Displaying 20 results from an estimated 83 matches for "benjk".
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benja
2004 Oct 06
10
Asterisk and SIP phones
I have Asterisk server providing phone service for my company.
The server is behind a PIX-515 FW and is assigned a private address
192.168.11.X/24.
With that said what is best to provide remote SIP phones (home offices)
securely.
If the solution is to put up another Asterisk server with a public IP
address I am opposed to that.
I am looking for the a secure reliable solution to set up remote SIP
2004 Oct 05
4
[OT] Has Sipura support been closed down?
Does anybody out there have any evidence that Sipura support is still
in operation?
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
2004 Sep 23
11
1.0 Mirrors
Hello,
Please be conscious of Digium's bandwidth and use a Mirror when
downloading 1.0. I have mirrored the tarballs at:
ftp://ftp.nacs.net/asterisk/
Direct links:
ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz
ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz
ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz
--
Vice President of N2Net, a New Age Consulting Service, Inc.
2004 Sep 13
4
PABX & VOIP Gateway
Hello,
I'm researching the possibility of using VOIP (SIP) with an existing
PABX system. Ideally, the setup would be to dial an outside line through
the PABX (that would actually link to the the VOIP gateway).
At this point I would prefer not to purchase a hardware-based VOIP
gateway. I would prefer to use a software-based gateway for research &
testing purposes. Could anyone please
2004 Jul 31
3
Asterisk on Sparc64
...ific
information for all OSes on that Wiki.
Is there anybody else who has build Asterisk on Opteron?
Please share your experiences with the community. This is
important!
You should also submit your changes to Digium in order to
get them into the CVS tree. Please contact Mark about
that.
thx
rgds
benjk
--
Sunrise Telephone Systems Ltd
9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
__________________________________________________
GANBARE! NIPPON!
Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE
http://mail.ganbare-nippon.yahoo.co.jp/
2004 Aug 27
2
Are there any graphic designers on this list?
...which I
am grateful for and like to say thank you again.
However, there hasn't been a single response from a
graphic designer to offer help with a custom icon. Are
there any graphic designers on this list at all? If so,
please take a look at the Wiki above and see if you can
help.
thanks
rgds
benjk
--
Sunrise Telephone Systems Ltd
9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
__________________________________________________
GANBARE! NIPPON!
Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE
http://mail.ganbare-nippon.yahoo.co.jp/
2004 Oct 08
5
SPA3000 as a replacement for X100P
I am still haveing problems (echo) with my X100P but I'm thinking it has
more to do with the server it is in which is not a negotiable item at
this time. My question then is to the use of SPA3000's as a replacement
from the FXO standpoint.
1. Can you setup the FXO port to recognize distinctinve ring and call a
different context like you can do with Zap channels? Being able to call
a
2004 Sep 07
2
OT - Experience using Gmail for Asterisk Mailing List
...sk
mailing lists -- and by extension for any other mailing lists, too.
So, if you know somebody who could invite you and you are not happy
using your regular email or some other webmail which breaks the
threading like Yahoo does, I recommend you bother them for an
invitation and try Gmail ;-)
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
2004 Aug 30
1
Voicetronix OpenLine4 immediately hangs up on every call
...9;
> Monitor got null event
> vpb/1-4: Event [12=>[03] Loop Drop]
vpb/1-4: Flushing event [12]=>[03] Loop Drop
> vpb/1-4: handle_notowned: mode=3, event[12][[03]
Loop Drop
]=[0]
> vpb/1-4: handle_notowned: mode=3, [12=>0]
thanks in advance
regards
benjk
--
Sunrise Telephone Systems Ltd
9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
__________________________________________________
GANBARE! NIPPON!
Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE
http://mail.ganbare-nippon.yahoo.co.jp/
2004 Jul 11
20
New Asterisk bounty: SIP simultaneous
...himself is able to do so
4) Caller asks for Mr. Road Warrior, secretary transfers
to internal extension of road warrior notebook's softphone
I am sorry but your bounty doesn't seem to make sense. It
looks more like one of those "Wanted: problem for given
solution" cases.
rgds
benjk
__________________________________________________
Do You Yahoo!?
http://bb.yahoo.co.jp/
2004 Jul 28
1
Please share your Solaris experiences on the Asterisk Solaris Wiki page
...s a
community around BSD, would be very helpful. This will
only happen if Solaris users start sharing their stuff in
a place where others can easily find it.
So, please share your experiences with the community ...
http://www.voip-info.org/tiki-index.php?page=Asterisk+Solaris+Support
thanks
rgds
benjk
--
Sunrise Telephone Systems Ltd
9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
__________________________________________________
GANBARE! NIPPON!
Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE
http://mail.ganbare-nippon.yahoo.co.jp/
2004 Sep 04
1
How do you avoid or reduce false hangups on X100P?
...ven tried
different versions of Asterisk and Zaptel, but the false hangups
persist.
BTW, the customer does not wish to convert his ISDN line to analog
lines for a variety of reasons.
Does anybody know how to adjust the zaptel driver so that false
hangups can be reduced?
thanks in advance
regards
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
2004 Sep 12
2
Overriding SIP From Header
Is there a way to override the SIP From Header that is used in the
extension.conf Dial command? The default is 'asterisk@host'. I do not want
to configure SIP accounts in sip.conf, but instead generate the SIP
From-User within extensions.conf from data the user has entered
interactively. Any idea?
Henrik
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2004 Sep 28
0
FW: FXO question
A better explanation can be found here...
http://www.digium.com/index.php?menu=faq#TDM%20&%20Analog_0
> -----Original Message-----
> From: Benjamin on Asterisk Mailing Lists
> [mailto:benjk.on.asterisk.ml@gmail.com]
> Sent: Monday, September 27, 2004 11:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] FXO question
>
> On Mon, 27 Sep 2004 11:51:16 -0300, Angel Diaz <adiaz@sinergis.com>
> wrote:
> > Hi all...
2004 Sep 06
3
iaxy vs sipura
I need a cheap simple adaptor for analog phones to use with Asterisk. It
should be some kind of "configure and forget" type of device, to use at
the office, or just throw it in a road warrior's bag and use it while
travelling, to call back to the "mothership".
I can't decide between iaxy and sipura. Can you guys help? Which one
would you use? (and why?)
I feel that iaxy
2004 Jul 27
2
g729 + GSM + g723
Folks!
We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found.
Here is the config I have used:
-------------------------------
Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2
User1 is in USA on Broadband Cable
User2 is in India on 64Kbps ISDN Line
User1 using SIPURA SPA 2000
user2 using Xten professsional(X-pro)
2004 Sep 25
2
Asterisk 1.0 & Zaptel 1.0 -- False Hangup Disaster
...n
user decide instead. There should be a setting "hangup=local-only"
that would have the effect that no channel will ever be hung up unless
the (non-Zap) local party has hungup.
As things stand now, we won't be able to deploy this 1.0 release if
Zaptel is required. What a pity.
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
2004 Jul 29
10
Asterisk GUIs at Astricon * REMINDER *
I'm working with the final details of the Astricon agenda. I haven't
got anything so far on Asterisk GUI's and there are plenty of projects
out there. I would like to invite developer's of Asterisk GUI's, both
open source and commercial, to participate.
What I'm thinking of is giving each GUI a slot of 10-15 minutes for
a presentation and then a panel discussion on the GUI
2004 Jul 28
4
X-Lite to Asterisk through NAT?
Hi there,
I have an X-Lite phone on my box and I'm trying to register it with a
remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I
know it's a pain to do because of SIP not working well with NATs, but I
know there are ways to do such a thing...moving the Asterisk box outside
the NAT is not a possibility at the moment. One thing we tried was
setting up a VPN, but
2003 Jul 07
0
Follow-up -- Using Asterisk with Nikotel
Hi
thanks to everybody who has been assisting me in solving the various
problems I had to dial out from Asterisk to a PSTN number with SIP using
Nikotel's VoIP service.
I have drafted a mini-how-to which is available at
http://www.akabeni.com/benjk/Using_Asterisk_with_Nikotel.pdf
This is a first draft, I will amend this further, in particular the
"verify and debug" section which is blank right now.
here is a plain text summary:
in sip.conf ...
------------------------------------------------------
; SIP Registration with Nikot...