( I wish ), Nufone please fix this ASSAP.
Later
Sathya
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. May be I am one unfortunate folk. Hope I too can soon have a rock
solid service.
Still waiting
Cheers
Sathya
Message: 8
Date: Sat, 24 Jan 2004 14:36:54 -0500
From: Frankie Gravato [EMAIL PROTECTED]
Organization: Cfsdigital
To: Sathya [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Has
than an email from paypal.
Friends, so I got a response from NuFone Sales.
This is just the start, hope I will get a rock solid service.
Cheers
Sathya
Message: 8
Date: Sun, 25 Jan 2004 21:06:48 -0500
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users
Hi,
I've tried almost every where to see whether there is a config parameter to
set dtmf on IAX channels. Just like in SIP where we can set dtmfmode as
inband, info or rfc. I am experiencing a problem with inbound DTMF where *
interoperates a digit in a string as couple or more (intermittently).
Hi,
I have a basic dial plan question;
Here is the scenario.
Call comes through IAX and my * authenticate, then collect the digits and
dials out, simple :).
Here is the dial plan;
[did-in]
;for did callers
exten = 866219,1,Ringing
exten = 866219,2,Wait,4
exten = 866219,3,Answer
Hi friends,
I am experiencing lot of duplicate digits especially when people dial-in
using Cellular phones.
here is my config;
PSTN(PRI)---Asterisk A---(IAX)-Asterisk B--SIP Phones,
H.323 G/W
Asterisk A switches calls to Asterisk B. Asterisk B answers the call and
getenv.agi:
agi_request: getenv.agigetenv.agi: agi_channel: SIP/-081524c0getenv.agi:
agi_language: engetenv.agi: agi_type: SIPgetenv.agi: agi_uniqueid:
1079819757.97getenv.agi: agi_callerid: Sathya
=getenv.agi: agi_dnid: unknowngetenv.agi:
agi_rdnis: unknowngetenv.agi: agi_con
Hi David,
Thanks, yes that was the problem.
Really appreciate your tip.
Cheers
Sathya
From: David Croft [EMAIL PROTECTED]
Organization: Sargasso Networks
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] can't get the full callerid php/agi
Reply-To: [EMAIL PROTECTED]
Your script
registration interval
to a higher value.
I tend to think that
nufone server is loosing my * and hence my callers get "number unreachable
message" due to the fact that I have to keep on registering with
it.
Any help
appreciated.
Cheers
Sathya
Hi,
Its a
dedicated server hosted at an ISP, my server it self got no firewall. I will
check with the hosting company though.
But,
how does a firewall triger my asterisk to send registration info periodically
?
thanks
Sathya
-Original Message-From:
[EMAIL PROTECTED
Hi,
No matter what I set
in my sip.conf, I always get '3' as amaflags in my mysql
cdr.
(a) How do I make
amaflags correctly set in mysql cdr
(b) Seperate note,
how can I set amaflags from agi
Thanks
SW
that ?
cheers
Sathya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Wednesday, June 23, 2004 9:03 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] help needed with read()
On Wed, 2004-06-23 at 10:12, Sathya wrote:
asterisk
12 ms is what I saw when I did * to * IAX.
In iax.conf, set:
notransfer=yes
That prevents IAX from transferring call to remote Asterisk, so it will
stay in path.
Sathya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Friday
$50 is a bit steep for a book
Usually author of a book makes 10 to 15 percent of the cover price. So
whoever who wrote this book will get 5 bucks a book. Unlike a book widely
used, for an example PHP or MySQl, which will sell lot more copies, a book
on * will have a narrower readership.
Hi,
Ineed to pass
the "call duration" and " Bill Sec" after a successfull call to an AGI
script. Is there a way to do this ?
Cheers
Sathya
Hi,
Can a wildcard T100P
be installed in a 1U server ??
Sathya
Hi,
I am trying to make a call from SIP to H.323 using chan_h323. Asterisk
CVS-HEAD-03/10/05-10:08:22. As given in chan_h323 readme I compiled pwlib
and open h323 versions 1.8.1 and 1.15.1. Call seems to be get connected but
no audio path.
I see following;
-- AGI Script Executing Application:
between extensions and to fwd works fine. Voice
prompt at fwd works fine too.
Anyone could help me here ?
Appreciate
Sathya
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Hello,
I am posting this after spending hours digging through the list archives.
Problem : When asteirsk plays a voice prompt, the voice clip is really
choppy.
I figure that this is something to with the sound card, the timing of
playback etc.
But cannot seems to find an answer.
Here is the
6 Phone number is received over DTMF
7 Asteirsk route the call
thanks in advance.
Cheers
Sathya
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Configuration
==
Channel map:
0 channels configured.
Anyone can shed some light here.
Cheers
Sathya
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have your voice prompt in wav, then use sox to convert it to
gsm. This way you can make any voice promt free and easy.
Sathya
-Original Message-
From: Darren Wiebe [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 04, 2004 5:41 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
I have a * server
which does only SIP to H323 completely in IP domain, there is no digium h/w in
it. In your experience, for this type of application, is it required to have a
timing source toprevent the calls being dropped.
Cheers
SW
___
Hi,
I cannot receive any
calls via icoonect. I can make outgoing calls, and also I can see
sipauth.deltathree.com registering me correctly (I am on public internet). When
I try calling-in I wouldn't even get an invite my way. I then hookup a
grandstream ata and without a problem it was
, not so good for * on the public
net ?
Cheers
Sathya
-Original Message-
From: Steven Sokol [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 10, 2004 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice asterisk patch
I can
for this great advice.
Cheers
Sathya
-Original Message-
From: Steve Rubin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 10, 2004 6:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iconnect incoming problems
Sathya Weerasooriya wrote
thought it was previousely in minutes. Besides now we have this
annoying Notice.
If this patch is needed for NAT users as Steave Sokol mentioned, then there
should be an option when this fix get to CVS so that it only applies to
Broadvoice Context when * behind NAT.
Sathya
Hi,
I am using * to function as the voice mail system for Vocal. Since I do not
have a context in sip.conf file for each vocal client, I can't set the
mailbox= in sip.conf. How do I get the MWI to a Vocal client ?
Cheers
Sathya
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. Is there a command to initiate the mysql
module, or just a reload of a config is enough ?
Thanks a bunch
Sathya
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phone I can see following setting;
Voice Frames per TX: 2(up to 10/20/32/64 frames for G711/G726/G723/other
codecs respectively)
Could there be a mismatch here ?
Cheers
Sathya
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:in-audio via RTP (RFC2833) via SIP INFO)
Cheers
Sathya
Date: Wed, 19 Nov 2003 06:15:35 -0600
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Reply-To: [EMAIL PROTECTED]
Jeremy McNamara wrote:
Don't try to do
I am sorry I mean dtmfmode=info
-Original Message-
From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 19, 2003 10:34 AM
To: Eric Wieling; [EMAIL PROTECTED] Digium. Com
Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Hi,
Thanks
, asterisk can handle
pass through.
Cheers
Sathya
-Original Message-
From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 19, 2003 11:01 AM
To: Eric Wieling; [EMAIL PROTECTED] Digium. Com
Subject: RE: Re: [Asterisk-Users] g723 to g723 SIP call - warning
4) Configuration - Which one is simpler and easier, again any pointers other
than readmes., Wiki and John Todd was my reference points for all the above,
but cant find any h323 stuff there.
Thanks in advance for your help.
Cheers
Sathya
___
Asterisk
Hi Gurus,
I we seen references to 'codec pass through feature' in the mailing list.
SIP to SIP and SIP to chan_h323 as well. Could someone help me to understand
this feature, or point me to some examples etc.
Appreciate any pointers here.
Thanks a bunch
Sathya
Hai all,
Iam a newbie to Asterisk.
I want to configure my Asterisk thru Command Line Interface to connect
two internal extensions and two external numbers and calls should
occur between any of the two numbers. Can anybody kindly send me the
configyration details for
extensions.conf anf sip.conf
card systems out there.
Cheers
Sathya
-Original Message-
From: Maxim Litnitsky [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 19, 2005 3:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Billing
http://www.asterisk-support.ru/Members
Hi
friends,
Do we have some
instructions to build an addon module to asterisk ? Lets say I write my
own addon module, addon.c, how do I go about linking this with asterisk. Any
pointers ??
SW
Hi,
I've just initiated
a new bounty for the above;
http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL
Any takers or any
contributors please respond to me privately. I do not know exactly how the
bounty process works, but I can coordinate on this ?
SW
Sathya
Just wondering
whether we have a resolution to iconnect incoming problem, which started
few days ago.
Cheers
SW
.
Thanks
Sathya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Raj
Sent: Monday, August 09, 2004 5:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] iconnect inbound - so do we know how to
fix it
May be you can find the solution in my post
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