asterisk users - Jul 2003 - Need help

I do part time consulting work. I need to setup an asterisk system to
allow me to record both inbound and outbound calls to clients. I have one
client that is just a PITA. The client has changed their mind three times
so far and we are at step one.

I have a spare slackware box and a seperate phone line for the consulting
work. I have MCI Neighorhood as my carrier.

What I need to know is:

1. What hardware is needed to record both inbound and outbound call with
the disclaimer added to the phone call. The hardward should be as
transpartent as possible. If I can get this to work I might setup it up on
the main line also but with out the recording capibilites.

2. Any special software patchs to apply to the base source.

3. Any tips on setting up the config files. For the recording and blocking
numbers that do not have proper caller I like telemarketers. The blocking
should be done before the call is passed thur.

4. Need to go as cheep as posible but still have the needed capibilites.

5. I know nothing about telcom except that I pick up the phone, get a dial
tone, and dial.


If any of the above can or cannot be done please let me know of any
alternative, if it possible.

Thank you,
Donn
-- 
dpike2@cox.no.spam.net (remove no spam)
Virgina Resident/4530 S@H WU

hi!
I wanted to ask if someone ever got the error "flexible rate not heavily
tested"
I am not able to dial from PSTN to iaxphones(on which agents are
logged in)...I have been successfully running this for some time
now..but today all of a sudden i got this error and I can`t get
connected to the agent...
Can anyone help,.....
plz

hi!
I wanted to ask if someone ever got the error "flexible rate not heavily
tested"
I am not able to dial from PSTN to iaxphones(on which agents are
logged in)...I have been successfully running this for some time
now..but today all of a sudden i got this error and I can`t get
connected to the agent...
Can anyone help,.....
plz

Sounds like a message from mpg123,

Is your asterisk crashing when this happens or are you just giving the
wrong input files for moh ?

Zoa.

amna saleem wrote:
>hi!
>I wanted to ask if someone ever got the error "flexible rate not
heavily tested"
>I am not able to dial from PSTN to iaxphones(on which agents are
>logged in)...I have been successfully running this for some time
>now..but today all of a sudden i got this error and I can`t get
>connected to the agent...
>Can anyone help,.....
>plz
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
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>
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This is a DTMF issue,

You must adjust this on the especific channel conf file.

For example, ia sip phone cannot dial any number during an active call, 
you must see sip.conf and the config in your hardphone or softphone.

Ismael.






"Tim Touhsaent" <touhsatj@hotmail.com> 
Enviado por: asterisk-users-bounces@lists.digium.com
04/29/2005 03:16 PM
Por favor, responda a
Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>


Para

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<asterisk-users@lists.digium.com>
cc





Asunto
[Asterisk-Users] need help






I am having an issue with the asterisk system not responding to dialed
numbers during an active
call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone
config? and worse I
don't even know what Keywords to search for.

Tim
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Is this a SIP phone?

I had to upgrade the firmware on my SIP phones to alleviate this. It seems that
the phone would actually disable it's own keypad after dialling.

Ian
>>> touhsatj@hotmail.com 29/04/2005 09:16 >>>
I am having an issue with the asterisk system not responding to dialed
numbers during an active
call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone
config? and worse I
don't even know what Keywords to search for.

Tim
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Asterisk-Users@lists.digium.com 
http://lists.digium.com/mailman/listinfo/asterisk-users 
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

> I just got Mediatrix 1204 from ebay,  but it is missing CD that conmtain
the
> software and drivers, I am wondering if anybody knows where I could
> downloaded from. 
Have you tried http://www.mediatrix.com/ ?

How do metratrix board wirk with asterisk? Any scenarios? 

|-----Original Message-----
|From: asterisk-users-bounces@lists.digium.com 
|[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
|Time Bandit
|Sent: Martes, 03 de Mayo de 2005 09:11 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Mediatrix 1204 Help
|
|> I just got Mediatrix 1204 from ebay,  but it is missing CD that 
|> conmtain the software and drivers, I am wondering if anybody knows 
|> where I could downloaded from.
|Have you tried http://www.mediatrix.com/ ?
|_______________________________________________
|Asterisk-Users mailing list
|Asterisk-Users@lists.digium.com
|http://lists.digium.com/mailman/listinfo/asterisk-users
|To UNSUBSCRIBE or update options visit:
|   http://lists.digium.com/mailman/listinfo/asterisk-users
|
|

Iam new in asterisk user, can helpme to install asterisk for 
applications callingcard ?
current ialready install asterisk with mysql db and already 
connected, and next i dont know how to create as calling card applications, 
adn how other how to setup using SIP sofphone if iam using expresstalk 
softphone, if i want use as callingcard applications.
pls help
     
thanks 
Dirgan
 
		
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 Yahoo! Asia presents Meetic - where millions of singles gather 
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Have you settled on a calling card application yet?  There are a host of 
different options.  I, of course, recommend astpp. :-)  The wiki will 
have much of the info you will need.

Darren Wiebe
darren@aleph-com.net

Dirgan Putra wrote:
>Iam new in asterisk user, can helpme to install asterisk for 
>applications callingcard ?
>current ialready install asterisk with mysql db and already 
>connected, and next i dont know how to create as calling card applications, 
>adn how other how to setup using SIP sofphone if iam using expresstalk 
>softphone, if i want use as callingcard applications.
>pls help
>     
>thanks 
>Dirgan
>
> ------------------------------------------------------------------------
> Meet your soulmate!
> *Yahoo! Asia presents Meetic* 
>
<http://us.rd.yahoo.com/prm/pers/mt/yma_sg/tgl/*http://asia.yahoo.com/meetic>
> - where millions of singles gather
>
>------------------------------------------------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

-- 
Darren Wiebe
darren@aleph-com.net
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

hi Darren
 
 i see ur webpage , about  asteriskcdrdb in ur astpp
 si same with AMP ?, i have install Asterisk Management portal, or any
differents ?, need in ur astpp instal AMP ? or u have web site configurations
for extension, sip, iax2 ?
 
 thanks
 Dirgan
 
Dirgan Putra wrote:
>Iam new in asterisk user, can helpme to install asterisk for 
>applications callingcard ?
>current ialready install asterisk with mysql db and already 
>connected, and next i dont know how to create as calling card applications, 
>adn how other how to setup using SIP sofphone if iam using expresstalk 
>softphone, if i want use as callingcard applications.
>pls help
>     
>thanks 
>Dirgan
>
> ------------------------------------------------------------------------
> Meet your soulmate!
> *Yahoo! Asia presents Meetic* 
>  
> - where millions of singles gather
>
>------------------------------------------------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

-- 
Darren Wiebe
darren@aleph-com.net
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp




		
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hi All
   
  need help, iam installing areskiCC and have a problem
  after that create extension for calling card and after dial 
   
  exten => 17000,3,DeadAgi,a2billing.php
   
  i see messages : a2billing.php no such file in directory, i tired copy that
file
  that file aready copy in agi-bin.
   
  any body have experience in same problem, i need a suggestion to solve this
problme
   
  thanks
   
  Putra

		
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Have you checked the permissions on the file?  Is it executable?

Garth


Dirgan Putra wrote:
> hi All
>  
> need help, iam installing areskiCC and have a problem
> after that create extension for calling card and after dial
>  
> exten => 17000,3,DeadAgi,a2billing.php
>  
> i see messages : a2billing.php no such file in directory, i tired copy 
> that file
> that file aready copy in agi-bin.
>  
> any body have experience in same problem, i need a suggestion to solve 
> this problme
>  
> thanks
>  
> Putra
>
> ------------------------------------------------------------------------
> Do you Yahoo!?
> Yahoo! Movies 
>
<http://sg.rd.yahoo.com/mail/sg/footer/def/*http://sg.movies.yahoo.com>
> - Search movie info and celeb profiles and photos.
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

>   need help, iam installing areskiCC and have a problem
>   after that create extension for calling card and after dial
>
>   exten => 17000,3,DeadAgi,a2billing.php
>
>   i see messages : a2billing.php no such file in directory, i tired copy
> that file that file aready copy in agi-bin.
Try chmod 755 /var/lib/asterisk/agi-bin/a2billing.php

benchev

hi All 
   
  I need your help , for used Digium Card TE405P, for setting this Board AS E1
ISDN PRI.
   
  1 .Current for make sure my config its rights or no I inform my configurations
in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the
Jumper.
   
   
  2. I Want To seeting E1 in ASterisk/PC  Back To Back    To Cisco E1 AS5300 Use
ISDN Signaling,
   
  my configutration :
   
  softphone --- > ASterisk TE405P  E1 --->  E1 AS5300
   
  I need your help me to rights configurations Zapata.conf and zaptel.conf
   
  FYI iam already load :
  modpobe zaptel
  modprobe wct4xxp t1e1override=15 debug=1
   
  and current /etc/zaptel.conf
  span=1,0,0,ccs,hdb3,crc4,yellow
  span=2,0,0,ccs,hdb3,crc4,yellow
  span=3,0,0,ccs,hdb3,crc4,yellow
  span=4,0,0,ccs,hdb3,crc4,yellow
   
  current i have problem error message in cisco as5300 after make a call, if
used debug isdn q931
   
  in cisco AS5300 config :
   
  isdn siwtch-type primary-net5
   
  controller e1 0
  framming crc4
  linecode hdb3
  pri-group 1-31
   
   
  interface serial0:15
  isdn switch-type primary-net5
  isdn incomming-voice modem
  isdn T310 60000
   
   
  dialpeer,.....blah,...blahh
   
   
  thanks for ur help
   
  Dirgan
   

		
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hello,

I have to test asterisk/gnugk is their somebody, sur
cette putain de liste, with a h323 terminal ?

harry 




	

	
		
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Folks. Thanks for running such a good forum, Just to let you know that I
have just started using Asterisk and let me tell you, I am liking this every
day... I need your help in fixing couple of issues

I have these DID's coming to my asterisk box and I have a dial plan and all
I want is that it should ring on my cell phone for 20 seconds and if I don't
pickup it would then ring on my home phone. It does the ringing and goes to
the next phone in 20 seconds and every thing works fine but If I pickup the
first phone (cell phone), there is NO voice at all. If I let it go to the
second phone, it works fine and I can talk crystal clear.

The second issue is that I don't get a ringback when I dial my DID which is
usually a tone which caller should get so he would know that the bell is
ringing on the other side.

I would really appreciate if you someone can help me here

Thanks

SM