_Description_
We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP)
developer with some Asterisk experience who is based in Western or
Eastern Europe or Asia. We can work with an individual or an
organization. You must be fluent in English.
We need you to help expand
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dominique kull
taridium.communications ltd
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Anybody ever managed to implement a solution where one could forward a
voicemail from one * server to another?
Dominique
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server. Works great.
Matthew
- Original Message -
From: Dominique Kull [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004 9:56 AM
Subject: [Asterisk-Users] Voicemail forward to a remote server?
Anybody ever managed to implement a solution where one could forward
attribute behind 60 stands for?
thx
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Thomas Küpper
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dominique kull
taridium.communications ltd
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http
Would be strange if it supported SIPRTP but not UDP... I think that
most SIP support at least 10 protocols:
SIP
SDP
RTP
UDP
DNS
TFTP
DHCP
TCP
IP
ARP
802.3 (Ethernet)
;-)
Dominique
Mike Reed wrote:
1) Who bought Pingtel's phone line?
2) Anyone seen this chinese-made VoIP phone that supports 8
Don't worry - the downgrade is pretty painless. Just change the config
to load the old firmware.
Dominique
Joel Vandal wrote:
All Cisco 7940 that I have upgrade to 7.1 no more try to get the
dialplan and ringlist files from tftp.
Now I must found a way to downgrade from 7.1 to 6.3.
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Joel
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dominique kull
taridium.communications ltd
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e
Why not use a public address for * ? A firewall, if properly configured
can protect your * server the same way as it would with NAT in a DMZ.
Dominique
Bastian Schern wrote:
Hello *,
I try to establish a Asterisk-Server for internal and external usage.
Perfect use case for a DMZ, or not?
My
Bodo Hahnke wrote:
Hello,
I found serveral discussions about the Zyxel ePhone Prestige P2000W and
the WiSip from Pulver Innovations on this mailings list but still have some
questions:
1) are there other affordable wireless SIP Phones on the market? I haven't
seen or found anything else till now
You are right, there is no SIP firmware for the 7920 - SCCP is currently
the only choice for *.
Ray Burkholder wrote:
yet. The only Wireless SIP phone I would use in a productive environment
would be the Cisco 7920.
I don't see a SIP load for the 7920. Are you sure it is SIP enabled?
Ray.
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firmware ?)
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dominique kull
taridium.communications ltd
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dominique kull
taridium.communications ltd
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v: fwd 268167
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dominique kull
taridium.communications ltd
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the drivers might be the same, but I have not tried this.
dominique kull
taridium.communications
the old lodge, london sw6 6ee uk
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=g729
dtmfmode=rfc2833
host=dynamic
nat=yes
qualify=300
canreinvite=no
My phone is set to use DTMF 'outband'
any ideas?
Dominique
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dominique kull, partner
the old lodge, london sw6 6ee uk
t: +44 207 731 1562
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,
With my config (as posted this morning) DTMF works.
I can log onto voicemail by selecting a mailbox number and password
Giles
- Original Message -
From: Dominique Kull [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 24, 2004 12:02 PM
Subject: [Asterisk-Users] ZyXEL Prestige 2000W
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dominique kull, partner
the old lodge, london sw6 6ee uk
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dominique kull, partner
the old lodge, london sw6 6ee uk
t: +44 207 731 1562
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v: fwd 268167
w: http://taridium.com
e: [EMAIL
options visit:
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Dominique Kull
The Old Lodge, London SW6 6EE UK
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Do a search on http://www.voip-info.org/ first. It is the best place for
Asterisk and related stuff.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Cisco%2079XX%20XML%20Services
cheers
Dominique
Matthew John Darnell wrote:
Aloha,
Has anyone written an XML application for a Ciso 7960
been playing around with the Pulver firmware WF.00.11/B.00.13/Apr 07
2004 and its not better in any way. Anbody made some progress with that
issue? I guess we will have to wait for ZyXEL releasing a real
production FW.
cheers
Dominique
Dominique Kull wrote:
Thanks for your replies. The hangup
a formal bug report
procedure
in place with proper tracking but to no avail.
Regards,
Lars...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Dominique
Kull
Sent: 02 June 2004 22:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
Does anybody have any experience with the ZyXEL Prestige 2000W? I am
having problems with the line tear down when I call another extension.
If nobody picks up at the other end when I hangup the 2000W, the other
extension continues to ring. Is there any way to hangup a SIP call if
there is no
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