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20050406
2005 Sep 06
4
Sipura Devices and Asterisk?
...;m mainly looking at the Sipura SPA sets
since they are the base of the pap2. Anyone else have experience using them,
and which one?
Thanks
Sherwood McGowan
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2005 Sep 06
2
Wireless router with built-in VOIP(FXS) ports for Ansterisk
...t all these products
are bundled with VOIP Service providers (vonage, lindo and at&t) .
Please sugest if any products avaiable in market.
Thanks,
Karun
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2005 Sep 07
0
CentOS-announce Digest, Vol 7, Issue 4
...en)
----------------------------------------------------------------------
Message: 1
Date: Tue, 6 Sep 2005 17:56:55 +0200
From: Tru Huynh <tru at centos.org>
Subject: [CentOS-announce] CESA-2005:756 Low CentOS 3 i386 cvs -
security update
To: centos-announce at centos.org
Message-ID: <20050906155655.GA23603 at sillage.bis.pasteur.fr>
Content-Type: text/plain; charset="us-ascii"
CentOS Errata and Security Advisory CESA-2005:756
cvs security update for CentOS 3 i386:
https://rhn.redhat.com/errata/RHSA-2005-756.html
The following updated file has been uploaded and is current...
2005 Sep 06
1
Routing depending on sip response code?
...iest
on how to make asterisk route depending on the status code returned over SIP
(483, Busy Here?). . . anyone know how to do this?
Thanks
Sherwood McGowan
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2005 Sep 06
3
TE406P audio drops
...t a time. We had about 100 of these small audio drops yesterday
randomly across a total of 14,000 calls.
Any help fixing this would be appreciated,
MATT---
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2005 Sep 06
1
Threeway calling uses up two FXO lines
...ng Hangup("Zap/3-1", "") in new stack
== Spawn extension (internal, h, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
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2005 Sep 05
1
AEC in speex
Hi
I was trying out the AEC in speex.
It would be helpful if some of you can answer these querries
1.After the error values are computated in teh daptation process a check -if
filter is completely mis-adapted - is performed. How could you explain the
criteria used for this check.
2.what exactly is meant by over-cancellation and under-cancellation ?
3.Also it was seen that without the
2005 Sep 05
1
Supported DSPs
Jean-Marc Valin wrote:
> I don't know all the details, but here's a (partial) list of archs on
> which I've heard of Speex running. I'm sure there are others (especially
> the float version should really run on any chip with an FPU).
>
> float:
> x86/x86-64 (SSE assembly optimizations provided)
> PowerPC
> SPARC
I've had floating decoding running on a
2005 Sep 05
0
[PATCH] fix gcc 3.4 warnings
...lin@USherbrooke.ca>
Universit? de Sherbrooke
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2005 Sep 06
0
WMP blues :^(
...http://www.gift-culture.org
initstring: http://www.initstring.org
artificial music machine: http://www.artificialmusicmachine.com
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2005 Sep 06
0
ssh launched from xterm still running after xterm killed on Cygwin
...tect
Ximbiot <http://ximbiot.com>
v: +1 717.579.6168
f: +1 717.234.3125
<mailto:derek at ximbiot.com>
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2005 Sep 06
1
moving fakeroot
...if there's a way of moving a fakeroot directory (that
was created with winetools) and how to do it.
I'd have to move it to another partition.
Thanks,
Dom
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2005 Sep 06
0
.dll and resolution error when run C&C Generals
...ut changing the resolution for a couple of
the games that I've installed (none of which work yet). Do I need to
configure Wine some more? Thanks for the help!
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2005 Aug 25
1
callerid...
Hi, asterisk Users, sorry for my bad English.
im really newbie with this excellent pbx. But I 've a problem with callerid
num when I recive a call from PSTN.
PSTN-> SipGateWay(Welltech3504)-> Asterisk-> BT100
How can I configure my asterisk to receive the callerid from callers and not
the callerid from the extension of the SipGAteway.
Extension of Gateway (sip.conf)
[115]
2005 Sep 04
1
Unable to hear.
Hi all,
Has anyone had problems with not being able to hear callers and them not
being able to hear you? And had any success on how to fix it? Our call
centre staff are complaining that this is a continual problem.
Appreciate any thoughts on this.
Regards
Jennifer Hales
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2005 Sep 05
3
GotoIf sample...
hi everyone. can anyone provide me concrete examples on how to use the GotoIf application? can't figure out how to use it in my dialplan coz im having errors....thanks! : )
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2005 Sep 05
0
atxfer featuremap
...confusing for users with different phones to transfer a call i know you
can use the transfer button but i wan't to use a code *1
not possible???
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2005 Sep 06
1
TDM 400p
...several locations. Once and again I'll need
to restart the drivers because there is no ringing.
Does anyone know a fix for this problem?
Greatings
Johan
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2005 Sep 06
1
Asterisk as SIP/H.323 Signalling Gateway
Hi,
I am wondering whether I can use Asterisk as SIP/H.323 Signalling Gateway.
The setup I envisage looks as follows:
H.323 end-point ---------(ETH)--------- Asterisk
---------(ETH)--------- SIP Proxy/Registrar ---------(ETH)---------
SIP end-point
(ETH: Ethernet)
In principle, Asterisk would just be used to integrate H.323 end-points
into a fully SIP-based core-network. Hence, there
2005 Sep 06
1
Application rxfax missing ?
Hello.
I just emerged spanDSP with all the packages needed. After a bit of
configuration i was read to test.
But i get this errormessage stating that application rxfax was not
found. I could't fint rxfax i teh modules directory.
I use asterisk 1.0.7.
I did reset the server after emerging SpanDSP
I use gentoo kernel 2.6
I don't know what else to do.
Regards,
Arne Morten