Displaying 20 results from an estimated 2000 matches similar to: "Is this server sufficient?"
2005 Sep 26
2
What ISDN hardware would you recommend?
Trying again...
*Summary:*
I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE
mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode;
What card(s) should I put in to these servers?
*The long story:*
I have 3 locations I want to connect using (*) servers.
1 of those has a single BRI with a Siemens DECT PABX.
1 of those has two BRI's with 2 Siemens DECT
2005 Aug 09
1
Com-On-Air (PCI/PCMCIA) chan drivers?
Does anybody know whether somebody ever implemented (linux) drivers or a
chan component for the Com-On-Air cards, or is the only way to make it
work the use of a Windows box as Dect to SIP gateway for *? (Or does it
work at all?)
I can still get the cards, which would be great to retire the old DECT
based PBX, but it would be of little use if it's not going to work, now is
it? ;-)
--
2005 Jun 21
4
Grandstream 100 pricing question
Hi All,
I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
pricing for these in Europe, so I'd like to hear from people here whether
that is a reasonable price for them?
TIA & BRgds
--
Francesco Peeters
----
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If your program doesn't recognize my signature, please visit
2005 Jun 27
0
Just let whiners whine... Please?
There'll be whiners on all lists, and no list is ever going to be pleasing
everybody... (And everyone should decide for themselves whether or not
they belong to that group, as I will not be mentioning names! I'm sure
there are those who'll think *I* am a whiner...)
So at some point just let it be and continue the thread. Keeping on and on
about it only increases noise to post
2005 Sep 20
0
What hardware would you recommend?
I have 3 locations I want to connect using (*) servers.
1 of those has a single BRI with a Siemens DECT PABX.
1 of those has two BRI's with 2 Siemens DECT PABX's, each serving a
different area.
1 of those has two BRI's and a 2 port Nova Compact PABX with DECT
First step would be to set up the (*) servers and have them
interconnected. When all of that works we'd go on to connect
2005 Aug 27
3
How to use * and # as part of number in dial command
Hi all,
I am struggling with the following and I can't get it work:
In the Netherlands where I live it is possible to use special codes
(aka vertical service codes) to set special 'behaviour' of phonecalls.
So e.g. when I want to dial out with a normal phone and I dial
*31*<phonenumber to dial> then it will turn off my numberindication
(CID) at the called party. They seem to
2004 Nov 28
17
Wiki down?
Hi All,
The wiki seems to be struggling this evening. Anyone else seeing this?
Michael
--
Michael Graves mgraves@pixelpower.com
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. mgraves@mstvp.com
o713-861-4005
o800-905-6412
c713-201-1262
2006 Jun 28
2
2 or more ISDN cards: which comes first ??
Hello,
I have setup an asterisk server with two EICON cards, a 4BRI and 2FX.
How do I know, which card is the first, so that I can setup capi.conf with the
right entries?
Thanks for your help,
Stefan
--
********************************************
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
2005 Jun 15
1
CellPhone BlueTooth adapater with Wireless Profile ??
All
Any body know of a generic bluetooth adpater for the
universal 2.5mm headset jack on a cell phone that supports
the wireless profile *NOT* the headset profile
I know jabra has the A210
http://www.jabra.com/JabraCMS/NA/EN/MainMenu/Products/Accessories/JabraA210/
JabraA210
but it only support the headset profile ..
I am trying to shoe horn my current braindead cell into DocknTalk
2005 Jun 15
1
This mailing list is being spam filtered on my site.
Sorry if this not the right place to post this.... BUT...
Since May 31st, ALL of these user list messages have been filtered by
"spamassassin" running on my Linux box. - Claim to be listed in "Bayes" as
spam. - Have no clue why this is happening.
Luckily, "spamassassin" sent the messages to the "probably-spam" folder on
the Linux box & I was able to
2005 Aug 27
1
Calling PSTN lines from VOIP softphone
Folks,
I am a newbie to the VOIP world and have a question (might as well
sound silly to some). I would like to set up a PC-to-Phone call from
my desktop to a regular PSTN number. Does the Asterisk PBX itself act
as a VOIP-PSTN gateway or do I have to subscribe to a VOIP provider
for this? Are there any free IP-PSTN gateways which I can subscribe
to? Do I need any specific hardware to set up such
2005 Sep 27
2
One-way audio with VPN
I've got a one-way audio problem, but I've looked through a few
documents on the subject and I'm not sure that it's the same issue.
User A calls a local Asterisk user B via a public SIP gateway
(voiptalk.org) using (sip:110@siptest.dmclub.net)
B is connected to the Asterisk server via VPN
B is registered (and has successful bi-directional conversations with
other users on the
2005 Sep 07
1
IAXy - no dailtone
I have a brand new IAXy I'm playing with. I do not get a dialtone on
the phone, or any response at ll on the phone. No sound, no dialing, no
ringing. The phone and wire are tested and known to be good. I think I
have it setup correctly. When I give the iaxprov command I get this:
#iaxyprov 192.168.1.90 iaxy.conf
02:
c0 a8 01 5a
05:
11 d9
03:
ff ff ff 00
04:
2005 Sep 27
3
analogue phone with asterisk
I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a
2005 Sep 19
1
Re: Welcome to the "Asterisk-Users" mailing list
"Unable to login to the MySQL database 'tiki' on 'localhost' as user
'tiki'
Go here to begin the installation process, if you haven't done so
already.
Lost connection to MySQL server during query"
Hi, Everyone:
What is the URL of the "list archive"?
The 'wiki' seems broken.
I want to connect to a single home POTS from an internet
2006 Feb 05
1
1 ISDN BRI to IAX2/SIP... (*) best tool or?...
I have a question,
I have to provide a solution for an office that will be almost abandoned,
and there will be one or sometimes two persons 2 days a week. The main
number however should be preserved.
They have several ISDN BRI connections, most of which will be dropped.
Only one will be retained, for 2 reasons:
1) It has the ADSL link
2) The number has been the main contact number for over 20
2006 Mar 24
3
Call terminated after 60 seconds
Hello,
I switched from my PSTN provider to a voip provider. (Voicedata in
the Netherlands)
>From the moment i switched all inbound calls are terminated after
aproximatly 1 minute.
The provider tells me it's not their issue since I have no other
configuration than all their other users.
What can I do.
I removed all asterisk functionality by forwarding the inboud call
directly to a local
2007 Mar 24
2
freepbx -> DB Error messages...
Hi all,
I am probably missing something ultimately obvious, but I have a problem
configuring freepbx...
Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu
installation guide on freepbx.org.
System pxe-boots from a server with NFS root on same
Using * 1.2 current (from source, not .deb's)
Using mISDN-streams (from source, not .deb's)
Using freePBX-2.2.1 (from source, not
2005 Dec 21
1
RE: Prototype: correct useage of onCompletewithAjax.PeriodicalUpdater
The only call that PeriodicalUpdater will make periodically to you is
the insertion call.
I think of onComplete as being ''I''ve done everything I was going to do'',
which for Updater is right after it''s done a single update, but for
PeriodicalUpdater, it''s after it''s finished all of its updates, so
you''ll only get it once for any call
2006 Jan 10
3
IAX & CallerID
Hi All
Apologises if this has been disussed and I missed it.
My SetUp
I have a sip phone registered to an asterisk box (a1) in one location 1.
This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone.
My Problem
The caller ID setup in the sip.conf for the phone registered to a1 is not