Displaying 20 results from an estimated 1000 matches similar to: "Make an iso image or a kickstart"
2007 Apr 04
2
Remastering asterisk
Anyone have an idea to re master centos,in other worlds I have an asterisk
on centos with all libraries and modules,how can I make it as an iso image
?
Regards
*********************************************
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of
2007 Aug 28
2
Black screen when maximazing
After opening a few windows, some start to open completely black. So I
have a frame, and a black rectangle. If I resize the window some
display ok, some don't. if I maximize them again, they go black
What is causing this? Does anyone know how to fix this?
I am running compiz + Ubuntu 7.10 gutsy
Xeon 2.4ghz + nvidia quadro nvs 280
Thanks
IMPORTANT NOTICE:
This message may
2007 Dec 10
2
asterisk linkedin group
asterisk linkedin group
I have created an asterisk linkedin group for anyone interested.
http://www.linkedin.com/e/gis/45252/66270A773F53
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
HIROTEC AMERICA
Board member of
Connectech Greater Detroit
www.connectech.org
________________________________
Please visit us on the web at www.hirotecamerica.com
HIROTEC AMERICA Ph.
2009 Oct 29
5
Dynamic DNS trunk
I have a trunk, and its host=dynamic dns.
The problem is, when the IP changes the
Sip show peers
Still show the old IP of the DNS, I have to reload and save the
configuration again so that asterisk recognize the new IP of the DNS.
Any idea how to automate such a thing? Or how can I keep asterisk to deal
with NAMES as NAMES, and IPs as IPs.
Let me know.
Thanks.
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2007 Jun 26
2
number of samples in input_frame
Hi all
Sorry if this is a dumb question: does the input_frame passed to
speex_encode_int *have* to be frame_size samples long?
e.g., If I only have 100 samples left to encode (which is less than the
frame_size of 160 samples), can I just use an array that contains 100
samples, or do I need to create an array containing the 100 "real"
samples plus 60 null samples at the end?
2005 Aug 22
1
Hangup Faster
Hello -
My single line extension users (connected via channel banks) need to be
able to hang up faster. If they just flash the hook it doesn't
disconnect right away. Any ideas on how to resolve this?
Thanks,
Dave
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2006 May 18
2
VoiceMail Groups
Has anyone seen good scripts or documentation on Voicemail groups? We are
looking to have a system where you can send a voicemail to multiple
mailboxes.
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2004 Sep 21
2
SIP termination in Brazil
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP termination solution.
But who as it up and running ?
Best regards,
Han
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2008 Jan 22
9
VNIC, non-global zone, dhcp & dns
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Bitstream Vera Sans"><br>
Hello,<br>
<br>
I am trying to have dns <b>automatically</b> configured through
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
2006 Mar 31
1
Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.
Thanks.
I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try
to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction.
-David
________________________________
From: asterisk-users-bounces@lists.digium.com
2007 Nov 28
1
Digium TE120P versus Sangoma A101D-X
Hello List,
We purchased a TE120P card from Digium and it works great. The only
problem is that we are still experiencing echo on some calls. I've tried
various echo cancellers (right now we are using OSLEC) and still no
luck.
My question has anyone gone from the TE120P to a Sangoma A101D-X Single
Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference?
Also I called
2007 Nov 08
3
Asterisk as a SIP to XMPP Jingle voice gateway
Hello,
I'm looking for a SIP to XMPP Jingle voice gateway.
I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client.
Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa?
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
2006 Mar 23
1
spam filtering with amavis
I'm filtering that is being deliverd to postfix mail server with
amavisd-new .
I want spam with spam f level 1 - 8 to ad a tag any everything above to
be delete is this posebol?
If yes how?
Met vriendelijk groet,
Bas van Dikkenberg
GISkit bv
BFVD1-RIPE
Tel: +3130-6340430
Fax: +3130-6342433
Prive Tel: +3130-6372769
Mob:
2006 Jan 09
8
Pri Gateway Hardware
Does anyone have any experience using a PRI gateway, I am looking for a way
to have multiple asterisk boxes use one PRI, and send that over the network.
I herd there are copper gateway devices (like a X100P card, only it
registers with asterisk using sip, and it doesn't have to be physically
connected to the box) Does anyone have any experience with a PRI gateway?
And could tell me the cost
2005 Mar 29
3
help w/ basics
Hello, I am new to Asterisk and new to this list. I got Asterisk setup and
running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone
to test out.
I cannot get the phone to talk to the Asterisk box. On bootup of the phone,
it tells me that it cannot contact boot server. Why is that? It gets an IP
fine, and I have also tried manually setting the IP of the phone and the
Asterisk
2011 Jan 15
14
Top Posting
Bruce et al.
I'm posting a new thread with the "Top Posting" subject so I won't draw
complaints about "hijacking" the 4-port thread.
Top Posting refers to the practice of sending a message with a reply at the
top and including the entire thread below the reply. I prefer this. If I'm
actively following a thread, the most-recent information appears at the top
2003 Mar 07
70
unsubscribe
Gautham Kasinath
Software Engineer
Arkin Systems Pvt Ltd
T. Nagar
Chennai
Ph. (91) (44) 8216686 Extn 14
2006 Nov 02
0
testing
<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:dt="uuid:C2F41010-65B3-11d1-A29F-00AA00C14882" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<META
2007 Jun 26
0
number of samples in input_frame
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
That is a dumb question :)<br>
<br>
The encoder expects SPEEX_GET_FRAME_SIZE at all times. If you are