Re: [Asterisk-Users] info regarding hardware

[Sandeep A.S]

In india no distributer for digium cards

If any body is going to us u can ask them to bring it.
I got in that way
-sandeep

Ankit wrote:



where did u purchase ur card frm, im not able to find ne distributor 
of digium cards in india, and if i order it frm their site it will 
have to pay arnd 2k rs for shipping :(


-ankit


On 8/9/05, *Gurminder Arora* <[EMAIL PROTECTED] 
> wrote:


I m using it on  POTS line and will start with ISDN soon :-).


Cheers
Gurminder
On 8/9/05, Ankit <[EMAIL PROTECTED] > wrote:
>
>  hi gurminder,
>  are you using it on isdn line or pots line?
>
>
> On 8/9/05, Gurminder Arora <[EMAIL PROTECTED]
> wrote:
> >
> > Hi
> >
> > Digium cards are compatible with indian telephony..
> > I am using it.
> > But there is problem I am facing to configure caller ID.
> >
> > What cidsignalling is used in india?
> >
> > Regards
> > Gurminder
> >
> >
> >
> >
> >
> >
> > On 8/8/05, Ankit <[EMAIL PROTECTED]
> wrote:
> > > Hi everybody,
> > >
> > >  I need a little clarification regarding the hardware to be
used with
> > > asterisk. I want to setup an asterisk box to make calls
through both
> > > internet and pstn, but i heard frm my friend (he was not
sure) that
> digium
> > > cards are incompatible with indian telephony systems, is it
so? If yes,
> then
> > > is there a way around this problem?
> > >
> > >  Thanks in advance,
> > >  Ankit
> > >
> > >  P.S- It would be greatly appreciated if someone could provide a
> technical
> > > explanation to why digium cards are incompatible with indian
(or
> anyother
> > > telephone system), i thought telephone network is same
everywhere.
> > >
> > > ___
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> > >
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Re: [Asterisk-Users] Call Meeting VS Call Confrence

[Sandeep A.S]

Hi  I am tried the patch  for outboud call from conferance

but the following error :
[EMAIL PROTECTED] asterisk]# patch -p0 < app_meetme.c_outboundcall_rev3.txt
patching file apps/app_meetme.c
Hunk #1 succeeded at 34 (offset 1 line).
Hunk #2 succeeded at 63 with fuzz 2 (offset 1 line).
Hunk #3 succeeded at 112 (offset 3 lines).
Hunk #4 succeeded at 163 (offset 4 lines).
Hunk #5 FAILED at 191.
Hunk #6 FAILED at 553.
Hunk #7 succeeded at 682 with fuzz 1 (offset 54 lines).
Hunk #8 succeeded at 1112 (offset 131 lines).
Hunk #9 FAILED at 1464.
Hunk #10 succeeded at 1726 (offset 38 lines).
3 out of 10 hunks FAILED -- saving rejects to file apps/app_meetme.c.rej

my asterisk version is :
#asterisk -V  gives
Asterisk CVS-HEAD-04/12/05-18:15:04

Pl suggest me what went wrong.

Thanks
Sandeep



Peter Svensson wrote:


On Thu, 2 Jun 2005, Mohamed A. Gombolaty wrote:

 


I was trying to make call confrence available but all the asterisk
documents use the meeting room concept, where those who wanna meet have
to dial an extension corresponding to the meeting room, while call
conference actually means that I am on exten 100 I can dial exten 200
and add it to confrence and again dial 333 and add it to the confrence
and so  on.
   



 


Is there any way to make call confrencing available and not meeting room
concepts?
   



There is a patch to add call out from within a meetme conference. See bug 
number 3405 on http://bugs.digium.com/.


Peter

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[Asterisk-Users] Replacing SIP Trunking With IAX Trunking

[Sandeep A.S]

I have the   sip trunking as below :
I tried with IAX Trunking .But no success
Can some one send  IAX Trunking config for the  below setup  replacing 
SIP  ?


PBX1 (192.168.10.2)
==
sip.conf
--
[pbx]
type=friend
username=pbx
secret=pbx
host=192.168.1.2

extensions.conf

exten => 1113,1, Dial(SIP/abc1,10,t)
exten => 1158,1, Dial(SIP/xyz1,10,t)
exten => _2XXX,1, Dial(SIP/pbx/${EXTEN})

PBX2 (192.168.1.2)
==
sip.conf
--
[pbx]
type=friend
username=pbx
secret=pbx
host=192.168.10.2

extensions.conf

exten => 2113,1, Dial(SIP/abc2,10,t)
exten => 2158,1, Dial(SIP/xyz2,10,t)
exten => _1XXX,1, Dial(SIP/pbx/${EXTEN})

Thanks
Sandeep
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[Asterisk-Users] SIP or IAX

[Sandeep A.S]

For bridging  VOIP with  PSTN Lines
Which one is giving better performance  SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
Also with single SIP/IAX channel can I use more than one call at a time ?

Thanks
Sandeep
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[Asterisk-Users] TDM400P Channels stop answering after some time .

[Sandeep A.S]

Hi

Need help on bridging SIP with TDM400P(4 FXO Modules )

My setup is as follows


US OFFICE -TDM400P(FXO) --SIP--- TDM400P(FXOs)INDIA OFFICE
(DSL Line)  Asterisk
Asterisk   PBX(Siemens) /DSL Line
 Server  
Server


Everithing works fine for one or two calls or maximum 4 calls over
the setup.

Ie after some time zap channels are not ringing.Then I have to reload
asterisk.Once restart everithing works fine for 2 or 3 calls over the setup
then the same issue .I need to restart asterisk again .

Is it the problem with TDM400P ?
OR the problem with 2.6 Kernel ?
or  Problem with SIP and TDM Card ?
How I can troubleshoot ?

I am using Fedora core3 Kernel 2.6.9-1.667

My zaptel.conf on both systes:
loadzone = us
defaultzone=us
fxsks=1-4

My zapdata.conf on both systems :

signalling=fxs_ks
rxwink=300
usecallingpres=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=yes
rxgain=4.9
txgain=6.9
busydetect=yes
callprogress=yes
progzone=us
musiconhold=default
jitterbuffers=4

My sip.conf on both systems
[pbx]
type=friend
username=pbx
secret=pbx
host=192.168.X.Y
dtmfmode=info
insecure=very
qualify=no
disallow=all
allow=ulaw

Do you want any more details ?

thanks
-Sandeep
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Re: [Asterisk-Users] [OFF TOPIC] Voip phone sellers in India

[Sandeep A.S]

check with webtel,Mob:32333033

On Sun, 2005-01-09 at 19:33 +0100, Vikram Rangnekar wrote:
> I am looking for some in India  to buy VOIP phones from. Please get in touch
> with me off the list on [EMAIL PROTECTED]
> 
> Sorry for the off topic mail I am just having such a hard time finding any
> voip phones in India that I got desperate and didnt know which list to post
> this on.
> 
> 
-- 
Sandeep A.S <[EMAIL PROTECTED]>
Netcontinuum Pvt Ltd 

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