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]

When u install trixbox/asterisk@home it also installs a2billing. Its part of
trixbox/asteriskathome.Also u can separately install a2billing which is quite
easy. The main part of a2billing is phpagi script and rate engine.Once you
install, you will have to build the rate engine. Let me know if u need
help.Prashantasterisk-india@yahoogroups.com wroteHi Kishor,On this note do have
experince in implementing a2billing on TRIXBOX?Just wanted to have a know how or
any help links if possible.thanksOn 13/09/06, 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
<asterisk-india%40yahoogroups.com>[mailto:>
asterisk-india@yahoogroups.com <asterisk-india%40yahoogroups.com>]> On
Behalf Of asteriskser> Sent: Wednesday, September 13, 2006 8:24 PM> To:
asterisk-india@yahoogroups.com <asterisk-india%40yahoogroups.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]>> >[Non-text
portions of this message have been removed]

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

Hi Kishor,

On this note do have experince in implementing a2billing on TRIXBOX?

Just wanted to have a know how or any help links if possible.

thanks



On 13/09/06, 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
<asterisk-india%40yahoogroups.com>[mailto:
> asterisk-india@yahoogroups.com <asterisk-india%40yahoogroups.com>]
> On Behalf Of asteriskser
> Sent: Wednesday, September 13, 2006 8:24 PM
> To: asterisk-india@yahoogroups.com <asterisk-india%40yahoogroups.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]
>
>
>


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

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]

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.

Mitul,

Thanks for the pointer, i will review the link.

Thanks.
- Naren



--- acehunky2 <acehunky2@...> wrote:

> Hi Naren,
>
> --- In asterisk-india@yahoogroups.com, "npasuparthy"
> <npasuparthy@...>
> wrote:
> >
> > Hi,
> >
> > I am naren and live in bangalore. I am itnerested
> to join the asterisk
> > group in bangalore. Trying to understand asterisk
> usage and support in
> > India.
> >
>
> Welcome to the list.
> you may also want to signup on
> http://asterisk.pbx.in/
>
> We have put up quite a few Asterisk Howto's for the
> starters, also if
> you feel like, you can contribute your howto on the
> site.
>
> Looking forward to have you on the Asterisk
> Community in India.
>
> Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions,
> The Enterprise Linux Company (R),
> www.enterux.com
>
>
>
>
>


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around
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Hi Naren,

--- In asterisk-india@yahoogroups.com, "npasuparthy" <npasuparthy@...>
wrote:
>
> Hi,
>
> I am naren and live in bangalore. I am itnerested to join the asterisk
> group in bangalore. Trying to understand asterisk usage and support in
> India.
>

Welcome to the list.
you may also want to signup on http://asterisk.pbx.in/

We have put up quite a few Asterisk Howto's for the starters, also if
you feel like, you can contribute your howto on the site.

Looking forward to have you on the Asterisk Community in India.

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

Hi,

I am naren and live in bangalore. I am itnerested to join the asterisk
group in bangalore. Trying to understand asterisk usage and support in
India.

Please advice !

- Naren

Hi friends,

Here I am posting my problem. I am getting this problem since 8 days. I have
studied documentation and looked previous posts in forums. But, I am unable to
solve this problem. Please show me a solution. I am from Andhrapradesh, India.

We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I  have
connected my PSTN (BSNL) line directly to first port. I am making outgoing 
calls and receiving incoming calls successfully through my Asterisk. The 
problem is: When I am receiving a call from outside (PSTN-Eg. Mobile), I am not
getting the callerid number of the caller and getting callerid as "Asterisk" in
my  softphones (XLite).

From Internet, I came to know that

  1) In India, we have to give dtmf and ring for cidsignallling and cidstart
respectively.
  2) Default Asterisk setup doesn't recognise callerid in India. To recognize
callerid in India, we have to do or change some modifications in chan_zap.c
source file. Is it right?

Here I am giving my config files and error message. Please see it.

zaptel.conf contents:
loadzone = us
defaultzone=us
fxsks=1-4

zapata.conf contents:
[channels]
context=incoming
signalling=fxs_ks
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
cancallforward=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callerid=asreceived
language=en
usecallerid=yes
hidecallerid=no
echocancel=yes
transfer=yes
immediate=no
musiconhold=default
ringtimeout=8000
cidsignalling=dtmf
cidstart=ring
group=1
callgroup=1
pickupgroup=1
channel => 1

sip.conf  contents:
[105]
type=friend
username=105
secret=ravi
callerid="RaviKanth"
host=dynamic
context=leader
canreinvite=no
nat=yes
dtmfmode=rfc2833
allow=all

extensions.conf contents:
[incoming]
exten => s,1,Wait(10)
exten => s,n,Answer
exten => s,n,Noop('Callerid is '${CALLERID})
exten => s,n,Noop('Calleridname is '${CALLERIDNAME})
exten => s,n,Noop(Calleridnum is '${CALLERIDNUM})
exten => s,n,SetMusicOnHold(default)
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Background(/tmp/virg2)
exten => s,n,Goto(s,1)
exten => s,n,Hangup()
include => leader

[leader]
exten => 105,1,Dial(SIP/105,15)
exten => 105,2,Voicemail(u105)
exten => 105,3,Voicemail(b105)
exten => 105,4,Hangup
exten => _9XXXXXXXXXX,1,Dial(Zap/1/${EXTEN:1})   ; Mobile  phone
exten => _5XXXXXXXX,1,Dial(Zap/1/${EXTEN:1})       ; Local Landline
include => internal

[internal]
exten => 105, 1, Dial(SIP/105,15)

When somebody calls from outside (Eg: mobile), I am getting this below error
message on Asterisk console:

*CLI>     -- Starting simple switch on 'Zap/1-1'
Aug 18 14:53:13 ERROR[15499]: callerid.c:276 callerid_feed: fsk_serie made mylen
< 0 (-16)
Aug 18 14:53:13 WARNING[15499]: chan_zap.c:6087 ss_thread: CallerID feed failed:
Success
Aug 18 14:53:13 WARNING[15499]: chan_zap.c:6131 ss_thread: CallerID returned
with error on channel 'Zap/1-1'
     -- Executing Wait("Zap/1-1", "10") in new stack
     -- Executing Answer("Zap/1-1", "") in new stack
     -- Executing NoOp("Zap/1-1", " 18082006-14:53:24") in new stack
     -- Executing NoOp("Zap/1-1", "CallerID is ") in new  stack
     -- Executing NoOp("Zap/1-1", "CallerID Name is ") in new stack
     -- Executing NoOp("Zap/1-1", "CallerID Number is ") in new stack
     -- Executing SetMusicOnHold("Zap/1-1", "default") in new stack
     -- Executing Set("Zap/1-1", "TIMEOUT(digit)=5") in new stack
     -- Digit timeout set to 5
     -- Executing Set("Zap/1-1", "TIMEOUT(response)=10") in new stack
     -- Response timeout set to 10
     -- Executing BackGround("Zap/1-1", "/tmp/virg2") in new stack
     -- Playing '/tmp/virg2' (language 'en')
   == CDR updated on Zap/1-1
     -- Executing Dial("Zap/1-1", "SIP/105|15|t|12") in new stack
     -- Called 105
     -- SIP/105-00798410 is ringing
     -- Nobody picked up in 15000 ms
     -- Executing VoiceMail("Zap/1-1", "u105") in new stack
     -- Playing  'vm-theperson' (language 'en')
     -- Playing 'digits/1' (language 'en')
     -- Playing 'digits/0' (language 'en')
     -- Playing 'digits/5' (language 'en')
     -- Playing 'vm-isunavail' (language 'en')
     -- Playing 'vm-intro' (language 'en')
   == Spawn extension (incoming, 105, 2) exited non-zero on 'Zap/1-1'
     -- Hungup 'Zap/1-1'

Please tell me. Looking forward to your response. Thank you.

Regards,
Chandra.

---------------------------------
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  Get on board. You're invited to try the new Yahoo! Mail Beta.

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

Hi to one and all

i am new to this group and new to this technologies too. i am working
as trainee in one organisation.

The purpose of this mail is to introduce myself with all the members
of this group.

Hi Ram Kumar!!!

i only called to ur mobile today!!

with regards
ashok

Hi,

   Good Sign .
   i am Ranjan from Chennai .Myu n0 9840914326 / 9444206169 .Pls inform about
next metting . i will join with you people.

   Regards,
   Ranjan

sadheesh radha krishnan <sadeesh@...> wrote:
           Hi
I hope i am little late :( failing to attend the bnglr-meet on
asterisk. I am sadeesh from bangalore. My number is 9342916943. In
case, there is any meet or any discussions abt to happen in bnglr on
asterisk, do let me know. I m moving into asterisk appl.development
from operations team in my company. My yahoo id is "sadheeshr"

SADEESH.R
sadeesh@...
9342916943

--- In asterisk-india@yahoogroups.com, Ram Kumar Lanka <lanka143@...>
wrote:
>
>
>
> --- Salil Khamkar <salil@...> wrote:
>
> > --- In asterisk-india@yahoogroups.com, "acehunky2"
> > <acehunky2@> wrote:
> > >
> > > I think you should have provided cell phone number
> > of couple of
> > > people, so that the last minute guys can feel
> > comfortable to join you.
> > >
> > > Again thats just my thought.
> > >
> >
> > Yup - correct observation but Ram had already sent
> > in his cell number
> > the earlier day, so there was a contact number for
> > people to call.
> >
> > In my haste to send out a quick reminder, forgot to
> > add my contact
> > number, usually a part of my business mail sig.
> >
> > Anyways our Bangalore meet was good. 3 guys in total
> > discussing
> > various aspects of Asterisk.
> >
> > How was the Mumbai meet ?
> >
> > I'll try and drop in at the next meet in Mumbai.
> >
> > --
> > Salil
> >
>
> HI Everyone:
> We have made a beginning at 1st Bangalore Asterisk
> Meet.
>
> As I was mentioning in earlier mail, even 2 is good
> for beginning. However we were three. So the crowd is
> bigger than expecations. ;)
>
> 1. Salil
> 2. Venkat S G
> 3. Ram Kumar Lanka ( its me )
>
> It was held at 5 PM sharp at Bangalore's Garuda Mall
> Cafe Coffee Day.
> We discussed some practical aspects of Asterisk and
> shared thoughts about the future of asterisk in india.
> Some ideas on sourcing IP phones and other hardware in
> india for installing Asterisk and using it for
> personal and other uses.
>
> It went on really well.
> We have decided that we should organize Asterisk
> Bangalore Meet once in a month. Since we started the
> first Meet on 2nd Saturday of the month, we wish to
> stick to the same every month.
> However suggestions are always welcome. I wish some of
> the bangalore Asterisk fans participate eagarly and
> contribute to the meets to increase the crowd.
>
> If anyone else has some thoughts on this please do not
> hesitate to call me on my mobile 98451 69195 or
> express in this forum.
> I wish Bangalore Asterisk group increases tremondously
> in the future months.
>
> With Warm Regards
>
> Ram Kumar
>





  __________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
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[Non-text portions of this message have been removed]

Hi
I hope i am little late :( failing to attend the bnglr-meet on
asterisk. I am sadeesh from bangalore. My number is 9342916943. In
case, there is any meet or any discussions abt to happen in bnglr on
asterisk, do let me know. I m moving into asterisk appl.development
from operations team in my company. My yahoo id is "sadheeshr"

SADEESH.R
sadeesh@...
9342916943


--- In asterisk-india@yahoogroups.com, Ram Kumar Lanka <lanka143@...>
wrote:
>
>
>
> --- Salil Khamkar <salil@...> wrote:
>
> > --- In asterisk-india@yahoogroups.com, "acehunky2"
> > <acehunky2@> wrote:
> > >
> > > I think you should have provided cell phone number
> > of couple of
> > > people, so that the last minute guys can feel
> > comfortable to join you.
> > >
> > > Again thats just my thought.
> > >
> >
> > Yup - correct observation but Ram had already sent
> > in his cell number
> > the earlier day, so there was a contact number for
> > people to call.
> >
> > In my haste to send out a quick reminder, forgot to
> > add my contact
> > number, usually a part of my business mail sig.
> >
> > Anyways our Bangalore meet was good. 3 guys in total
> > discussing
> > various aspects of Asterisk.
> >
> > How was the Mumbai meet ?
> >
> > I'll try and drop in at the next meet in Mumbai.
> >
> > --
> > Salil
> >
>
> HI Everyone:
> We have made a beginning at 1st Bangalore Asterisk
> Meet.
>
> As I was mentioning in earlier mail, even 2 is good
> for beginning. However we were three. So the crowd is
> bigger than expecations. ;)
>
> 1. Salil
> 2. Venkat S G
> 3. Ram Kumar Lanka ( its me )
>
> It was held at 5 PM sharp at Bangalore's Garuda Mall
> Cafe Coffee Day.
> We discussed some practical aspects of Asterisk and
> shared thoughts about the future of asterisk in india.
> Some ideas on sourcing IP phones and other hardware in
> india  for installing Asterisk and using it for
> personal and other uses.
>
> It went on really well.
> We have decided that we should organize Asterisk
> Bangalore Meet once in a month. Since we started the
> first Meet on 2nd Saturday of the month, we wish to
> stick to the same every month.
> However suggestions are always welcome. I wish some of
> the bangalore Asterisk fans participate eagarly and
> contribute to the meets to increase the crowd.
>
> If anyone else has some thoughts on this please do not
> hesitate to call me on my mobile 98451 69195 or
> express in this forum.
> I wish Bangalore Asterisk group increases tremondously
> in the future months.
>
> With Warm Regards
>
> Ram Kumar
>

Thanks for the response Vikram. I've already tried
the steps below with the following results:

1. the card is on its own IRQ (no IRQ sharing)
2. USB, serial ports disabled
3. I get 1000 interrupts per sec
   (cat /proc/interrupts; sleep; cat /proc/interrupts)
4. zttool shows 0 IRQ misses
5. zttest shows 99.987763% accuracy (PROBLEM!!)
6. tried fxotune (latest May 16 patch) with no improvements
7. Didn't try ztspeed though ... all it seems to do is run
    a busy loop to load the CPU. Don't know what the 'expected'
    or 'ideal' number ought to be...
8. HDD is IDE, in UDMA2 mode
9. Even tried changing the opermode settings in the zaptel
    driver with no effect (default opermode=in)
10. X is turned off

   Any ideas about what other stuff could be tried out ?

Thanks,
Atul


--- In asterisk-india@yahoogroups.com, "Vikram Rangnekar"
<vicram@...> wrote:
>
> you will need to debug the card use the zaptel ztspeed, zttest
other
> provided apps to see if this card is working at over 99.99%. If
its not then
> you have a problem. Also check /proc/interrupts and check if the
card is
> generating 1000 intrrupts/sec if not thn again u have a problem.
Problem
> could be that something else on your box is eating resources check
what mode
> your HDD is in.
>
> On 5/30/06, atul_narkhede <atul_narkhede@...> wrote:
> >
> > Hello all,
> > I'm trying to configure the TDM400P card
> > with Asterisk to connect to PSTN, and am
> > running into audio quality problems. The
> > audio is 'clipped' (as if the gain is too
> > high). Tried playing around with txgain/rxgain
> > values ... that reduced the 'clipping' to
> > some extent but did not remove it completely.
> > There is no echo at all, but this clipping
> > problem is quite annoying, and is more pronounced
> > for calls made from cellphones.
> >
> > Has anybody gotten the TDM400P to work
> > with BSNL lines with good qudio quality ?
> >
> > Any ideas about how to fix this ? Also,
> > are there phone numbers in the exchange which
> > can be used to tune the line (miliwatt tone,
> > silence with comfort noise, etc) ?
> >
> > Would appreciate any help in this matter...
> >
> > - Atul
> >
> >
> >
> >
> >
> >
> >
> >  SPONSORED LINKS
> >   Communication and networking<http://groups.yahoo.com/gads?
t=ms&k=Communication+and+networking&w1=Communication+and+networking&w
2=Asterisk&w3=Computer+security&w4=Computer+training&c=4&s=94&.sig=8y
hxS-5cMb-xTGU5ZpVVew>
> > Asterisk<http://groups.yahoo.com/gads?
t=ms&k=Asterisk&w1=Communication+and+networking&w2=Asterisk&w3=Comput
er+security&w4=Computer+training&c=4&s=94&.sig=s5e0-
nmQjYnStZZt5Tsq_Q>  Computer
> > security<http://groups.yahoo.com/gads?
t=ms&k=Computer+security&w1=Communication+and+networking&w2=Asterisk&
w3=Computer+security&w4=Computer+training&c=4&s=94&.sig=wT4cT_PtZhmUS
d29XXK5Pw>   Computer
> > training<http://groups.yahoo.com/gads?
t=ms&k=Computer+training&w1=Communication+and+networking&w2=Asterisk&
w3=Computer+security&w4=Computer+training&c=4&s=94&.sig=fn7UYNwZ8YUuU
z1oyMV92Q>
> >  ------------------------------
> > YAHOO! GROUPS LINKS
> >
> >
> >    -  Visit your group "asterisk-
india<http://groups.yahoo.com/group/asterisk-india>"
> >    on the web.
> >
> >    -  To unsubscribe from this group, send an email to:
> >     asterisk-india-unsubscribe@yahoogroups.com<asterisk-india-
unsubscribe@yahoogroups.com?subject=Unsubscribe>
> >
> >    -  Your use of Yahoo! Groups is subject to the Yahoo! Terms of
> >    Service <http://docs.yahoo.com/info/terms/>.
> >
> >
> >  ------------------------------
> >
>
>
>
> --
> Vikram Rangnekar
>
>
> [Non-text portions of this message have been removed]
>

you will need to debug the card use the zaptel ztspeed, zttest other
provided apps to see if this card is working at over 99.99%. If its not then
you have a problem. Also check /proc/interrupts and check if the card is
generating 1000 intrrupts/sec if not thn again u have a problem. Problem
could be that something else on your box is eating resources check what mode
your HDD is in.

On 5/30/06, atul_narkhede <atul_narkhede@...> wrote:
>
> Hello all,
> I'm trying to configure the TDM400P card
> with Asterisk to connect to PSTN, and am
> running into audio quality problems. The
> audio is 'clipped' (as if the gain is too
> high). Tried playing around with txgain/rxgain
> values ... that reduced the 'clipping' to
> some extent but did not remove it completely.
> There is no echo at all, but this clipping
> problem is quite annoying, and is more pronounced
> for calls made from cellphones.
>
> Has anybody gotten the TDM400P to work
> with BSNL lines with good qudio quality ?
>
> Any ideas about how to fix this ? Also,
> are there phone numbers in the exchange which
> can be used to tune the line (miliwatt tone,
> silence with comfort noise, etc) ?
>
> Would appreciate any help in this matter...
>
> - Atul
>
>
>
>
>
>
>
>  SPONSORED LINKS
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--
Vikram Rangnekar


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

Hello all,
I'm trying to configure the TDM400P card
with Asterisk to connect to PSTN, and am
running into audio quality problems. The
audio is 'clipped' (as if the gain is too
high). Tried playing around with txgain/rxgain
values ... that reduced the 'clipping' to
some extent but did not remove it completely.
There is no echo at all, but this clipping
problem is quite annoying, and is more pronounced
for calls made from cellphones.

Has anybody gotten the TDM400P to work
with BSNL lines with good qudio quality ?

Any ideas about how to fix this ? Also,
are there phone numbers in the exchange which
can be used to tune the line (miliwatt tone,
silence with comfort noise, etc) ?

Would appreciate any help in this matter...

- Atul

> > Why do u want to take those services. you can have your own service.
>
>
> Things look very easy technically. I would suggest that you check for
> the legal issues first before doing anything like that. AFAIK,
> telecom regulations in India are less permissive than the US.
>
> --
> VaibhaV Sharma
> http://vsharma.net
>

    I  tried with net2phone setup in voip-info.org
and that one is not working.I tried with etherial to get the packet
trace also.They have cluster of voip servers and each time it uses
different server ip address and some encryption code.
If any of you have done this, please share the setup.

   Using Innomedia VOIP adapter and integrate with Digium FXO Cards are
   the next alternative(Both are availabe to me).
   Problem with this is a big delay.
   User need to wait atleast one miniute to get the ringing sound of
   US phone.
   Any workaround for this ?

Problem:

      It seems Siemens EPABX is giving different TDM signal
compared to normal PSTN TDM signals.So even though voice clarity
is good,it does't detect the hangup.

      So I am doing a command line restart of asterisk on
perticular intervel to get it work.
     How to troubleshoot these type of problesms ?
     OR any good solution for this ?

Thanks for ur suggestions

-Sandeep

I would prefer CentOS. Install CentOS 4. It is a replica of RHEL. You will find
more information on google why you should use CentOS.

Then download asterisk at home source code. follow the installation instruction
written in README/INSTALL. it is pretty simple as it downlowds required
compatible libraries (if you are connected to internet)

You dont need to post such question on the forum. all the information is
available on voip-info.org (Starting point). If you have any specific question
you can ask other guys.

Prashant



On Thu, 25 May 2006 hafiz rashid wrote :
>hi
>  
>  i installed fedora now tell me complete steps to install asterisk and
nesessary hardware.
>  
>  thanks
>  
>  
>           
>---------------------------------
>Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone.  Get Yahoo!
Messenger with Voice
>[Non-text portions of this message have been removed]
>SPONSORED LINKS
>Communication and networking
>Asterisk
>Computer security
>Computer training
>YAHOO! GROUPS LINKS
> Visit your group "
>asterisk-india
>" on the web.
> 
> To unsubscribe from this group, send an email to:
> 
>asterisk-india-unsubscribe@yahoogroups.com
> 
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>Yahoo! Terms of Service
>.


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

hi

   i installed fedora now tell me complete steps to install asterisk and
nesessary hardware.

   thanks




---------------------------------
Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone.  Get Yahoo!
Messenger with Voice

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

Hi all,

I dunno how many times I have to stress this email and repeat what I think,
but here it goes.

Its true is illegal, but do u know how many voip nerds are doing this
illegally?  Atleast they should do a ruling change that for personal use,
i.e their own registered phone should be okay.  Unless its going to be a
headache for the gov, already asterisk is starting to catch up in India, and
more and more people buying the x100p cards from ebay and making a bridge
for their own personal use(this is different from running a calling card
business or termination of some kind).  And with the gsm gatways coming in,
its going to be more fun:).  Technology is a 2 sided sword.  While India
does have a big market for telcos, there is a small group of users like
asterisk-india, who wants to implement voip and bridge it for their own
personal use.

You don't need to be a rocket scientist to run a asterisk, just load the
asterisk@home cd, plug in a fxo(x100p) and your off.  No special hardware
needed.  With the fxo cards are nothing but Intel or Motorola modems, its
little the government authorities can do.  Except maybe slap a fine.  But
who is this going to hurt? Big time voip players, who run illegal voip
exchanges, who profit, lakhs of money?  Maybe reliance pay's under the table
for their India2call service, who knows.  But one thing is clear, you don't
need a telco switch to terminate.  And everyone can do it now.  If your not
a registered telco company and your doing this for your own use, what is the
police going to do? I would like someone to answer this.

There is lot of vonage adaptors floating on the Indian soil now, and its
only a matter of time before India gets flooded with DID's from all over the
world.  I don't think anyone these days uses a normal phone to call their
loved ones in India.  Most ppl use skype, IM or some even use ata's from
vonage ..etc.  I mean think of it, who is going to pay Rs.16 for a min call
to US, when u can call for 45mins for the same amount of money on voip?

Seriously if you talk about licence losses for the big telcos, then gov are
shooting themselves in the foot.  There is only some time before some sort
of announcement will be made on voip regulation.  Look at some of the
developed countries US,UK Australia, Singapore,  HongKong and how gov's are
embrasing the voip in their homeland.  With data and voice converging, its
only sooner than latter.

What is annoying to me is, Indian gov wants BPO money on taxes, yet they are
against voip:P Like all technologies, it has good and bad things.  I hope
that mr maran, will make some headway into this issue.  Its funny reading
some articles on voip regulations in India.  Somewhere I read cisco got a
semi call center status, oh please, does every tom dick harry needs a semi
call center licence? I don't think so:P


Regards,
Dinesh.


-----Original Message-----
From: asterisk-india@yahoogroups.com [mailto:asterisk-india@yahoogroups.com]
On Behalf Of Prashant Bhalesain
Sent: Wednesday, May 24, 2006 9:10 AM
To: asterisk-india@yahoogroups.com
Subject: Re: Re: [asterisk-india] Asterisk with any VOIP service provider

If this person was asking to provide asterisk service, how he thinks he can
do that? If you read the regulation document related to termination and
origination service in india, it specifically says that you cannot have any
dialing equipment connected to internet which can be used for making
overseas/phone calls. Unless you are trying to implement it illigally it is
not yet allowed.

And if vonage is doing that, I am sure they will have to pay for it, as no
foreign company is allowed to provide such facility. You will find more
information on www.dot.gov.in. The only word I had after reading those
documents was "sucks". To provide such service you have to be an indian
company with 51% investement with net company worth 25cr. (NRI investment is
considered as foreign investment :-) )

Prashant


On Wed, 24 May 2006 VaibhaV Sharma wrote :
>On May 23, 2006, at 3:31 PM, Prashant Bhalesain wrote:
>>
>> I thought he is doing that in USA. (Since he mentioned Vonage/USA/
>> Canada)
>Nope, he wants to let his users (I assume in India) to call numbers 
>in US/Canada using an IP account over the internet.
>> In india its not possible unless you have 25 crores in your pocket. 
>> (www.dot.gov.in)  Also there are thousands of regulations for 
>> terminating and originating service. Even if you are an ISP 
>> provider u have to pay 20 lakhs for B grade city and with that too 
>> u can provide only Pc-2-PC service (I mean legally) Illigally u can 
>> see people having kiosks running all over gujrat and bby.
>>
>> I wonder how reliance is doing that. "under the table"? I wonder 
>> when India will be out of those wealthy family's hands which are 
>> holding every possible business for themselves.
>Reliance owns a lot of marine OFC. They basically bought Flag3 a few 
>years back when they lost the bid for VSNL to the TATAs. Is Reliance 
>offering calling plans to the US too?
>Reliance otherwise is a telecom service provider and is terminating 
>calls *from* the US/Canada/UK to its switches / passing them to other 
>providers by paying a fee sharing scheme. They were caught passing 
>those calls as local calls sometime back and they paid for it.
>--
>VaibhaV Sharma
>http://vsharma.net
>YAHOO! GROUPS LINKS
> Visit your group "
>asterisk-india
>" on the web.
> 
> To unsubscribe from this group, send an email to:
> 
>asterisk-india-unsubscribe@yahoogroups.com
> 
> Your use of Yahoo! Groups is subject to the
>Yahoo! Terms of Service
>.


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





Yahoo! Groups Links

If this person was asking to provide asterisk service, how he thinks he can do
that? If you read the regulation document related to termination and origination
service in india, it specifically says that you cannot have any dialing
equipment connected to internet which can be used for making overseas/phone
calls. Unless you are trying to implement it illigally it is not yet allowed.

And if vonage is doing that, I am sure they will have to pay for it, as no
foreign company is allowed to provide such facility. You will find more
information on www.dot.gov.in. The only word I had after reading those documents
was "sucks". To provide such service you have to be an indian company with 51%
investement with net company worth 25cr. (NRI investment is considered as
foreign investment :-) )

Prashant


On Wed, 24 May 2006 VaibhaV Sharma wrote :
>On May 23, 2006, at 3:31 PM, Prashant Bhalesain wrote:
>>
>> I thought he is doing that in USA. (Since he mentioned Vonage/USA/
>> Canada)
>Nope, he wants to let his users (I assume in India) to call numbers 
>in US/Canada using an IP account over the internet.
>> In india its not possible unless you have 25 crores in your pocket. 
>> (www.dot.gov.in)  Also there are thousands of regulations for 
>> terminating and originating service. Even if you are an ISP 
>> provider u have to pay 20 lakhs for B grade city and with that too 
>> u can provide only Pc-2-PC service (I mean legally) Illigally u can 
>> see people having kiosks running all over gujrat and bby.
>>
>> I wonder how reliance is doing that. "under the table"? I wonder 
>> when India will be out of those wealthy family's hands which are 
>> holding every possible business for themselves.
>Reliance owns a lot of marine OFC. They basically bought Flag3 a few 
>years back when they lost the bid for VSNL to the TATAs. Is Reliance 
>offering calling plans to the US too?
>Reliance otherwise is a telecom service provider and is terminating 
>calls *from* the US/Canada/UK to its switches / passing them to other 
>providers by paying a fee sharing scheme. They were caught passing 
>those calls as local calls sometime back and they paid for it.
>--
>VaibhaV Sharma
>http://vsharma.net
>YAHOO! GROUPS LINKS
> Visit your group "
>asterisk-india
>" on the web.
> 
> To unsubscribe from this group, send an email to:
> 
>asterisk-india-unsubscribe@yahoogroups.com
> 
> Your use of Yahoo! Groups is subject to the
>Yahoo! Terms of Service
>.


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

On May 23, 2006, at 3:31 PM, Prashant Bhalesain wrote:

>
> I thought he is doing that in USA. (Since he mentioned Vonage/USA/
> Canada)

Nope, he wants to let his users (I assume in India) to call numbers
in US/Canada using an IP account over the internet.

> In india its not possible unless you have 25 crores in your pocket.
> (www.dot.gov.in)  Also there are thousands of regulations for
> terminating and originating service. Even if you are an ISP
> provider u have to pay 20 lakhs for B grade city and with that too
> u can provide only Pc-2-PC service (I mean legally) Illigally u can
> see people having kiosks running all over gujrat and bby.
>
> I wonder how reliance is doing that. "under the table"? I wonder
> when India will be out of those wealthy family's hands which are
> holding every possible business for themselves.

Reliance owns a lot of marine OFC. They basically bought Flag3 a few
years back when they lost the bid for VSNL to the TATAs. Is Reliance
offering calling plans to the US too?

Reliance otherwise is a telecom service provider and is terminating
calls *from* the US/Canada/UK to its switches / passing them to other
providers by paying a fee sharing scheme. They were caught passing
those calls as local calls sometime back and they paid for it.

--
VaibhaV Sharma
http://vsharma.net

 
I thought he is doing that in USA. (Since he mentioned Vonage/USA/Canada)

In india its not possible unless you have 25 crores in your pocket.
(www.dot.gov.in)  Also there are thousands of regulations for terminating and
originating service. Even if you are an ISP provider u have to pay 20 lakhs for
B grade city and with that too u can provide only Pc-2-PC service (I mean
legally) Illigally u can see people having kiosks running all over gujrat and
bby.

I wonder how reliance is doing that. "under the table"? I wonder when India will
be out of those wealthy family's hands which are holding every possible business
for themselves.

Prashant

On Wed, 24 May 2006 VaibhaV Sharma wrote :
>On May 23, 2006, at 10:42 AM, Prashant Bhalesain wrote:
>>
>> You can do that, sip.conf or iax.conf depends on what your VOIP 
>> provider uses.
>>
>> Why do u want to take those services. you can have your own service.
>Things look very easy technically. I would suggest that you check for 
>the legal issues first before doing anything like that. AFAIK, 
>telecom regulations in India are less permissive than the US.
>--
>VaibhaV Sharma
>http://vsharma.net
>SPONSORED LINKS
>Communication and networking
>Asterisk
>Computer security
>Computer training
>YAHOO! GROUPS LINKS
> Visit your group "
>asterisk-india
>" on the web.
> 
> To unsubscribe from this group, send an email to:
> 
>asterisk-india-unsubscribe@yahoogroups.com
> 
> Your use of Yahoo! Groups is subject to the
>Yahoo! Terms of Service
>.


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

On May 23, 2006, at 10:42 AM, Prashant Bhalesain wrote:

>
> You can do that, sip.conf or iax.conf depends on what your VOIP
> provider uses.
>
> Why do u want to take those services. you can have your own service.


Things look very easy technically. I would suggest that you check for
the legal issues first before doing anything like that. AFAIK,
telecom regulations in India are less permissive than the US.

--
VaibhaV Sharma
http://vsharma.net

 
You can do that, sip.conf or iax.conf depends on what your VOIP provider uses.

Why do u want to take those services. you can have your own service.

prashant
On Tue, 23 May 2006 Sandeep A.S wrote :
>Hi
>  Anybody integrated asterisk with any of the VOIP
>  service providers like vonage,n2p
> 
>  I am looking for something like that so that I can give
>  a bridged connection to my users for US and canada calling
> 
>  -Sandeep
> 
>YAHOO! GROUPS LINKS
> Visit your group "
>asterisk-india
>" on the web.
> 
> To unsubscribe from this group, send an email to:
> 
>asterisk-india-unsubscribe@yahoogroups.com
> 
> Your use of Yahoo! Groups is subject to the
>Yahoo! Terms of Service
>.


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

On May 23, 2006, at 8:59 AM, Sandeep A.S wrote:

>   Anybody integrated asterisk with any of the VOIP
>   service providers like vonage,n2p
>
>   I am looking for something like that so that I can give
>   a bridged connection to my users for US and canada calling

Vonage does not allow you to connect their phone accounts with a PBX.
However, a few people have figured out the details of using their
soft phone accounts with Asterisk as the client. I do not have the
URLs off-hand. You can google for the details.

Otherwise, making asterisk talk to another provider depends on the
type of connectivity the provider gives you. You can use the Vonage
adapter to an analog phone port and it works like a normal phone
line. The only hitch being the hangup issue with their adapters which
can be fixed by calling vonage and using some social engineering. A
lot of other providers have tutorials on their website on how to
configure an SIP client.

It is not very difficult but you would need to read through the
documentation and experiences on voip-info.org to get that working.

--
VaibhaV Sharma
http://vsharma.net

Hi

Sandeep

Which type of account u got SIP or IAX

I have done with both
U will need to create trunk

Reg
kishor

-----Original Message-----
From: asterisk-india@yahoogroups.com [mailto:asterisk-india@yahoogroups.com]
On Behalf Of Sandeep A.S
Sent: Tuesday, May 23, 2006 6:29 PM
To: asterisk-india@yahoogroups.com
Subject: [asterisk-india] Asterisk with any VOIP service provider


Hi

   Anybody integrated asterisk with any of the VOIP
   service providers like vonage,n2p

   I am looking for something like that so that I can give
   a bridged connection to my users for US and canada calling


   -Sandeep








Yahoo! Groups Links

Hi

   Anybody integrated asterisk with any of the VOIP
   service providers like vonage,n2p

   I am looking for something like that so that I can give
   a bridged connection to my users for US and canada calling


   -Sandeep

 


Hi,

Problem:

How to configure GrandStream HT488 with asterisk. (Like an originating
service).I am trying to find out if I could set a toll free number using
grandstream and transfer all my calls to asterisk. then those calls would be
forwarded to my PSTN or CDMA gateways.(just like termination service)

prashant

On Thu, 18 May 2006 kuldeep dhanker wrote :
>Prashant,
>It would help if you could describe what exactly is
>the problem you are having with this.
>regards
>Kuldeep
>__________________________________________________
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