at voipds dot org'.
Thanks for your time,
Balaji NJL
balaji at voipDS dot org
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
___
--Bandwidth and Colocation sponsored
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
Balaji NJL wrote:
Announcing Voice over IP Directory Services
(http://www.voipDS.org)
Sorry to be negative, but this kind of services came
up in tons
when e-mail was a new service. All of the addresses
registred in there
is now
Hi All,
I recently signed up with a VOIP provider that
supports SIP. By default the provider supports X-term.
i downloaded X-lite configured as mentioned in their
site and got it working.
These are the details for configuring x-lite
Enter the
at 192.168.0.8 port 14096
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
--- Tony Hoyle [EMAIL PROTECTED] wrote:
Balaji NJL wrote:
-- Call accepted by 65.39.205.121 (format
ulaw)
-- Format for call
Hi,
My asterisk server is behind firewall and i am trying
to connect to FWD. i hv configured as mentioned in
this link
http://www.freeworlddialup.com/advanced/iax. i am able
to register my server with FWD. But when i dial
393612, i always get 'No one is available to answer
this time, try again
Correction.
i hv port forwarded udp 4569 and 5060.
-B
--- Balaji NJL [EMAIL PROTECTED] wrote:
Hi,
My asterisk server is behind firewall and i am
trying
to connect to FWD. i hv configured as mentioned in
this link
http://www.freeworlddialup.com/advanced/iax. i am
able
to register my
Hi All,
I am trying to find if there is a windows client for
sending fax using txfax - spandsp. i did a search but
i couldnt find anything significant. Any help
appreciated.
thanks,
-Balaji
Start your day with Yahoo! - make
-- Time Bandit [EMAIL PROTECTED] wrote:
On 6/18/05, Balaji NJL [EMAIL PROTECTED] wrote:
Hi All,
I am a new bee to *. I just installed
[EMAIL PROTECTED] on
FC3. I hv a FXO card. I hv configured two
extensions
one x-lite and other iaxComm. I configured * using
AMP. The following
0
Appreciate your help.
-B
--- Balaji NJL [EMAIL PROTECTED] wrote:
Hi All,
I am a new bee to *. I just installed [EMAIL PROTECTED]
on
FC3. I hv a FXO card. I hv configured two extensions
one x-lite and other iaxComm. I configured * using
AMP. The following setup works
- x-lite (x
Can someone connect to my server and leave a message.
i appreciate it.
-B
--- Balaji NJL [EMAIL PROTECTED] wrote:
Can someone leave a message at x 200 on my * server.
External IP two one six . nine . zero . three four
Connect as x 202
password zxc123
using IAX2
thanks,
-B
Hi All,
I am a new bee to *. I just installed [EMAIL PROTECTED] on
FC3. I hv a FXO card. I hv configured two extensions
one x-lite and other iaxComm. I configured * using
AMP. The following setup works
- x-lite (x 200) to iaxComm (x 201)
- PSTN to x-lite
- PSTN to iaxComm
Voice mail, weather etc
Can someone leave a message at x 200 on my * server.
External IP two one six . nine . zero . three four
Connect as x 202
password zxc123
using IAX2
thanks,
-B
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
man qmail-inject. this is one way u can inject mail in
qmail queue.
- Original Message -
From: Ing Isianto Istiadi
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 9:45 PM
Subject: RE: [Asterisk-Users] Sending voicemail with
qmail
On Thu, 15 Jan 2004
i didnt mean to say sendmail is a secure email server.
what i meant was its
not good to install * and qmail on the same box and
use it for internet
mail. In my case i hv qmail on a separate server which
is on DMZ. My * is on
a separate machine in my internal network and not
exposed to Internet. i
you can do that. But are u installing qmail and * on
same box. i wont
recommend that. i use qmail and *. qmail is strictly
for internet email. *
is on separate server not exposed to Internet. * box
also has sendmail. i hv
configured sendmail to use smart host (qmail server).
This way its safe and
Hey i stay at Redmond too. Prbly we all can meet.
-B
- Original Message -
From: Brett Schwarz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 2:00 PM
Subject: RE: [Asterisk-Users] linux journal article on
asterisk
arrggh, sorry about the website. It should be
usually it takes 2 - 3 months for the article to
appear on the website. LJ
doesnt post all the articles immediately. (otherwise
people wont buy LJ :-))
-B
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 7:21 AM
Subject: RE:
Hi All,
i just applied this patch. i need to test whether its
working. Can someone
connect to my server and leave me a vm at extension
2000.
Server : ojoobala.com
Phone
Extension : 2005
pwd : mytest
auth: md5.
pl leave a vm on extension 2000.
thanks a lot,
-B
- Original
.
Works great over the internet. Didn't need patches
or anything else.
I hope that helps you.
-C
www.ntfs.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Balaji NJL
Sent: 27 December 2003 08:34
To: [EMAIL PROTECTED]
Subject
before buying spa-2000 i would recommend ur sister to
use one of the
softphones (X-tern etc). First make sure she can
connect to ur asterisk then
try to establish a softphone-to-softphone
communication.
X-tern (Ur sister) --- * - X-Tern (You). Get
this setup working first.
-B
- Original
Add this to ur sip.conf ..that would help u.
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
-B
And sip.conf contains this
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
Hi All,
Can we stop this thread pl. This lady has no
intentions to learn asterisk.
She is just a troll and wasting our time. With her
corporate attitude, what
she expects is support that available with paid
commercial products. Her
company has enough money to buy commercial products,
let she go
i second it.
i had no previous experience with any telecom
equipment or even the lingo. I
didnt know what channel meant in telecom world. i
started with the *
handbook and did a lot of googling, searched the
archives. Search, search
and search. There is lot of info here. i am happy to
say that i
great over the internet. Didn't need patches
or anything else.
I hope that helps you.
-C
www.ntfs.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Balaji NJL
Sent: 27 December 2003 08:34
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users
Hi All,
i updated * last night. And after that i am unable
to make any outbound calls from GS. all my config files are same no change. When
ever i try to make a call from GS i constantly get this warning in *
console.
Warning: Filechan_sip.c line471 Maximum
retires exceeded on call some
i tried with other softphones. the only phone thats
working with GS is Xtern. MSN and SJ doesnt work. Is this a known
issue.
Thanks,
-B
- Original Message -
From:
Balaji NJL
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 7:05
PM
Subject: Re
resending.
Can anyone help me in trying to understand what
would be the problem. appreciate ur time. i need to get this
working.
thanks a lot,
-B
- Original Message -
From:
Balaji NJL
To: [EMAIL PROTECTED]
Sent: Monday, December 22, 2003 8:15
PM
Subject
On Mon, 2003-12-22 at 20:42, Balaji NJL wrote:
use this
[3001]
type=friend
;username=3001
;fromuser=Craig1
;secret=secret
host=dynamic
mailbox=3001
context=sip
dtmfmode=info
auth=plaintext
make sure ur MSN version is 4.7.0105.
-B
anyone help me in trying to
understand what would be the problem. appreciate ur time. i need to get
this working.
thanks a
lot,
-B
- Original Message -
From: Balaji NJL
To: [EMAIL PROTECTED]
Sent: Monday
use this
[3001]
type=friend
;username=3001
;fromuser=Craig1
;secret=secret
host=dynamic
mailbox=3001
context=sip
dtmfmode=info
auth=plaintext
make sure ur MSN version is
4.7.0105.
-B
- Original Message -
From:
Craig
Waddington
To: [EMAIL PROTECTED]
Sent:
Hi All,
i dont what changes i made recently but i am unable
to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS
and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine
too.
my SIP details
[general]port = 5060bindaddr =
0.0.0.0context =
Hi,
i am interested. send me more details on ur project
and where r u located.
-Balaji
- Original Message -
From:
Craig
Waddington
To: [EMAIL PROTECTED]
Sent: Sunday, December 21, 2003 4:12
PM
Subject: [Asterisk-Users] MSN messenger
and *
I have
Do a search on this list for details on how to
configure MSN. Also post more info like what version u r using and send the
details of ur sip.conf.
-B
- Original Message -
From:
Craig
Waddington
To: [EMAIL PROTECTED]
Sent: Sunday, December 21, 2003 4:12
PM
Hi All,
Can i install * on a beowulf cluster or Is *
compatible to clusters. I am planning to install a 4 node beowulf cluster using
few cheap hardwares. If no one had tried before i can spend some time on
installing and configuring * on this cluster. Let me know.
thanks,
-Balaji
Do you
Hi All,
i received my X100P and Grandstream phone last
week. i started configuring my * and with the help of ur mailing lists i was
able to configure it. (when ever i got struck i searched this list and found my
answer. thanks a lot and this list is awesome). i still hv a small problem and
: Balaji NJL
To: [EMAIL PROTECTED]
Sent: Monday, December 15, 2003 8:47 AM
Subject: [Asterisk-Users] unable to configure my
Grandstream phone
snip
Attempting native bridge of SIP/2003-b895 and
SIP/2000-53e2
WARNING[5126]: File chan_sip.c, Line 1954
(process_sdp): No compatible
codecs
any call from my GS.
i am no longer getting codec not compatible error
message anymore. i am
still unable to place any calls using my GS (to my
internal MSN extensions
or to external PSTN).
thanks for ur help,
-B
- Original Message -
From: Balaji NJL [EMAIL PROTECTED]
To: [EMAIL PROTECTED
: Roy Sigurd Karlsbakk [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 1:33 AM
Subject: Re: [Asterisk-Users] unable to make it work
with MSN Messenger
Prolly change the auth= to plaintext...
On Wed, 2003-12-03 at 10:07, Balaji NJL wrote:
Hi All,
I am
)exten =
2000,103,Hangup
;2001exten = 2001,1,Dial(SIP/2001,20)exten =
2001,2,Voicemail(u2001)exten = 2001,102,Voicemail(b2001)exten =
2001,103,Hangup
;2999 Voice mailexten = 2999,1,VoicemailMain(${CALLERIDNUM})
voicemail.conf
[general]
;format=wavformat=gsm
[local]
2000 = 1234,Balaji NJL
Voice mailexten = 2999,1,VoicemailMain(${CALLERIDNUM})
voicemail.conf
[general]
;format=wavformat=gsm
[local]
2000 = 1234,Balaji NJL,[EMAIL PROTECTED]2001 = 5678,Ojasvi
Sinha,[EMAIL PROTECTED]
2002 = 1234, Balaji NJL,[EMAIL PROTECTED]
Any idea whats the issue. any help appreciated.
thanks
Resending this. Any help appreciated.
Thanks,
-B
- Original Message -
From:
Balaji NJL
To: [EMAIL PROTECTED]
Sent: Sunday, October 19, 2003 6:36
PM
Subject: [Asterisk-Users] newb - want to
create a Dialpad like system
Hi all,
i am planning
Hi all,
i am planning to create Dialpad like system for
fun. i want to build itin such a way that one can use either web based app
or GnoPhone / MsnMessenger to connect to my server and then dial a land
line.i did a search on the archives but couldnt find any good
pointers. i would
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