Hai Cool Buddies

Good evening to everybody !!!!

Thanks to all in advance .

I have Doubt in meetme

How can i configure below

COnfrance Room NUmber : 65896

COnfrance ID : 1555 ( Project 1 )
Confrance ID : 1885 ( Project 2 )

Advice please

I am trying to configure Asterisk for BSNL, india network.
I have successfully configured it for outgoing calls.

Right now trunk is able to make any call to outside world and receive
any call from outside world. As when it makes any call to outside
world then it sends proper signals for ringing, disconnection, etc.
So outside world number will receive the call and when the call is
disconnected it notifies properly to outside number about call
disconnection.
But when any outside number make any call to trunk then it receives
the call properly but when the call is disconnected by outside number
then it fails to identify disconnection tone. So it shows as if the
line is connected even after disconnection from outside world call.

Outside number receives busy signal instead of disconnect tone.


My system information is as follows:

OS and components:

CentOS 4.5
Asterisk 1.4.17
Zaptel 1.4.7.1
Libpri 1.4.3

extensions.conf
[globals]
OUTBOUNDTRUNK=Zap/4

[incoming]
; incoming calls from FXO
exten => s,1,Dial(Zap/1)
exten => s,n,Hangup()

[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Playback(hello-world) ;For testing the extension, it
will ;play "hello world"
exten => 500,n,Hangup()

[outbound-dialing]
;Outbound dialing
exten => _X.,1,Verbose(1|Outside number|${EXTEN})
exten => _X.,n,Dial(${OUTBOUNDTRUNK}/${EXTEN})

[phones]
include => internal
include => outbound-dialing

zaptel.conf file:

fxsks=4
fxoks=1
loadzone=uk
defaultzone=uk

# /sbin/ztcfg -vv this linux command gives following output:
Zaptel Version: 1.4.7.1
Echo Canceller: MG2
Configuration
======================

Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

2 channels to configure.

zapata.conf file looks like:
[trunkgroups]
; Define

[channels]
;hardware channels
;default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
cidstart=ring
cid_signalling=dtmf

;define channels
signalling=fxo_ks ;Use FXO signaling for FXS channel
context=phones
channel => 1

signalling=fxs_ks ;Use FXS signaling for FXO channel
busydetect=yes
busycount=6
busypattern=0,750
hanguponpolarityswitch=yes
callprogress=no
progzone=uk
ringtimeout=8000
context=incoming
channel => 4

I found following disconnection tone string:

400@-30,400@-30;4(.8/.8/1+2)

on this forum for bsnl india network.
But I don't know where to put this string in configuration files. And
whether this string will be helpful for me or not.

Any sort of help will be appreciated.

Thanks in advance

Prashant

In which case, is there a way with which one can make out the
difference between a clone that'd support the 3.3V bus and one that
would not ? Physical inspection ? Supplier / card-model-P/N checklist
etc ?

--- In asterisk-india@yahoogroups.com, Henry Cobb <henry_cobb@...>
wrote:
>
> --- gautam chekuri <gautam.chekuri@...> wrote:
> > Hi,
> >
> > Can anyone tell if there are any stores selling Digium X100P or an
> > Atcom AX-100p cards in Hyderabad ?
>
> Be very careful with the X-100P clone cards.
>
> Many are sold with the 3.3v PCI notch cut, but don't really support
the 3.3v
> PCI bus so your system will at best not boot at all when they are
installed.
>
> -HJC
>
> http://www.io.com/~hcobb
>
>
>
______________________________________________________________________
______________
> Looking for last minute shopping deals?
> Find them fast with Yahoo! Search.
http://tools.search.yahoo.com/newsearch/category.php?category=shopping
>

Hello group,

This is my first message here. Hope someone can give me some pointers.
During a call I intend to pause the call and play custom messages to the
caller and then return him back to the call and would like it at some
pre defined interval. How could this be done. Please guide.
I see Timeout(absolute) variable for such interrupt. But I dont want to
hangup the call and would like to return the caller back to it after the
message.

Thanks
Saurabh

if you want it i can arrange the cards for you.


On Dec 9, 2007 7:27 PM, gautam chekuri <gautam.chekuri@...> wrote:

>   Hi,
>
> Can anyone tell if there are any stores selling Digium X100P or an
> Atcom AX-100p cards in Hyderabad ?
>
> Thanks,
> Gautam
>
>
>


[Non-text portions of this message have been removed]

--- gautam chekuri <gautam.chekuri@...> wrote:
> Hi,
>
> Can anyone tell if there are any stores selling Digium X100P or an
> Atcom AX-100p cards in Hyderabad ?

Be very careful with the X-100P clone cards.

Many are sold with the 3.3v PCI notch cut, but don't really support the 3.3v
PCI bus so your system will at best not boot at all when they are installed.

-HJC

http://www.io.com/~hcobb


      
________________________________________________________________________________\
____
Looking for last minute shopping deals?
Find them fast with Yahoo! Search. 
http://tools.search.yahoo.com/newsearch/category.php?category=shopping

Hi,

Can anyone tell if there are any stores selling Digium X100P or an
Atcom AX-100p cards in Hyderabad ?

Thanks,
Gautam

we charge 75,000 as consultancy fees for this. we can setup this things at
the expenses of the hardware

On 9/11/07, Amit [Netahoy] Agarwal <netahoy@...> wrote:
>
>   Was wondering if anyone has setup an Asterisk@Home at home for making
> a small epabx using Airtel+Mtnl+Vonage/Skype/Gizmo
> I would really appreciate if someone can guide me in archiving the above.
>
> Would a P3 with 512MB machine do good enuf. I believe a Digium type of
> card would also be required but what type/model?
>
> Thanks...
> nH
>
>
>


[Non-text portions of this message have been removed]

Was wondering if anyone has setup an Asterisk@Home at home for making
a small epabx using Airtel+Mtnl+Vonage/Skype/Gizmo
I would really appreciate if someone can guide me in archiving the above.

Would a P3 with 512MB machine do good enuf. I believe a Digium type of
card would also be required but what type/model?

Thanks...
nH

Job Description

Responsibilities:

To provide expert advice on the capabilities and configuration of the
Asterisk PBX
To provide technical lead for all Asterisk aspects of a major development.
To assist in the design of a high availability Asterisk deployment
To advise on the best development environment and tools to for
Linux/Asterisk development.
To advise which add-ons to use in conjunction with Asterisk
To advise the best way in which to integrate a bespoke application
with an Asterisk PBX.

Essential:

3 + yrs exp in a Linux networking/application development environment.
3+ yrs exp of deploying/developing Asterisk based solutions.
Experience with telecommunications technology and the Asterisk PBX

Experience of all the following:

AGI
AIX
SIP
Manager API
VoIP Networking
Desirable

8+ yrs exp as software engineering
Asterisk clustering and high availability solutions
Proficient in the C programming language as well as Linux-based
development tools e.g.
Eclipse.
Experience in formal application development project management.
Ability to communicate effectively with customers and colleagues.
Must be a self-starter, well organized and operate with minimal
supervision.

Outline of Routine Tasks/Duties/Responsibilities

Working with the Next Gen team to produce a pilot system. The purpose
of this role is to provide the technical lead for all Asterisk aspects
of the Next Generation ICCS.

Contact :: nikhil_kesari@...,nikhil.grk@...

Hi all,
I am quite new to Asterisk and facing an issue in configuring my X100P
card with Asterix, it would be great if I can draw some help from this
community !
Following are the details.

I am trying to configure X100P with my Airtel connection.

I am running
Zaptel 1.2.18
Asterisk 1.2.20
Libpri 1.2.4(i think)

My zaptel.conf looks like

fxsks=1
loadzone = uk
defaultzone = uk

Zapata.conf looks like
[channels]
signalling=fxs_ks
context=incoming
channel=>1 ;X100P

extentions.conf looks like
[incoming]
Exten => s,1,Wait(2)
Exten => s,2,Record(/tmp/asterisk-recording:gsm)
Exten => s,3,Wait(2)
Exten => s,4,Playback (/tmp/asterisk-recording)
Exten => s,5,wait(2)
Exten => s,6,Hangup

Now the problem is that, when I call to Asterisk, I always get busy
tone from remote phone. If I detach the line from the card I get
ringing tone of course so looks like the card is working but does not
start from hang up mode.

Asterisk Debug info:-
cat /proc/zaptel/*
results in
Span 1: WCFXO/0 "Wildcard X100P Board 1"

1 WCFXO/0/0 FXSKS (In use)

*CLI> pri show span 1
No PRI running on span 1
Thanks in advance,
Lalit

for two weeks now i av bn having problems linking my two asterisks.

   i used polycom300 on both ends of my disjointed network. 172.16.50.0 and
172.16.100.0
   .the network is linked together via win2003 server router.
   am aware that polycom does not support IAX2 trunking.i av tried SIP trunking
it is still not working.i want you to put me right on how to go about it.
   thanks

   PLEASE HELP MEEEEEE


---------------------------------
Now that's room service! Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.

[Non-text portions of this message have been removed]

My company would be interested in the project. Can you send us the details
and how we go about it?

-----Original Message-----
From: asterisk-india@yahoogroups.com [mailto:asterisk-india@yahoogroups.com]
On Behalf Of acehunky2
Sent: January 17, 2007 6:18 PM
To: asterisk-india@yahoogroups.com
Subject: [asterisk-india] Re: call distribution and Voice recording
usingAstrisk-Need Help

Rajan,

--- In asterisk-india@yahoogroups.com, "RANJAN.G" <ranjan_babu@...> wrote:
>
> Hi ,
>
> We are planning to start anew project .Need input and any body
interested they can join in this project. Need your valuable input.
>
> 1. Setup call distribution to agent / support persons ( Like CISCO
IPCC) by Asterisk .
> 2. Recording all calls using asterisk
> 3. Recording   the all call received by cisco Callmanager by Astrisk .

We can provide commercial support for the setup, please contact us off
the list for the same.

Thanks & Regards,
Mitul Limbani,
Enterux Solutions,
The Enterprise Linux Company (R),
www.enterux.com




Yahoo! Groups Links

Rajan,

--- In asterisk-india@yahoogroups.com, "RANJAN.G" <ranjan_babu@...> wrote:
>
> Hi ,
>
> We are planning to start anew project .Need input and any body
interested they can join in this project. Need your valuable input.
>
> 1. Setup call distribution to agent / support persons ( Like CISCO
IPCC) by Asterisk .
> 2. Recording all calls using asterisk
> 3. Recording   the all call received by cisco Callmanager by Astrisk .

We can provide commercial support for the setup, please contact us off
the list for the same.

Thanks & Regards,
Mitul Limbani,
Enterux Solutions,
The Enterprise Linux Company (R),
www.enterux.com

I was wondering if anybody provides services to setup asterisk/SPA3K
configuration in Mumbai, for residential purposes ?
Please contact,

Hello,



I am looking to set up a call center in Hyderabad based on an
Asterisk's (www.voicerd.com <http://www.voicerd.com/> ) PBX system.  I
need an experienced Asterisk consultant to drive the project in
Hyderabad and interface with another Asterisk company here in the US.
  If there is a good fit with our team we would like to keep the person
on staff to serve as a IT director role.  This would be a tremendous
learning opportunity for whoever is involved.  If you are interested
or know someone that would be, please contact me directly.


Best regards,



Avi



_________________________________________________________
Avinash G. Samudrala
Associate
AEA Investors LLC | 65 E. 55 Street, NYC, NY 10022
Phone: 212-702-0544 | Fax: 212-702-1342
asamudrala@... <mailto:asamudrala@...>  |
http://www.aeainvestors.com <http://www.aeainvestors.com/>

Hi ,

We are planning to start anew project .Need input and any body interested they
can join in this project. Need your valuable input.

1. Setup call distribution to agent / support persons ( Like CISCO IPCC) by
Asterisk .
2. Recording all calls using asterisk
3. Recording   the all call received by cisco Callmanager by Astrisk .

Regards,
Ranjan



[Non-text portions of this message have been removed]

On Nov 6, 2006, at 2:12 PM, acehunky2 wrote:

>
> We did that last year and at the same time had met Vaibhav too. Hey
> vaibhav are you going to be around this time ?

FOSS.IN managers don't get to get out the venue between 24th and 26th
Nov.. ;)

Depending on the situation at that time, I might make it to the BOF.

--
VaibhaV Sharma
http://vsharma.net





[Non-text portions of this message have been removed]

On Mon, 2006-11-06 at 19:12 +0000, acehunky2 wrote:

> All those who are in for Asterisk BOF @ Foss.In please stand up ;)

+1

-indra


--
Indranil Das Gupta
Chief Technology Architect

L2C2 Technologies
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
[ Innovations that make a Difference ]
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

Email : indradg@...
Phone : (0)-98300-20971
WWW   : http://www.l2c2.org

Nadir,

Do contact me off the list :)

Mitul
--- In asterisk-india@yahoogroups.com, Nadir Nizamuddin <nadirn@...>
wrote:
>
> Hello,
>
> I am the CEO and founder of a new startup company located in San
Francisco. I am looking to open a development center in India (exact
city yet to be decided). We are about to be funded by one of the best
VCs around. The product is a next generation enterprise class cellular
switch based upon Asterisk.
>
> Initially I am looking for couple of very senior leaders with
following qualifications:
> - able to manage and direct team of 10-20 engineers
> - solid background in cellular and pbx designs
> - atleast 2 years in development in Asterisk.
>
> Also looking for Asterisk engineers.
>
> Would be interested in hearing back from potential interested
candidates.
>
> Thanks
> Nadir
>
>
>
> [Non-text portions of this message have been removed]
>

Hey Dinesh,

I think we have to meetup :)
I shall be there in Foss.in and shall be taking a Asterisk BOF, I am
sure we shall be able to manage a decent crowd.

We did that last year and at the same time had met Vaibhav too. Hey
vaibhav are you going to be around this time ?

All those who are in for Asterisk BOF @ Foss.In please stand up ;)

Mitul
--- In asterisk-india@yahoogroups.com, "dbirla" <dinesh@...> wrote:
>
> How i would love to make it:).
>
> If i can i will surely visit, but i have to check up on my
> schedule.
>
> Thanks Vaibhav:)
>
> Dinesh.
>
> --- In asterisk-india@yahoogroups.com, "VaibhaV Sharma"
> <vaibhav@> wrote:
> >
> > > Hello All,
> > >
> >
> > > I will be in bangalore on the 22nd(Wednesday) of November 2006.
> >
> > You should consider attending FOSS.IN - http://foss.in - (Nov 24-
> 26th)
> >
> > --
> > VaibhaV
> > http://vsharma.net
> >
>

Count me in also and let me know the venue.I will have to brush up my reading of
VoIP and learn how to use it better.The newbie workshop looks to be a perfect
solution.

best regards,

Parvez

M/s Technoclean Solutions and Services,
Bangalore

VaibhaV Sharma <vaibhav@...> wrote: > Hello All,
>

> I will be in bangalore on the 22nd(Wednesday) of November 2006.

You should consider attending FOSS.IN - http://foss.in - (Nov 24-26th)

--
VaibhaV
http://vsharma.net





Yahoo! Groups Links







---------------------------------
  Check out the New Yahoo! Mail - Fire up a more powerful email and get things
done faster.

[Non-text portions of this message have been removed]

How i would love to make it:).

If i can i will surely visit, but i have to check up on my
schedule.

Thanks Vaibhav:)

Dinesh.

--- In asterisk-india@yahoogroups.com, "VaibhaV Sharma"
<vaibhav@...> wrote:
>
> > Hello All,
> >
>
> > I will be in bangalore on the 22nd(Wednesday) of November 2006.
>
> You should consider attending FOSS.IN - http://foss.in - (Nov 24-
26th)
>
> --
> VaibhaV
> http://vsharma.net
>

Hello,

I am the CEO and founder of a new startup company located in San Francisco. I am
looking to open a development center in India (exact city yet to be decided). We
are about to be funded by one of the best VCs around. The product is a next
generation enterprise class cellular switch based upon Asterisk.

Initially I am looking for couple of very senior leaders with following
qualifications:
- able to manage and direct team of 10-20 engineers
- solid background in cellular and pbx designs
- atleast 2 years in development in Asterisk.

Also looking for Asterisk engineers.

Would be interested in hearing back from potential interested candidates.

Thanks
Nadir



[Non-text portions of this message have been removed]

> Hello All,
>

> I will be in bangalore on the 22nd(Wednesday) of November 2006.

You should consider attending FOSS.IN - http://foss.in - (Nov 24-26th)

--
VaibhaV
http://vsharma.net

Hello All,

I will be comming in india shortly(to bangalore actually) and
intrested to meet a few of the fellow linux and asterisk users.

I can give some asterisk tutorials for free for newbies if you guys
are intrested and can bring in some linux cds from Singapore where
the bandwidth is there.

Please let me know if you would like some.  I would like to post on
the linux user groups as well, but i thought this would be a better
place.

I will be in bangalore on the 22nd(Wednesday) of November 2006.

It doesn't have to be a formal meeting, but can share with you some
of the problems, pitfalls in asterisk building.  Ideally I would
like to meet all asterisk intrested people and discuss on some
issues.  Would also like to touch up intergration of asterisk with
Cisco Callmanager 4.0, SIP URI based dialling to Call Manager, Call
back services and some neat tricks using asterisk.

regards,

Dinesh Birlasekaran
Network Engineer,
ComIT, Institute of Molecular and Cell Biology
61 Biopolis Drive, Singapore 138673
HP : 92962676 DID : 65869804 Fax : 67791117
Email : dinesh@...
WWW: www.imcb.a-star.edu.sg

I try to but i think its rediff\'s issue. I have tried to configure it but
couldnt.Prashant

[Non-text portions of this message have been removed]

On Sep 15, 2006, at 2:02 PM, Prashant Bhalesain wrote:

> Dear, Then u will have to come own with ur own AGI script (either
> perl or php) . Either have a look at a2billing\'s phpagi or Astbill
> \'s perl AGI.User Agent|-- AGI if u need |Asterisk|-- Agi scitpt --
> > radius- free radius|-- PSTNasterisk-india@yahoogroups.com
> wroteHi Dear,I wants fallowing wayUser Agent|asterisk-----
> radiusclient-------freeradius| |PSTN Network Database( Mysql 5.0) /
> Billing | PSTN Users( User Agent)User agent send the auth request
> to asterisk after that asterisk authenticate the user from the
> database. After that user agent make a call on PSTN network now
> start the accounting and when stop the call then stop the
> accounting & generate the CDR.ThanksA. K.Kishor Gavali
> <kishor.gavali@...> wrote: Why u not try freepbx and
> a2billing, that is having accounting facilityRegardsKishor
> GavaliSr. System AdministratorAtlogix Software Solutions Pvt.
> Ltdwww.atlogix.com_____ From: asterisk-india@yahoogroups.com
> [mailto:asterisk-india@yahoogroups.com]On Behalf Of
> asteriskserSent: Wednesday, September 13, 2006 8:24 PMTo: asterisk-
> india@...: [asterisk-india] how to Asterisk
> +FreeRadiusHi All,I want to configure the Asterisk with freeradius.
> In following way.1. I wants AAA.2. I wants authentication from the
> database using freeradius by sipuser ( Using Asterisk as sip proxy
> server).3. Authorization.4. I wants Accounting using freeradius
> ( radius database) and asterisk.I appreciate any help.RegardsA. K.
> [Non-text portions of this message have been
> removed]---------------------------------Get your own web address
> for just $1.99/1st yr. We\'ll help. Yahoo! Small Business.[Non-text
> portions of this message have been removed]


Prashant:

Your emails are barely readable. Can you make sure that you indent
your emails properly in your email client?

Thanks,

--
VaibhaV Sharma
http://vsharma.net





[Non-text portions of this message have been removed]

Dear, Then u will have to come own with ur own AGI script (either perl or php) .
Either have a look at a2billing\'s phpagi or Astbill\'s perl AGI.User Agent|--
AGI if u need |Asterisk|-- Agi scitpt --> radius- free radius|--
PSTNasterisk-india@yahoogroups.com wroteHi Dear,I wants fallowing wayUser
Agent|asterisk-----radiusclient-------freeradius| |PSTN Network Database( Mysql
5.0) / Billing | PSTN Users( User Agent)User agent send the auth request to
asterisk after that asterisk authenticate the user from the database. After that
user agent make a call on PSTN network now start the accounting and when stop
the call then stop the accounting & generate the CDR.ThanksA. K.Kishor
Gavali <kishor.gavali@...> wrote: Why u not try freepbx and
a2billing, that is having accounting facilityRegardsKishor GavaliSr. System
AdministratorAtlogix Software Solutions Pvt. Ltdwww.atlogix.com_____ From:
asterisk-india@yahoogroups.com [mailto:asterisk-india@yahoogroups.com]On Behalf
Of asteriskserSent: Wednesday, September 13, 2006 8:24 PMTo:
asterisk-india@...: [asterisk-india] how to Asterisk
+FreeRadiusHi All,I want to configure the Asterisk with freeradius. In following
way.1. I wants AAA.2. I wants authentication from the database using freeradius
by sipuser ( Using Asterisk as sip proxy server).3. Authorization.4. I wants
Accounting using freeradius( radius database) and asterisk.I appreciate any
help.RegardsA. K. [Non-text portions of this message have been
removed]---------------------------------Get your own web address for just
$1.99/1st yr. We\'ll help. Yahoo! Small Business.[Non-text portions of this
message have been removed]

[Non-text portions of this message have been removed]

Hi Dear,

   I wants fallowing way

   User Agent
            |
   asterisk-----radiusclient-------freeradius
          |                                            |
   PSTN Network             Database(  Mysql 5.0) / Billing
          |
   PSTN Users( User Agent)



   User agent send the auth request to asterisk after that asterisk authenticate 
the user from the database.  After that user agent make a call on PSTN  network
now start the accounting and when stop the call then stop the accounting  &
generate the CDR.


   Thanks
   A. K.



Kishor Gavali <kishor.gavali@...> wrote:
Why u not try freepbx and a2billing, that is having accounting facility

   Regards

   Kishor Gavali

   Sr. System Administrator

   Atlogix Software Solutions Pvt. Ltd

   www.atlogix.com

   _____

   From: asterisk-india@yahoogroups.com [mailto:asterisk-india@yahoogroups.com]
   On Behalf Of asteriskser
   Sent: Wednesday, September 13, 2006 8:24 PM
   To: asterisk-india@yahoogroups.com
   Subject: [asterisk-india] how to Asterisk +FreeRadius

   Hi All,

   I want to configure the Asterisk with freeradius. In following way.

   1. I wants AAA.
   2. I wants authentication from the database using freeradius by sip
   user ( Using Asterisk as sip proxy server).
   3. Authorization.
   4. I wants Accounting using freeradius( radius database) and asterisk.

   I appreciate any help.

   Regards
   A. K.

   [Non-text portions of this message have been removed]






---------------------------------
Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small
Business.

[Non-text portions of this message have been removed]