly.Darren Darren, how is the proper way of setting notransfer and canreinvite lineson IAX. TIA. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options
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Re: [Asterisk-Users] TDM400P (TDM02B) ceased to work...

[Robert P. McKenzie]

Did you recompile and reinstall the zaptel source?  I had to do this myself 
recently on a
fedora core 2 update/upgrade


Francois Meehan wrote:
 Hi all,
 
 I am runing Asterisk on Centos 4. This morning I have updated the system
 using yum, a whole bunch of stuff was upgraded.
 
 Since, when I try to start zaptel, I have the following error:
 Waiting for zap to come online ...OK
 Loading zaptel hardware modules:
 Running ztcfg:  ZT_CHANCONFIG failed on channel 1: No such device or
 address (6)
[FAILED]
 
 Did recompile Asterisk with version 1.0.8, reboot a couple of time, still
 the same error, the card seem dead...
 
 Any ideas?
 
 Francois
 
 
 Random Thought:
 ---
 Nothing is so awesomely unfamiliar as the familiar that discloses itself at 
 the end of a journey. - Cynthia Ozick
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-- 
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[EMAIL PROTECTED] | [EMAIL PROTECTED]
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[Asterisk-Users] help regarding h323.conf

[SYED ADEEL ALI]

i m using several IAX  SIP softphones Now I've got an IP phone (NetPhone) that supports H.323 protocol  plz tell me how should i configure it to work with asterisk ... i m comfortable with sip.conf  iax.conf but what should i do to use h323.conf do i need to install something or should i just copy h323.conf( the sample file ) to /etc/asterisk directory ???Don't just search. Find. MSN Search Check out the new MSN Search!

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Re: [Asterisk-Users] TDM400P (TDM02B) ceased to work...

[Francois Meehan]

Hi Robert,

Did recompile, several time actually, upgraded from 1.0.7 to 1.0.8 with
same results, there is no light in the back of the card.

Regards,

Francois

 Did you recompile and reinstall the zaptel source?  I had to do this
 myself recently on a
 fedora core 2 update/upgrade


 Francois Meehan wrote:
 Hi all,

 I am runing Asterisk on Centos 4. This morning I have updated the system
 using yum, a whole bunch of stuff was upgraded.

 Since, when I try to start zaptel, I have the following error:
 Waiting for zap to come online ...OK
 Loading zaptel hardware modules:
 Running ztcfg:  ZT_CHANCONFIG failed on channel 1: No such device or
 address (6)
[FAILED]

 Did recompile Asterisk with version 1.0.8, reboot a couple of time,
 still
 the same error, the card seem dead...

 Any ideas?

 Francois


 Random Thought:
 ---
 Nothing is so awesomely unfamiliar as the familiar that discloses itself
 at the end of a journey. - Cynthia Ozick
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 --
 Robert P. McKenzie, CSTA   |   GammaRay Technical Services Ltd
 [EMAIL PROTECTED] | [EMAIL PROTECTED]
 http://www.uk-experience.com   |  http://www.gammaray-tech.com

 Ecademy Profile:   http://www.ecademy.com/account.php?op=viewid=64014




Random Thought:
---
Four things cannot be hidden -- Love, smoke, a pillar of fire,
and a man striding across the open bled.

  -- Fremen Wisdom
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RE: [Asterisk-Users] Horrible MeetMe performance

[Dan Morin]

Make sure that if you're using anything other than zaptel hardware, it
is running uLaw as the codec.  Anything else will produce ever
increasing delays.

My setup has all of our VoIP lines coming into my main box, and then I
have a separate box running asterisk only for meetme with an iax2 trunk
between the two running uLaw.  It seems to work fairly well.

Dan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Rosen
Sent: Sunday, June 26, 2005 2:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Horrible MeetMe performance

Did you ever have any luck with improving the MeetMe performance? We're
running into the same problems

Andy

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Lixfeld
Sent: Thursday, December 09, 2004 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Horrible MeetMe performance

Hey folks,

Using FreeBSD 5.2.1 and I've got the current zaptel driver installed 
from ports (0.8_1) and current ports asterisk (1.0.1).  I've set 
options HZ=1000 in my kernel config, recompiled and rebooted and as far 
as I can tell, I've done everything right but when I try to use the 
conference, the audio is very delayed, choppy and segmented -- totally 
unusable.

At the suggestion of someone on #asterisk, I cvsup'd * against digium 
and used that instead of ports, but that didn't seem to help either.

FYI:  When I said above when I try to use the conference I meant 
using two non-voip phones, specifically a cell phone and a land line.  
I'd dial the number for my asterix box which is in itself a b channel 
on a PRI answered by a T100P on a friend's * box and sent via IAX over 
to my * box.  Not sure if that matters, but I figure I'd mention it 
anyway.

Anyone have any ideas here?

# meetme.conf
[rooms]
conf = 97531,24680

# extensions.conf
[conf]
exten = 1,1,Answer
exten = 1,2,Wait(1)
exten = 1,3,Authenticate(5447847)
exten = 1,4,MeetMe(97531,Mas,24680)
exten = 1,5,Playback(vm-goodbye)
exten = 1,6,Hangup()
exten = 2,1,MeetMe(97531,Ms,24680)

[EMAIL PROTECTED]://~ ]$ kldstat
Id Refs AddressSize Name
  15 0xc040 5e16d8   kernel
  24 0xc231e000 2f000zaptel.ko
  31 0xc234f000 6000 wcfxo.ko
  41 0xc2355000 a000 wcfxs.ko
  51 0xc235f000 2000 ztdummy.ko
[EMAIL PROTECTED]://~ ]$

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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Orlando Guitián]

here is the information.  The website is in spanish (www.sebell.com), 
therefore, i will send you the information in english on monday.  To answer 
your question, the service has been sold primarly to international banking 
institutions and financial organizations via word of mouth.  We are 
currently translating the web site to english.


The service provides calling within the USA and Canada as well as 
international access.  Users (and supervisors) have realtime access to their 
phone calls and billing.


The DIDs are provided for Miami (area codes 305 and 786) as well as Buenos 
Aires (+54 11), Argentina.


The web site:
www.sebell.com

My corporate email:
[EMAIL PROTECTED]



From: Andrew Kohlsmith [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt
Date: Sun, 26 Jun 2005 15:20:08 -0400

On Sunday 26 June 2005 14:32, Orlando Guitián wrote:
 If anybody is interested, we offer a VoIP solution.  we have 
manufactured

 our own equipment and network from the ground up.  The service has been
 selling successfully selling for over one year (both domestic and
 internation).  If interested, let know and i will send you pricing and
 information, [EMAIL PROTECTED]

I'm sorry but anyone selling their service for over a year without 
bothering
to mention their company name and indeed, using an msn account already has 
me

sufficiently suspicious to decide against giving them any money.

-A.
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[steve szmidt]

On Sunday 26 June 2005 15:20, Andrew Kohlsmith wrote:
 On Sunday 26 June 2005 14:32, Orlando Guitián wrote:
  If anybody is interested, we offer a VoIP solution.  we have manufactured
  our own equipment and network from the ground up.  The service has been
  selling successfully selling for over one year (both domestic and
  internation).  If interested, let know and i will send you pricing and
  information, [EMAIL PROTECTED]

 I'm sorry but anyone selling their service for over a year without
 bothering to mention their company name and indeed, using an msn account
 already has me sufficiently suspicious to decide against giving them any
 money.

It would seem that people just don't realize how it makes them look.

-- 

Steve Szmidt

They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety.
Benjamin Franklin
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[Asterisk-Users] Prepaid for mysql and simple auth

[Ricardo Poppi]

Hi list.

I´ve been considering in the last few days witch prepaid solution I´m 
going to test into my asterisk environment.


The main issue about it is that I could not find a solution to use into 
my already-installed-and-fine-tunned mysql db that can autenticate the 
user without using those long-and-anoying PIN numbers but just with the 
from/contact or even digest of the sip message.


Do anybody has a clue to help me?

Thanks in advance,

Ricardo Poppi.
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[Asterisk-Users] DID in 513 Cincinnati

[Jeff Glassman]

Message: 19
Date: Sun, 26 Jun 2005 12:12:46 -0400
From: John Kington [EMAIL PROTECTED]
Subject: [Asterisk-Users] 
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii; format=flowed

Does anyone have a recommendation for a DID local to Cincinnati (513)? I
am
looking for a pay as you go solution for incoming calls with light
usage. I 
would
prefer IAX but can use SIP solution.
Regards,
John


Try telasip.com for SIP I am using them in 614 (Columbus) or
www.teliax.com, I have not used them but have heard good things.

Jeff


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Re: [Asterisk-Users] LiveVoip is Bankrupt

[trixter http://www.0xdecafbad.com]

On Sun, 2005-06-26 at 11:32 -0700, Brian Litzinger wrote:
 Anytime a small new organization asks for up front payment, I wonder 
 about a locally famous case.
 
 Back in the days of beepers, a local company was selling beeper
 service for about 30% less than anyone in exchange for a relatively
 good portion of payment up front.  Can't remember if it was
 3 months or 1 year.  May have been both.
 
 They collected up the payments and paid the money out in huge salaries.
 
 Then they went bankrupt.  It is apparently difficult for bankruptcy
 courts to recover salary payments.
 
 It is apparently a well known scam executed in a number of different
 ways.

I recall a case against someone in New York City where a lady was doing
that with travel, selling cruises below her cost, etc.  Presales upto 6
months ahead went to pay for tickets today.  She paid herself $100k for
her services.  They indicted her on fraud becuase it is illegal to sell
stuff below cost, knowing that you cant possibly make good on what you
sell.  

Perhaps the same could be true of livevoip for anyone that lost any big
amount of money for prepayment on services they couldnt render.  And
certainly for payments where the 'writing was on the wall', ie they knew
they were going to file bankrupcy yet accepted payments for months they
knew they wouldnt be in business.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


signature.asc
Description: This is a digitally signed message part
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[trixter http://www.0xdecafbad.com]

On Sun, 2005-06-26 at 14:32 -0400, Orlando Guitián wrote:
 If anybody is interested, we offer a VoIP solution.  we have manufactured 
 our own equipment and network from the ground up.  The service has been 
 selling successfully selling for over one year (both domestic and 
 internation).  If interested, let know and i will send you pricing and 
 information, [EMAIL PROTECTED]

A year of selling 'domestic and internation' and you have an msn.com
email address not one that is off the domian of the company you
represent?  Interesting concept, does that really yield higher sales?
Could be a new marketing stragety I am unfamiliar with.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] handle wrong extensions in Dialplam

[Wayne]

Mahmoud Badran wrote:


Hello;
i am trying to make a dial plan that can handle any wrong extensions 
dialled from the local sip phone for example so that if i dialled the 
right extension it rings but if i dialled wrong or existing extension 
it redirect him to the Main menu for example?


thanks in advance 


This seems to be popular at the mo...
Heres one I created earlier... 
http://www.planetwayne.com/forums/viewtopic.php?t=218



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Re: [Asterisk-Users] LiveVoip is Bankrupt

[snacktime]

For those that paid by credit card, you can call your bank and get any
amount they owe you refunded.  You are not a creditor as far as the
bankruptcy is concerned, the acquring bank is.

Chris
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RE: [Asterisk-Users] LiveVoip is Bankrupt

[Alexander Lopez]

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 trixter http://www.0xdecafbad.com
 Sent: Sunday, June 26, 2005 4:21 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt
 
 On Sun, 2005-06-26 at 11:32 -0700, Brian Litzinger wrote:
  Anytime a small new organization asks for up front payment, 
 I wonder 
  about a locally famous case.
  
  Back in the days of beepers, a local company was selling beeper 
  service for about 30% less than anyone in exchange for a relatively 
  good portion of payment up front.  Can't remember if it was
  3 months or 1 year.  May have been both.
  
  They collected up the payments and paid the money out in 
 huge salaries.
  
  Then they went bankrupt.  It is apparently difficult for bankruptcy 
  courts to recover salary payments.
  
  It is apparently a well known scam executed in a number of 
 different 
  ways.
 
 I recall a case against someone in New York City where a lady 
 was doing that with travel, selling cruises below her cost, 
 etc.  Presales upto 6 months ahead went to pay for tickets 
 today.  She paid herself $100k for her services.  They 
 indicted her on fraud becuase it is illegal to sell stuff 
 below cost, knowing that you cant possibly make good on what 
 you sell.  
 
 Perhaps the same could be true of livevoip for anyone that 
 lost any big amount of money for prepayment on services they 
 couldnt render.  And certainly for payments where the 
 'writing was on the wall', ie they knew they were going to 
 file bankrupcy yet accepted payments for months they knew 
 they wouldnt be in business.
 
 
 -- 
 Trixter http://www.0xdecafbad.com Bret McDanel
 UK +44 870 340 4605   Germany +49 801 777 555 3402
 US +1 360 207 0479 or +1 516 687 5200
 FreeWorldDialup: 635378

If I recall recently LiveVoIP touted it 'merger' with a large
corporation (daddy BigBucks). I also remember that it was posted that it
was not in the best interest of LiveVoip to move forward on the merger.
In hind sight this is probably the result of the due-diligence done on
the larger corporation's side. Citing non-disclosure and other standard
agreements entered during transactions such as these it does not
surprise me that we (the customers) were not told about the writing on
the wall..



Alex
 
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Re: [Asterisk-Users] ASTCC not billing

[Darren Wiebe]

Ade Agbero wrote:

I set notransfer=yes and on SIP you set canreinvite=no, but ASTCC  
is still not billing.
 
I formated and reinstalled [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
and got the latest CVS of Astcc, but ASTCC  is still not billing.
 
What version of [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] can be confirmed 
working with Astcc.


I can try to find that out tomorrow.  I don't know off the top of my 
head.  Did you try the latest stable version of asterisk?  That is what 
I did to resolve the issue.


Darren Wiebe
[EMAIL PROTECTED]



*/Darren Wiebe [EMAIL PROTECTED]/* wrote:

On IAX you set notransfer=yes and on SIP you set canreinvite=no

Darren

Juan Luis Moyano wrote:

On Sab, 25 de Junio de 2005, 5:20 pm, Darren Wiebe dijo:


Do you have the notransfer and reinvite lines set properly? I
had this
same problem with ASTCC but found that if I removed asterisk
including
the source and did a clean reinstall it worked suddenly.

Darren



Darren, how is the proper way of setting notransfer and
canreinvite lines
on IAX. TIA.




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How much free photo storage do you get? Store your holiday snaps for 
FREE with Yahoo! Photos. *Get Yahoo! Photos* 
http://us.rd.yahoo.com/mail/uk/taglines/default/photos/*http://uk.photos.yahoo.com/ 





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Re: [Asterisk-Users] zeroconf help

[Matt Riddell]

stevanus wrote:

hi,

recently I installed zeroconf for asterisk...
I've already followed the asterisk+zeroconf how to (which is too short), 
but it came with an error message...


asterisk: relocation error: /usr/lib/asterisk/modules/res_zeroconf.so: 
undefined symbol: DNSServiceRegister

Ouch ... error while writing audio data: : Broken pipe

it's weird since I've double checked the library and header from 
zeroconf and it seems that everything has been in the right place..


Is there anyone can help me? Well, it seems I hit another dead end this 
time...


You might want to try asking the Astmasters Zeroconf Project Team  
[EMAIL PROTECTED] 


--
Cheers,

Matt Riddell
___

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Re: [Asterisk-Users] Asterisk Manager Interface Remote Buffer Overflow Vulnerability

[Matt Riddell]

Zoa wrote:


Haha, fun.


Why use the bufferoverflow if you already have the permissions to
execute any linux command using the manager interface :p


LOL that's what I was thinking!

A couple of weeks ago I used the manager interface to recreate whole 
files on a dead PC.


I ended up having problems with the ! mode and so used addexten to add 
extensions that ran system commands to recreate the files when I dialled 
a particular extension.


Took a while, but I got there in the end!

:)

Not that I'm complaining about people doing security audits though, it 
must be nearly a year since the last lot was done.


--
Cheers,

Matt Riddell
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Re: [Asterisk-Users] TDM400P (TDM02B) ceased to work...

[Francois Meehan]

Got it,

For some strange reasons, neither wctdm nor wcfxs get loaded when starting
zaptel (/etc/init.d/zaptel start).

By manually modproble wctdm everything works.

Have all a nice week.

Francois


 Hi all,

 I am runing Asterisk on Centos 4. This morning I have updated the system
 using yum, a whole bunch of stuff was upgraded.

 Since, when I try to start zaptel, I have the following error:
 Waiting for zap to come online ...OK
 Loading zaptel hardware modules:
 Running ztcfg:  ZT_CHANCONFIG failed on channel 1: No such device or
 address (6)
[FAILED]

 Did recompile Asterisk with version 1.0.8, reboot a couple of time, still
 the same error, the card seem dead...

 Any ideas?

 Francois


 Random Thought:
 ---
 Nothing is so awesomely unfamiliar as the familiar that discloses itself
 at the end of a journey. - Cynthia Ozick
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Random Thought:
---
I suppose we acquire most of our feelings about our bodies too early, and in 
ways too complicated, to make them easy to account for. - Charis Wilson
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[Asterisk-Users] Changing Caller ID

[Jeff Glassman]

I have two X100P clone cards working perfectly
in my asterisk box, these lines are off an analog extension from a PRI.



They each have DID # assigned to them and I can call the DID
and receive calls. When I make an
outgoing call using the Zap trunk the caller ID is of the PRI line. Is there any way to change the caller ID
to the DID assigned to the line?



Thanks in advance,



Jeff






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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Matt Riddell]

Andres wrote:

So it looks like Livevoip went Bankrupt


Sh1t.

Looks like the Daily Asterisk News will need a new host.

So, unless anyone can donate space for a custom php and mysql based 
site, it will be hosted in either New Zealand or Italy.


Offers?

--
Cheers,

Matt Riddell
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RE: [Asterisk-Users] LiveVoip is Bankrupt

[Joshua Colp]

Matt - catch me on IRC (it's file).

- Joshua Colp. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Sunday, June 26, 2005 6:30 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

Andres wrote:
 So it looks like Livevoip went Bankrupt

Sh1t.

Looks like the Daily Asterisk News will need a new host.

So, unless anyone can donate space for a custom php and mysql based site, it
will be hosted in either New Zealand or Italy.

Offers?

--
Cheers,

Matt Riddell
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Peter Corlett]

Matt Riddell [EMAIL PROTECTED] wrote:
[...]
 Looks like the Daily Asterisk News will need a new host. So, unless
 anyone can donate space for a custom php and mysql based site, it
 will be hosted in either New Zealand or Italy.

 Offers?

How much bandwidth does it consume?

-- 
You fall out of your mother's womb, you crawl across open country under fire,
and drop into your grave.
- Quentin Crisp
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[Asterisk-Users] Re: Horrible MeetMe performance

[Tony Mountifield]

In article [EMAIL PROTECTED],
Dan Morin [EMAIL PROTECTED] wrote:
 Make sure that if you're using anything other than zaptel hardware, it
 is running uLaw as the codec.  Anything else will produce ever
 increasing delays.

Hey, now that's a snippet of information I hadn't seen before!

All my work with SIP and MeetMe is using aLaw, since I'm in the UK.

Do you know why it causes ever increasing delays? I would have thought
that a transcoding would just introduce a contstant (small) delay, not
an accumulating one. So if you're right, then it ought to be fixable,
once the mechanism is understood.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Andres]

snacktime wrote:


For those that paid by credit card, you can call your bank and get any
amount they owe you refunded.  You are not a creditor as far as the
bankruptcy is concerned, the acquring bank is.

Chris
___


 


Hi Chris,

I am curious to know how this would work in this case.  Lets assume 
someone purchased $100 worth of LiveVoip service using his Bank of 
America Visa card (and did not get a chance to use the service).  So now 
LiveVoip is bankrupt and lets assume the owners fled with the money they 
made so the bank accounts are cleaned out.  If the person now calls Bank 
of America to dispute the charge, then who loses the $100 in this case?  
Visa, Bank of America, or the consumer?



--
Andres



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Re: [Asterisk-Users] Zaptel card AND Ztdummy together?

[Rod Bacon]

In case anyone is interested, loading Ztdummy AND a card driver at the same 
time will result in unpredictable timing issues.

We heard intermittent echo/feedback on PRI channels.


Rod Bacon wrote:
I had a weird (unforeseen) situation today. We have a remote office with 
an * server and ISDN 10 service. We connect to each other over an IAX 
trunk with G729.


Today, some of Sydney experienced a power surge which knocked out their 
ISDN services. Without a clock source on their PRI card, my IAX calls to 
them resulted in one-way audio (they could hear me, but I not them).


Is it possible to load *both* the relevant card driver *and* ztdummy to 
guard against this occurrance?

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Re: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!

[harry gaillac]

I agree you.

Does asterisk (Digium) project provide a good
documentation ?

Does Asterisk Handbook has been released ?

When developpers improve Asterisk where are you
looking for help, mailing list, wiki, asteriskdocs,
...-:(

It's the job to all Asterisk developpers/users to
provide docs on Asterisk.org.

In fact Digium host both a open project and a
commercial site.
www.digium.com. 86299   IN  A  
69.16.138.164

www.asterisk.org.   86254   IN  A  
69.16.138.164

Asterisk/Digium don't provide docs so you have to pay
for help or waste time to google.

Open source community as often criticize enterprises
like Microsoft, Cisco, 
However these ones pay RD.

May be enterprises like Digium an others want to earn
money with works of open source community

I tell to these enterprises you want to earn money do
like Microsoft

Harry from France


 Je soupconne que votre application fonctionne tres
 bien, mais que vous RENDEZ VONLONTAIREMENT
 L'INSTALLATION DIFFICILLE. Pour que tout le monde
soit
 oblige de vous contacter et de vous payer: C'EST UNE
 PRATIQUE TRES COURANTE DANS LE MONDE GPL, mais C'EST
 MALHONNETE!
 

  Jeam Marie K







___ 
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Téléchargez cette version sur http://fr.messenger.yahoo.com
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Listacc]

are you guys still looking for space? I can donate you some space tell 
me how much you need? I own a communications company based in Tulsa, OK. 
OCOSA Communications, LLChttp://www.ocosa.com We don't generally do 
we just started and mySQL as well give me a quote and I 'll will get you 
hooked up if your interested!


Otis Surratt Jr.

Peter Corlett wrote:


Matt Riddell [EMAIL PROTECTED] wrote:
[...]
 


Looks like the Daily Asterisk News will need a new host. So, unless
anyone can donate space for a custom php and mysql based site, it
will be hosted in either New Zealand or Italy.
   



 


Offers?
   



How much bandwidth does it consume?

 




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Re: [Asterisk-Users] Changing Caller ID

[Bryce Chidester]

The callerid on outside lines is set by your carrier. Talk to them.

Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:  
[EMAIL PROTECTED]/305



On Jun 26, 2005, at 14:25, Jeff Glassman wrote:


I have two X100P clone cards working perfectly in my asterisk box,  
these lines are off an analog extension from a PRI.




They each have DID # assigned to them and I can call the DID and  
receive calls.  When I make an outgoing call using the Zap trunk  
the caller ID is of the PRI line.  Is there any way to change the  
caller ID to the DID assigned to the line?




Thanks in advance,



Jeff


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RE: [Asterisk-Users] LiveVoip is Bankrupt

[trixter http://www.0xdecafbad.com]

On Sun, 2005-06-26 at 16:42 -0400, Alexander Lopez wrote:

 If I recall recently LiveVoIP touted it 'merger' with a large
 corporation (daddy BigBucks). I also remember that it was posted that it
 was not in the best interest of LiveVoip to move forward on the merger.
 In hind sight this is probably the result of the due-diligence done on
 the larger corporation's side. Citing non-disclosure and other standard
 agreements entered during transactions such as these it does not
 surprise me that we (the customers) were not told about the writing on
 the wall..

My writing on the wall reference was not towards customers, instead it
was towards livevoip (or any other company) when they accepted money for
service they knew they could not provide.  

In america at least (most other countries most likely have laws against
this as well) it is illegal to accept money for services you know you
cannot provide.  It is also illegal (falls under fraud) for companies to
sell services below cost knowing they will drive themselves into the
ground and file bankrupcy.

While some may be able to get credit card refunds (depending on a
variety of factors, like how long ago they were charged, any court
orders in place right now, etc - most banks wont give you a refund if
they know they wont get any money from the merchant, unless you can
prove fraud to some degree) there are more than likely more customers
that will not.  The only way to go after anything would be to go after
the people involved (a corporation does not shield oneself against
illegal actions - if that were the case CEOs across the country wouldnt
be in jail, have been in jail, facing jail, or trying to appeal their
jail sentences).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] LiveVoip is Bankrupt

[trixter http://www.0xdecafbad.com]

On Sun, 2005-06-26 at 23:29 +0200, Matt Riddell wrote:
 Andres wrote:
  So it looks like Livevoip went Bankrupt
 
 Sh1t.
 
 Looks like the Daily Asterisk News will need a new host.
 
 So, unless anyone can donate space for a custom php and mysql based 
 site, it will be hosted in either New Zealand or Italy.
 
 Offers?
 

sourceforge asterisk daily news documentation project?  They have some
bandwidth, file space, php and mysql are reported to work...

Dunno if this will fit your goals though.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Sahil Gupta]

E-mail me off-list, we'll help out :-)

Regards,


Sahil Gupta
VoiceValley

On Sun, 26 Jun 2005, trixter http://www.0xdecafbad.com wrote:


On Sun, 2005-06-26 at 23:29 +0200, Matt Riddell wrote:

Andres wrote:

So it looks like Livevoip went Bankrupt


Sh1t.

Looks like the Daily Asterisk News will need a new host.

So, unless anyone can donate space for a custom php and mysql based
site, it will be hosted in either New Zealand or Italy.

Offers?



sourceforge asterisk daily news documentation project?  They have some
bandwidth, file space, php and mysql are reported to work...

Dunno if this will fit your goals though.

--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Rich Adamson]

  If anybody is interested, we offer a VoIP solution.  we have manufactured
  our own equipment and network from the ground up.  The service has been
  selling successfully selling for over one year (both domestic and
  internation).  If interested, let know and i will send you pricing and
  information, [EMAIL PROTECTED]
 
 I'm sorry but anyone selling their service for over a year without bothering 
 to mention their company name and indeed, using an msn account already has me 
 sufficiently suspicious to decide against giving them any money.

I'll second that one big time!


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Re: [Asterisk-Users] LiveVoip is Bankrupt

[snacktime]

 
 While some may be able to get credit card refunds (depending on a
 variety of factors, like how long ago they were charged, any court
 orders in place right now, etc - most banks wont give you a refund if
 they know they wont get any money from the merchant, unless you can
 prove fraud to some degree) there are more than likely more customers
 that will not. 

Doesn't work that way.  Issuing banks are guaranteed payment by
acquiring banks.  It's the acquiring bank that has to eat the loss,
not the issuing bank.  Issuing banks eat losses when a cardholder
defaults, but never when a merchant defaults.

And in cases where the service is delivered over an extended period of
time, the clock for when you can chargeback doesnt' start ticking
until that time period is up.  That's why acquirers don't like prepaid
plans or extended length subscriptions.  Someone like livevoip can
charge a bunch of people and the acquiring bank can be eating losses
over a year out.

Chris
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RE: [Asterisk-Users] Re: Horrible MeetMe performance

[Dan Morin]

I read somewhere, probably the Wiki, that the core of MeetMe uses uLaw.
So when sound comes in, it is transcoded to uLaw first, then mixed with
all of the other audio at that moment, then sent out again.  At that
point, it is transcoded again to the original format.

So, if everything is in uLaw, you bypass 2 transcoding processes.  And
if you take into account both of those transcoding processes (with other
codecs), they add 10 or so milliseconds each time which results in ever
increasing delays. 

If you are using all uLaw connections, try changing one of them to iLBC
and try another conference.  I noticed 10 second delays after 5 minutes
of conference.

Let me know if you find anything else out about this.
Dan


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Sunday, June 26, 2005 5:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Horrible MeetMe performance

In article
[EMAIL PROTECTED],
Dan Morin [EMAIL PROTECTED] wrote:
 Make sure that if you're using anything other than zaptel hardware, it
 is running uLaw as the codec.  Anything else will produce ever
 increasing delays.

Hey, now that's a snippet of information I hadn't seen before!

All my work with SIP and MeetMe is using aLaw, since I'm in the UK.

Do you know why it causes ever increasing delays? I would have thought
that a transcoding would just introduce a contstant (small) delay, not
an accumulating one. So if you're right, then it ought to be fixable,
once the mechanism is understood.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Re: Horrible MeetMe performance

[qrss]

In my experience, several seconds of delay becomes apparent over time when
using an internal clock source.  Seems its a clocking/timer issue. 
Certainly we are dealing with clocking differences over time and unsynced
samples are being significantly buffered and then replayed later.  What
are you using as a clock source? I think that basically we are looking at
50 packets/sec at 20ms each packet for alaw/ulaw voip. That is - 1 second
of speech.  Now, zaptel seems to time from whatever hardware clock is
available and almost assumes that the clock will be precisely 8000 samples
per second or 1000 interrupts/second (also 1 second of speech).  It seems
that the voip clock is slightly faster than the hardware clock that zaptel
is timing from.  The extra samples/second must be being buffered.  Of
course, this buffering would add up over time until the point that a VOIP
sample is played back several seconds out of phase.  Seems that either the
zaptel clock source must be brought to closer tolerance, or the extra data
that is being buffered must be thrown away in order to stay in sync. Any
thoughts?

-Original Message-
From: Tony Mountifield
Sent: Sun, June 26, 2005 5:35 pm

In article [EMAIL PROTECTED],
Dan Morin [EMAIL PROTECTED] wrote:
 Make sure that if you're using anything other than zaptel hardware, it
 is running uLaw as the codec.  Anything else will produce ever
 increasing delays.

Hey, now that's a snippet of information I hadn't seen before!

All my work with SIP and MeetMe is using aLaw, since I'm in the UK.

Do you know why it causes ever increasing delays? I would have thought
that a transcoding would just introduce a contstant (small) delay, not
an accumulating one. So if you're right, then it ought to be fixable,
once the mechanism is understood.

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Missing first second of voice on outgoing SIP/IAX calls

[snacktime]

I've been curious about why this happens for the longest time.  When I
make an outgoing SIP/IAX call the first half second or so of the voice
never makes it to me.  This is consistant on every provider I have
used except for voicepulse, and it always happens.  With voicepulse it
never happens.  It doesn't seem to make any difference whether it's
SIP or IAX.  I don't really want to mention any names because this
isn't really a gripe, it's just got me very curious.

On these same providers I do not get the lost voice on incoming calls.

Anyone have any ideas?   The only thing I can think of is that maybe
voicepulse isn't using asterisk?

Chris
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[Asterisk-Users] APP - ValetParking on CVS-HEAD -- instructions on its use, anyone?

[Chris Coulthurst]

I've finally got this beast installed, but now I don't see how to use it.
I've looked over the web near and far, and it just seems that no one has
anything on its implementation.. This is all I get from the CLI:

*CLI 
  -= Info about application 'ValetParking' =- 

[Synopsis]
Valet Parking

[Description]
ValetParking(exten|lotname|timeout[|return_ext][|return_pri][|ret
urn_context])
Auto-Sense Valet Parking: if exten is not occupied, park it, if it is
already parked, bridge to it.


I can guess that everything after 'timeout' is optional because of the
brackets, but I'm confused on everything else.  What is the 'exten'?  I was
under the impression that it was auto-generating on this particular app.  I
have no idea what 'lotname' is, but I feel that there should maybe be some
lines added to features.conf for this thing?  Just no docs to tell me what
to do next.

Help, anybody?  I'd love some real-dialplan working examples...

Desparately yours,

Chris Coulthurst
[EMAIL PROTECTED]
 


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Re: [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK

[Steve Totaro]

I just purchased two rev I boards and they cannot be recognized at all but a
revision H (that reports E/F) board works ok.


- Original Message - 
From: John Novack [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 22, 2005 6:08 AM
Subject: Re: [Asterisk-Users] How can you check that egTDM04B
hardwareinstalled and drivers OK


 Probably means that your perfectly good motherboard can't see the TDM
card.
 There are many motherboards that this card doesn't seem to work with,
 Digium doesn't seem willing to address the issue or even acknowledge
 that is the case, and usually answers  try another motherboard rather
 than 'fess up that there is a design problem with the PCI interface and
 correct it.
 PCI 2.2 is a stated requirement, but there is certainly more to the
 story than that.

 In addition, when the board CAN be seen, report rev E/F when  the
 silkscreen reads Rev H, someone mentioned there is now a Rev I ( good
 luck getting an exchange ) and Digium 's answer is  if we can see it
 through remote access then there is no reason to replace it, and if we
 can't, try another MB.

 Overall, if it works, lucky you, if not, Too bad.
 Hard to support Digium and suggest others purchase such a product.
 Best you look for other interfaces to Asterisk.

 John Novack




 Angus Comber wrote:

  If I try dmesg - no mention of a Wildcard TDM400.
 
  Sorry I am fairly new to Linux.  In Windows I suppose I would run some
  hardware program which came with the card to see if I could manually
  set IRQ's etc.  What should I be looking at now?
 
  Please feel free to point me to a good book or whatever you feel is
  appropriate.  Could the card be faulty?
 
  My motherboard is an Intel D865GLC.
 
  I am running [EMAIL PROTECTED] version 1.0
 
  Angus
 
 
 
 
  - Original Message - From: Mike M
  [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  Sent: Tuesday, June 21, 2005 3:14 PM
  Subject: Re: [Asterisk-Users] How can you check that eg TDM04B
  hardwareinstalled and drivers OK
 
 
  On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote:
 
 
   I am struggling to get my TDM04B working.  Just to rule out a
  hardware  problem how can I check
  that the hardware works?  How can I then
   check that the drivers are loaded correctly?
  
 
  1. from the linux command line, type 'dmesg' and look for
   Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
  if you see that, the TDM card is recognized by the OS.
 
 
  Here's what I get on a working system:
 
  [EMAIL PROTECTED] src]# modprobe wctdm
  [EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for 
  device
  01:0a.0
  Jun 21 10:10:55 b2 kernel: Freshmaker version: 71
  Jun 21 10:10:55 b2 kernel: Freshmaker passed register test
  Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO
  Jun 21 10:10:55 b2 kernel: Module 1: Not installed
  Jun 21 10:10:55 b2 kernel: Module 2: Not installed
  Jun 21 10:10:55 b2 kernel: Module 3: Not installed
  Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV
  E/F (4 modules)
  Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States /
North
  America)
 
  [EMAIL PROTECTED] src]# cat /proc/interrupts
CPU0
   0:   17893766  XT-PIC  timer
   1:  2  XT-PIC  keyboard
   2:  0  XT-PIC  cascade
   4:  357411641  XT-PIC  eth0, wanpipe1
   8:  1  XT-PIC  rtc
  10:3381408  XT-PIC  Intel ICH2
  11:  178236906  XT-PIC  wctdm
  14:  50492  XT-PIC  ide0
  15:  0  XT-PIC  ide1
  NMI:  0
  ERR:  0
  [EMAIL PROTECTED] src]# cat /proc/interrupts
CPU0
   0:   17894203  XT-PIC  timer
   1:  2  XT-PIC  keyboard
   2:  0  XT-PIC  cascade
   4:  357419974  XT-PIC  eth0, wanpipe1
   8:  1  XT-PIC  rtc
  10:3381408  XT-PIC  Intel ICH2
  11:  178241275  XT-PIC  wctdm
  14:  50494  XT-PIC  ide0
  15:  0  XT-PIC  ide1
  NMI:  0
  ERR:  0
 
  -- 
  Mike
 

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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Marie]

It depends on how the actual purchase was worded whether or not you
should be able to get a chargeback. I didn't buy from them, so I don't
know.

With some clever legal wording, it is possible to sell something that
the end user considers prepay/future use (like calling card
minutes), but as far as the credit card company etc. are concerned it
was a final sale over and done with like a normal purchase.

This is not to say that the issuing bank is going to give one and they
will just as likely process a chargeback as normal (and later reverse
it as long as someone is still at the other end shooting back the
boilerplate rebuttal). I'd suggest people wait as long as they can
before filing a chargeback -- merchants only get so many days (10 on
my account) to respond before it's automatically settled in the
customer's favor. If you wait as long as you can, there's a better
chance someone won't be sitting there replying.

I used to work for a shady company that sold calling cards
online/phone by credit card. It was a big thing to make sure that the
sales material/call-scripts were worded to make sure that once the
customer took posession of the pin code the transaction was
completed in terms of the credit card company. They often lost
accounts or discontinued programs that customers still had minutes
in, and they were able to escape from chargebacks by sending the fine
print to their bank as their rebuttal to the customer's complaint.

I didn't stay long after finding this out, the pay wasn't worth having
a company like that on my CV.

If you read up on the rumors around Dr. Phil, supposedly it's quite
common (and in some isolated areas still legal) to do a similar thing
with health clubs. Sell one year membership contracts, factor the
contract to someone else, close. The customer is still responsible for
completing their payments to the factor. The customer can't chargeback
payments they already made via credit card because the way the
contract is worded it doesn't matter if the health club is still open
or not.

With lawyers a dime a dozen these days, I can't imagine that LiveVOIP
didn't make sure to put every protection they could in their terms of
service or what have you. Most people don't even read them, or just
don't care what it says. I know nothing about LiveVOIP, so I'm not
trying to suggest that they were indeed shady -- just letting people
know that chargeback rules aren't a fix-all.
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Re: [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK

[Rich Adamson]

If I recall correctly, there was a cvs change recently to support the
new card. My current cvs-head wctdm.c is dated June 22. You probably 
need to pull a fresh copy of the zaptel directory.



 I just purchased two rev I boards and they cannot be recognized at all but a
 revision H (that reports E/F) board works ok.
 
 
 - Original Message - 
 From: John Novack [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, June 22, 2005 6:08 AM
 Subject: Re: [Asterisk-Users] How can you check that egTDM04B
 hardwareinstalled and drivers OK
 
 
  Probably means that your perfectly good motherboard can't see the TDM
 card.
  There are many motherboards that this card doesn't seem to work with,
  Digium doesn't seem willing to address the issue or even acknowledge
  that is the case, and usually answers  try another motherboard rather
  than 'fess up that there is a design problem with the PCI interface and
  correct it.
  PCI 2.2 is a stated requirement, but there is certainly more to the
  story than that.
 
  In addition, when the board CAN be seen, report rev E/F when  the
  silkscreen reads Rev H, someone mentioned there is now a Rev I ( good
  luck getting an exchange ) and Digium 's answer is  if we can see it
  through remote access then there is no reason to replace it, and if we
  can't, try another MB.
 
  Overall, if it works, lucky you, if not, Too bad.
  Hard to support Digium and suggest others purchase such a product.
  Best you look for other interfaces to Asterisk.
 
  John Novack
 
 
 
 
  Angus Comber wrote:
 
   If I try dmesg - no mention of a Wildcard TDM400.
  
   Sorry I am fairly new to Linux.  In Windows I suppose I would run some
   hardware program which came with the card to see if I could manually
   set IRQ's etc.  What should I be looking at now?
  
   Please feel free to point me to a good book or whatever you feel is
   appropriate.  Could the card be faulty?
  
   My motherboard is an Intel D865GLC.
  
   I am running [EMAIL PROTECTED] version 1.0
  
   Angus
  
  
  
  
   - Original Message - From: Mike M
   [EMAIL PROTECTED]
   To: asterisk-users@lists.digium.com
   Sent: Tuesday, June 21, 2005 3:14 PM
   Subject: Re: [Asterisk-Users] How can you check that eg TDM04B
   hardwareinstalled and drivers OK
  
  
   On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote:
  
  
I am struggling to get my TDM04B working.  Just to rule out a
   hardware  problem how can I check
   that the hardware works?  How can I then
check that the drivers are loaded correctly?
   
  
   1. from the linux command line, type 'dmesg' and look for
Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules)
   if you see that, the TDM card is recognized by the OS.
  
  
   Here's what I get on a working system:
  
   [EMAIL PROTECTED] src]# modprobe wctdm
   [EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for 
   device
   01:0a.0
   Jun 21 10:10:55 b2 kernel: Freshmaker version: 71
   Jun 21 10:10:55 b2 kernel: Freshmaker passed register test
   Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO
   Jun 21 10:10:55 b2 kernel: Module 1: Not installed
   Jun 21 10:10:55 b2 kernel: Module 2: Not installed
   Jun 21 10:10:55 b2 kernel: Module 3: Not installed
   Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV
   E/F (4 modules)
   Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States /
 North
   America)
  
   [EMAIL PROTECTED] src]# cat /proc/interrupts
 CPU0
0:   17893766  XT-PIC  timer
1:  2  XT-PIC  keyboard
2:  0  XT-PIC  cascade
4:  357411641  XT-PIC  eth0, wanpipe1
8:  1  XT-PIC  rtc
   10:3381408  XT-PIC  Intel ICH2
   11:  178236906  XT-PIC  wctdm
   14:  50492  XT-PIC  ide0
   15:  0  XT-PIC  ide1
   NMI:  0
   ERR:  0
   [EMAIL PROTECTED] src]# cat /proc/interrupts
 CPU0
0:   17894203  XT-PIC  timer
1:  2  XT-PIC  keyboard
2:  0  XT-PIC  cascade
4:  357419974  XT-PIC  eth0, wanpipe1
8:  1  XT-PIC  rtc
   10:3381408  XT-PIC  Intel ICH2
   11:  178241275  XT-PIC  wctdm
   14:  50494  XT-PIC  ide0
   15:  0  XT-PIC  ide1
   NMI:  0
   ERR:  0
  
   -- 
   Mike
  
 
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Re: [Asterisk-Users] LiveVoip is Bankrupt

[Rich Adamson]

 With lawyers a dime a dozen these days, I can't imagine that LiveVOIP
 didn't make sure to put every protection they could in their terms of
 service or what have you. Most people don't even read them, or just
 don't care what it says. I know nothing about LiveVOIP, so I'm not
 trying to suggest that they were indeed shady -- just letting people
 know that chargeback rules aren't a fix-all.

Since that was an LLC operation and unless management pierced the corp
vail, the LLC has far more liabilities then it does assets so the LLC
is bankrupt. There is a legal pecking order as to who receives payments
after the assets are disposed. As user's of the service, we're on the
bottom of that list and will probably take at least a year or two
to reach closure.


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RE: [Asterisk-Users] Re: Horrible MeetMe performance

[Rob Thomas]

 In my experience, several seconds of delay becomes apparent over time
when
 using an internal clock source.  Seems its a clocking/timer issue.

Yes. Meetme can have horrible issues with timing. This _has_ been fixed.
If you download the CVS Zaptel drivers, use a 2.6 kernel, you can use
RTC support in ztdummy. Change the '#if 0' to '#if 1'.

This is documented at http://www.aussievoip.com.au/wiki-AMP-Zaptel 

--Rob

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Re: [Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.

[Marie]

I'm running 1.0.6 and my registrations to VoipJet survive network outages.

Not sure if it's the old version of Asterisk (before this change you
speak of maybe?) or just some strange variation of network outage.

I often disconnect my connection (beyond the router) just to move
cables and such, or to reboot the modem if my connection gets wonky.

Asterisk always reregisters once the connection is backup quickly
enough that I've never noticed it not doing it.

I haven't rebooted or done any sort of reload on this particular box
in at least 2 months -- and I've had quite a few random network
outages in that time.

Sorry I have no advice, but I thought I'd share that it seems to be
doing what you want by default in my version.
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RE: [Asterisk-Users] LiveVoip is Bankrupt

[Michael Di Martino]

Hey pooch are u ever going to put up the howto's from the Atlanta
asterisk conference? You only said you would. Don't be like LiveVOIP and
follow thru on your word.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Sunday, June 26, 2005 4:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

Yair Hakak wrote:
 well, i can't say i'm surprised. any company whose approach to
 customers is you are all scum trying to cheat us, don't ask
 questions, and we'll help you when we feel like it isn't going to be
 around for a long time.
 

I agree totally.  After seeing some of the issues people were having 
with their customer support (or better, flying off the handle at their 
customers) I decided to stay clear of them.

Survival of the fittest . . .

B.
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[Asterisk-Users] TDM400P-04B fails after reboot

[Andrew]

I have a strange problem with the TDM400P-04B running on Fedora FC2 2.6.10
(and also FC2 2.6.8). I have one in an Acer server and one in a standard
white box. The card will work fine until I reboot. After rebooting the only
way to get the card to work again is to move it to another PCI slot. Or I
can also remove it, boot up, shutdown and then installed it again.

After reboot it will always error when I modprobe wctdm with;
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm

A 'lspci' and 'cat /proc/pci' shows that the card is present (and not in
conflict). I have just tried installing the latest CVS of the Zaptel drivers
but still have the same problem. 

I would greatly appreciate any help as one of my servers is now located in
another state :(. It makes it difficult to reinstall the card or use other
linux distros!

-- Andrew


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RE: [Asterisk-Users] HT-488 vs. SPA-3000?

[John Cianfarani]

I've been looking into this more for a small deployment.
Is it at all possible to put some other line adapter to amplify/increase
signal before it goes into the spa3k?

Something like these? (Found these after a quick google search)
http://www.harriscomm.com/catalog/default.php?cPath=1141_47_167

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=1503item=5784527504;
rd=1ssPageName=WDVW

Would love to know if anyone has tried them.

Thanks
John Cianfarani

---
I had exactly the same experience with the SPA-3000. Too bad too since
it's nice device...if it were 6 db hotter.

I also installed a TDM-400, which was better in a lot of ways but not
perfect. When I rebuild my server I ended up simply call forwarding my
POTS lines to a DID provided by an ITSP. This has been the best as far
as quality is concerned. If my DSL line goes down I simply defeat the
call forwarding on the main line and answer an analog phone for a
while, or call forward to me cell.

Michael

On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote:
  Just want to tap the collective wisdom of this list as to experiences
  pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters...
 
 I've not played with the ht488, but I believe others have posted this
 device does not provide access to the pstn-fxo port. The spa3k does
 provide that access (if you want it).
 
  Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be
  the top of the pick..Any comments and experiences esp. with Asterisk
  compatibility would be great, before I plonk in the bucks.
 
 The spa3k works fine with asterisk as many have posted. However, once
 in awhile it does act a little strange in two different ways:
  1. the spa3k will sometimes interpret some voices as tones which cause
  a little disturbance to any conversation going on. It is sort of like
  the old telephony talk off that existed years ago. Doesn't happen
  all that often and seems to be more sensitive to female voices based
  on my one-year of experience.
  2. sometimes it seems to operate in half-duplex mode, where if you try
  to talk at the same time as the other end is talking, the other end
  won't hear you.
 
 Neither one of those have been all that objectionable to me, but they
 happen and others have posted roughly the same issues. I've not heard
 of anyone that has found a way to minimize those two issues.
 
 The down side of the spa3k right now is that Cisco bought the company
 and there likely won't be much advancement of the code until after the
 ownership (and development efforts) are sorted out by both companies.
 (The same kind of product delays has been seen with their Linksys
 purchase, as well as when other companies are bought/sold.)
 
 Its fairly common knowledge that ex-Cisco folks started Sipura for the
 sole purpose of selling the company for a hugh profit. Their success
 in accomplishing that objective could only be measured in terms of
 producing Sipura products that had at least some acceptance of those
 products by end users. With those previous objectives accomplished,
 how will Cisco handle the Sipura products in the future? (It's any-
 one's guess at this point since Cisco also has at least some track
 record of mismanaging purchased companies for whatever reason.)
 
 From an internal Cisco strategic perspective, they now own the assets
 that can make a major dent in the mass-market end-user voip product
 arena, and hopefully they'll take that in a positive direction.
 
 Given the price of the spa3k, I don't have any issue with purchasing
 more of them right now. Excellent choice for the one-to-three pstn-fxo
 market space.
 
 
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--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

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



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Re: [Asterisk-Users] LiveVoip is Bankrupt

[trixter http://www.0xdecafbad.com]

On Sun, 2005-06-26 at 20:49 -0600, Rich Adamson wrote:
  With lawyers a dime a dozen these days, I can't imagine that LiveVOIP
  didn't make sure to put every protection they could in their terms of
  service or what have you. Most people don't even read them, or just
  don't care what it says. I know nothing about LiveVOIP, so I'm not
  trying to suggest that they were indeed shady -- just letting people
  know that chargeback rules aren't a fix-all.
 
 Since that was an LLC operation and unless management pierced the corp
 vail, the LLC has far more liabilities then it does assets so the LLC
 is bankrupt. There is a legal pecking order as to who receives payments
 after the assets are disposed. As user's of the service, we're on the
 bottom of that list and will probably take at least a year or two
 to reach closure.
 

LLC/Corporations do not protect officers of the company if the officers,
through official job duties, commit crimes.  Taking money for services
you know you cant provide.  Its prima facia if you sell below cost and
cant prove that you thought you have VC money or something else to
offset that 'promotional' period and then file bankrupcy.  This is to
prevent someone from basically doing a ponzi scheme, where people late
in the game are paying for the people today, eventually the bubble
bursts and the late comers are left holding the bag.  While this is
specific to US law, livevoip in this case was a US based company so that
applies.  This may not apply to other companies doing basically the same
thing in other jurisdictions.  And I dont know that they were doing
this, but I am certain they didnt decide to file bankrupcy and file the
same day, there had to be a period when they started to file but kept
accepting new customers knowing those customers werent going to get what
they paid for.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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