Displaying 13 results from an estimated 13 matches for "varaha".
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2007 Sep 14
0
Speex echo canceller creating some problems. No voice coming.
...100,256,300 etc but I found no success here.
Can anyone please suggest me anything? I am using WM5 devices on both end.
Is there anything that this echo canceller will work with speex codec only?
Do I need to do anything special that I am missing?
Regards,
Digish Gabhawala [mailto: digish@varaha.com]
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2008 Jan 31
1
Incoming call from SIP proxy to asterisk
Hi,
I have asterisk register two users (client-1, client-2) with a SIP proxy.
I have the same two SIP client registered with asterisk. Now my dial plan
setup is such that any call from client-1/client-2 is forwarded to the SIP
proxy and the SIP proxy then takes the routing decision. Calls coming from
SIP proxy will dial out the respective user. Asterisk is required to stay in
the signaling as
2007 Dec 27
1
SIP Channel jitter buffer issue
Hi,
I have a SIP client which is registered to asterisk. Asterisk is
registered to a SIP trunk and also handles the media. Now since my client
has some issues in its RTP Tx, which seems to have some amount of jitter
(mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and
max delta is 85 ms), to over come that I have enabled jitter buffer in the
SIP channel by setting sip.conf
2008 Jan 17
1
asterisk-users Digest, Vol 42, Issue 51
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> URL:
> http://lists.digium.com/pipermail/asterisk-users/attachments/20080114/3f3aca82/attachment-0001.htm
>
> ------------------------------
>
> Message: 9
> Date: Tue, 15 Jan 2008 06:22:27 +0530
> From: "Mayur" <mninama at varaha.com>
> Subject: Re: [asterisk-users] Asterisk RFC2833 to SIP INFO DTMF
> conversion erros.
> To: <david.cantera at IBSOneCall.com>, "'Asterisk Users Mailing List -
> Non-Commercial Discussion'" <asterisk-users at lists.digium.com>
> Message-ID:
> &...
2004 Sep 17
3
how to get caller ID
i cannot see caller ID of the call originated from outside zap channel.
i hv configured both zapata.conf and extensions.conf.
i m right now in india
i think asterisk only supports Bellcore enable caller ID.
so is it the same bug of BT caller ID problem in UK?
or it is the bug of my asterisk configuration?
i hv enabled callerID from my TELCO.
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An HTML
2004 Sep 17
1
?
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2800.1106" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial
2008 Feb 05
0
Asterisk does not handle INVITE authentication by Proxy
Hi,
I have used asterisk 1.4.17 to interwork with a SIP Proxy. Asterisk acts
as a UAC and registers three users with the SIP proxy. It handles the proxy
authentication for register requests well and all three users get registered
with the SIP Proxy. However when the Proxy challenges the INVITE sent by
asterisk, asterisk is unable to use the credentials for each of those users.
If I use
2008 Feb 21
2
Allow INVITE for hold to pass through
Hi,
I would like to configure asterisk to allow INVITE for hold to pass
through it and not provide music on hold by itself. Can anyone help me out
here?
Regards,
Mayur
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2011 Apr 14
0
Asterisk modifies from header
Hi,
I have configured few users in my system with '-' E.g. 1000-1001,
1000-1002 etc..
I am successfully able to registered as 1000-1001 and dials number
937575346. The other end sees 10001001 (missing '-') as a caller.
I did verified in the packet capture surprisingly it is asterisk
which did this.
So, here calling is not a problem and my extensions.conf is bare
simple.
Can
2008 Jan 08
1
Early media support for Asterisk behind NAT
Hi,
I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk
for PSTN calling. Asterisk is configured to support nat with nat=yes in
sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media
using 183 Session Progress. So If I call a PSTN number which has IVR message
played before the call is connected (via 183), those media RTP packets do
not reach the
2004 Sep 19
1
How To get response of command from another socket
hi
i logged on to manager API from other terminal
by
telnet IPADDR 5038
now logged in with username mark
let's say this connection Window A
now i opened another connection with Manager API with same usename
lets say this window B
now if i give a command like originate,Redirect
through window A connection ,
can i able to see its
response:success/failure
Originate:failed/succesfully
2008 Jan 12
2
Asterisk RFC2833 to SIP INFO DTMF conversion erros.
Hi,
I am using asterisk 1.4.17 which is connected to a SIP trunk supporting
rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have
set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for
SIP clients I have set dtmfmode=info. So when I make a call to a cell number
using the sip trunk and then press digits I can see the 2833 dtmf events
coming to asterisk