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2006 Nov 02
0
testing
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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
2008 Jan 01
2
(no subject)
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2006 Nov 13
3
FW: Desktop integration
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=UTF-8" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000066"> Hi Dean,<br> <br> I will check that site - thanks for the hint.<br> The biggest problem I see with
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. -------------- next
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 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 Apr 23
1
Make an iso image or a kickstart
Dears Can anyone guide me ?? I want to put my asterisk system on an iso image like trixbox ,or how to make a. how can I do that ,I am using centos 4.4 final 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
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?
2013 Aug 27
0
[LLVMdev] unsubscribe
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2009 Jul 20
0
No subject
asterisk -rx 'core show channels' | grep DAHDI | sort -n Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ----- "Mike" <list at virtutel.ca> wrote: > > Hi, I have just recently been using DAHDI, and I wanted to know how to
2008 Nov 16
0
No subject
=========== process 5496: arguments to dbus_connection_ref() were incorrect, assertion "connection != NULL" failed in file dbus-connection.c line 2497. This is normally a bug in some application using the D-Bus library. org.freedesktop.DBus.Error.Spawn.ExecFailed: dbus-launch failed to autolaunch D-Bus session: PuTTY X11 proxy: wrong authentication protocol attemptedAutolaunch error:
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 -------------- next part -------------- An HTML attachment was scrubbed... URL:
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/777a7b83/attachment.htm
2009 Jul 20
0
No subject
asterisk -rx 'core show channels' | grep DAHDI | sort -n Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ----- "Mike" <list at virtutel.ca> wrote: > > Hi, I have just recently been using DAHDI, and I wanted to know how to
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f1043e19/attachment.htm
2009 Jan 16
0
No subject
Telco, location, ect?) At X times of day? =20 Ect, ect. =20 It sounds like bleed over, which can be causes by some many things the best place to start is to find a pattern if there is one. =20 James Shigley Monroe Telephone Answering Service =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC Sent: Tuesday, May
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
2007 Jul 12
0
No subject
"We have created an easy and cost effective way to have customized recordings done quickly and with no hassle." I thought this was rather amusing, as: 1. If you want multiple prompts recorded, you need to submit a new order for each, which means that even prompts of a couple of words are still charged at $12. That is NOT cost effective. You could record all your prompts as a single