Phone Recommendation
Anders Svensson wrote:
We use Grandstream GPX2000 for this. It works ok. Support 11 lines in
basic.
Anders
I also use this phone, have read about the 11 lines, but how does one
'manage' these lines? The first 4 are easy, you have buttons for that,
but how can you use
We use Grandstream GPX2000 for this. It works ok. Support 11 lines in basic.
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Duracom ISP
Lists
Sent: den 12 december 2005 23:36
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
The only one I can think of to decent price level is the Grandstream GXP
2000. Also have headset jack¨
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
Sent: den 28 november 2005 17:27
To: asterisk-users@lists.digium.com
Subject:
http://ipswitchboard.thorben.dk/index.php?option=com_contenttask=viewid=26
Itemid=46
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: den 24 november 2005 20:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
We have tried both but given
up hope about them. So now we only use Quintum DX series. Amazing machine
Anders Svensson Bobas
Communication
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Goran Donev
Sent: den 22 november 2005 16:41
To:
asterisk-users
We changed to newest fw released yesterday and out came a new phone. Solved
a LOT of problems. Perhaps yours to.
Anders Svensson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of George Pajari
Sent: den 22 november 2005 23:15
To: Asterisk Users Mailing
Hi all!
Someone who can recommend a good E1 gateway for
terminating VoIP traffic. H323 or Sip
Regards
Anders Svensson
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Asterisk-Users
: [Asterisk-Users] E1 Gateway
Anders Svensson wrote:
Someone who can recommend a good E1 gateway for terminating VoIP
traffic. H323 or Sip
Asterisk!
/O
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Asterisk
If you mean a just a router without modem we sell Draytek in Europe for
about 170 usd. Built in QS, 32 VPN tunnels etc. Works great. They also have
a bigger model 3300 with up to 8 fxs ports built in. That model also comes
with 4 WAN so you can have redundant lines, loadbalance and so on. I am
I don't think they want to solve it. It's the same with the Sipura boxes.
Only SPA 2100 supports 2 G729 sessions.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard James
Blundell II
Sent: den 10 november 2005 18:40
To: Asterisk Users Mailing List -
We have the PBX 2000
running as a testbox. Works very good. I am pretty sure that is Asterisk
inside. Have not tested the phone. And Planet is a companu that buy most
products OEM from other manufacturers so quality can be different on a
different product
Anders
From:
[EMAIL
gets ring
tone as of no one answer the call. The providers ssw treats the call as
answered and get no errors
Any hints where to start looking?
Regards
Anders Svensson
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Sorry. Forgot to say that
if I connect an ip phone directly to the provider it works without problwm
Anders
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Anders Svensson
Sent: den 8 november 2005 11:09
To:
asterisk-users@lists.digium.com
Subject: [Asterisk
Yes you can connect the fxo to a asterisk using sip
I have cut out a piece of the manual. It works for m
5.2.7 VoIP-to-PSTN Calls
To make a VoIP-to-PSTN call, users need to dial the FXO SIP account phone
number first. A ring tone is played once followed by a dial tone. At this
time, users can
Hi!
How do I uninstall AMP and FOP from my Asterisk?
Regards
Anders Svensson
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I agree. It is like we newbie's on Asterisk is just trouble for the list
members. Pity there is no newbie list. But all were newbie's in the
beginning and not so pompous as some on this list
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
was suggesting to remove it from your linux installation
by using the rm -rf command as under
cd /var/www
rm -rf *
which will efectively remove all the web pages associated with the AMP
instalation.
-Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anders
We have a few satellite trunks for VoIP in Africa and have some experience.
Please mail me off list and we can discuss it
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: den 2 november 2005 18:01
To: Asterisk Users
Price is high that is correct but latency is not correct. We have a number
of Satellite VoIP Trunks in Africa and no location has more then 500 ms
latency. In all locations we have 2 Mbit dedicated lines using C-band and
the hub is in the US. But price is HIGH. 6000 usd per month
Anders
Have you read this?
http://voipspeak.net/index.php?option=c . d=
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley
Sent: den 30 oktober 2005 16:12
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
http://voipspeak.net/index.php?option=com_contenttask=viewid=24Itemid=27
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley
Sent: den 30 oktober 2005 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Only the pricing is not that fantastic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent: den 28 oktober 2005 15:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] GSM cards / mobile phone cards
To: ;tag=as0f1899e3
Call-ID: [EMAIL PROTECTED]
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
Regards
Anders Svensson
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Newbie warning
Hi!
Can I setup an extension that dial out directly to the
phone number I have with my sip provider. Like dial exten 110 and it connects
to my sip phone number
Regards
Anders Svensson
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Look in rtp.conf. You
must have the same udp-ports open as the settings in rtp.conf
Anders
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Furdyk
Sent: den 17 oktober 2005 21:02
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE:
Hi!
I would recommend a 8
port fxs MOSA 3708 from Vodtel. Works perfect with Asterisk and a
reasonable priced compared to Quintum Tenor. Can be bought on www.bobascom.com. The webshop is in Europe but sell equipment to customers all over the world
Anders
From:
[EMAIL
Hi!
Has anyone tested this IAX ATA?
Their free softphone is GREAT
https://www.virbiage.com/products.php
Regards
Anders Svensson
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Discussion
Subject: Re: [Asterisk-Users] IAX
ATA
Are you looking on purchasing one?
francis
www.VoIPware.ca
On 10/13/05, Anders
Svensson [EMAIL PROTECTED]
wrote:
Hi!
Has anyone tested this IAX ATA?
Their free softphone is GREAT
https
Subject: Re: [Asterisk-Users] IAX
ATA
I have other IAX ATA's available at VoIPware.ca - I have tested them
personally and they work great.
thanks,
Francis
www.VoIPware.ca
On 10/13/05, Anders
Svensson [EMAIL PROTECTED]
wrote:
Yes I was interested to test
Hi!
Perhaps newbie but I cant find somewhere to set the
TTL for sip registration when * acts as client
Regards
Anders Svensson
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What you are looking for is a pc2phone dialer. This can be preconfigured
with all settings and when it connects to your * it ask for username and
password or just a pin. There are many of these out on the net. Most is
however locked to a provider but you will also find many that you can buy
with
Hi!
I have problem with my AAH. I have set up a sip
channel. It works perfect both ways with one exception. When someone calls in I
only get 1 signal. The caller have normal ringtone until message is played.
Anyone who can help?
Regards
Anders Svensson
when you have more then 1 provider?
Regards
Anders Svensson
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http://lists.digium.com/mailman/listinfo/asterisk-users
Hi!
What exactly are you looking for regarding the fxo gateway. Perhaps we can
help you
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: den 5 oktober 2005 23:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Anders Svensson
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To UNSUBSCRIBE or update options visit:
http
in the
outbound routing. But * always use the incountry trunk because the 0. dialpattern
is also true for international calls
How to fix this?
Regards
Anders Svensson
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Hi!
Where can a newbie find some info about how to set up
an auto attendant extension?
Regards
Anders Svensson
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Asterisk-Users
Hi!
How do I configure my * to have a remote extension if
the asterisk is behind a nat?
Regards
Anders Svensson
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Asterisk-Users
Just to clarify. These products are not produced by this company, its
Taiwanese brands. The SIP-CPE Gateway is a rebranded VodTel MOSA 3700
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: den 3 oktober 2005 18:12
To:
Hi all!
How do I create a dialpattern in an outgoing rout
that sends all calls starting with 00 AND calls starting with 1-9 to same
trunk
Regards
Anders Svensson
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Hi!
Cant get my Grandstream GXP 2000 to register on my
AAH. All other Grandstream units work fine. Something extra to think about?
Regards
Anders Svensson
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Hi all!
Is it possible to have a setup with a server only
dedicated for transcoding from ulaw/alaw to G729. What is the capacity of a
server like that in simultaneous calls?
Regards
Anders
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will pick uo traffic from 1 Tollfree number
sent to us by IAX with ulaw codec. Must be transcoded to G729 and sent out as
SIP. What is possible capacity on 1 server using required hardware?
Regards
Anders Svensson
CTO
BoBas Communication
Glimminge 2045
S-280 60 Broby
Sweden
Phone: +46 (0)40
Hi!
Finally I have been able to install AAH and its up
and running. I am behind a router and believe I have to configure this
somewhere but cant do this with AMP. Can somebody hint a newbie about how to do
it
Regards
Anders Svensson
-Users] NAT
At 12:14 9/29/2005, Anders Svensson, wrote:
Hi!
Finally I have been able to install AAH and its up and running. I am
behind a router and believe I have to configure this somewhere but cant do
this with AMP. Can somebody hint a newbie about how to do it
You should be able to do
Hi
Thanks!
2 questions
Can externalip be a dns-address?
How do I configure the Incoming settings in the siptrunk? I can call out
using the trunk but get busy tone when I try to dial in. Use AAH
Thanks
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Thanks!!
Problems solved.
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Samy Antoun
Sent: den 29 september 2005 20:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] NAT
Anders,
There are 2 ways to
Here comes a newbi question.
I have got a used Dell
Optiplex GX240 P4 1.7 Ghz with 512 Ram. I have downloaded [EMAIL PROTECTED] and
burnt a bootable CD. But the Computer doesnt start the installation.
What can be wrong? The computer is empty with
anything on the harddrive.
Is it
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory
Anders Svensson wrote:
Here comes a newbi question.
I
installation
Let me ask you one major thing. Look at the CD filesystem on another
computer. Did you perchance burn the ISO as a file on the CD instead of
burning the ISO image to the CD?
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Anders Svensson
-Sent
What provider to use depends of course of witch country you live in. We have
a lot of customers in Africa who use iwayafrica. Many of the providers block
Voip because they have own Voip service. For US we use New Era Systems, Inc
Anders
-Original Message-
From: [EMAIL PROTECTED]
Hi! I asked this question a couple of days ago but
got no answer so I try again.
Is it possible to route a call in * based on used
codec, meaning that if a user use G723 that call is routed to siptrunk 1 and a
user using G.729 is routed to siptrunk 2?
Regards
Anders Svensson
There is also www.talkycallshops.com
Very good rates, no
monthly fee Unlimited number of numbers. Voip DIDs in 22+ countries.
Anders
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nana Tandoh
Sent: den 25 september 2005 12:20
To: Asterisk Users Mailing List
]
[mailto:[EMAIL PROTECTED] On Behalf Of Shawn Rutledge
Sent: den 25 september 2005 23:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Best Voip provider
On 9/25/05, Anders Svensson [EMAIL PROTECTED] wrote:
There is also www.talkycallshops.com
That looks
Hi all. Is it possible to get * to send calls to
different sip trunks depending on what codec the incoming call use? This to
avoid transcoding
Anders
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Hi!
Here comes a newbi question.
I now that transcoding of codecs take a lot of cpu
load. But if I want to receive all traffic as IAX and then want to send it out
as SIP. Is it the same? Requires a lot of CPU and RAM?
Regards
Anders Svensson
This is a good link
http://www.erlang.com/calculator/lipb/
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Journo
Sent: den 23 september 2005 11:20
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Which
codec?
Is there a guy somewhere on
Hi!
Probably another newbie question. Is it possible to
run * on one processor and MySql on the other in a double cpu server?
Anders
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Have you tested Aastra. Works great with * and reasoable pricing
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: den 20 september 2005 20:57
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Cisco Ip phones
Many of the satellite companies block voip because they have the sevice for
sale them selfes. And dedicated satellite internet is VERY expensive. We
arranged a 512/512 connection today for a callcenter in Nigeria and they
will pay 6000 usd per month.
-Original Message-
From: [EMAIL
The MOSA 3700 family from
Vodtel have working T.38. They come from 2 to 16 ports. Can be bought on www.bobascom.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moody
Sent: den 14 september 2005 14:22
To: Asterisk
Users Mailing List - Non-Commercial
Skype prices is not that low. F.ex buying price today for Argentina Buenos
Aires is between 0,0050 and 0,0056 Euro and Skype charge 0,0170 euro.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: den 13 september 2005 11:48
To:
Have you read this
article? Its about Sipura 2000 and Asterisk but have much valuable info.
http://voxilla.com/modules.php?op=modloadname=Newsfile=articlesid=39
Anders
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sander
Sent: den 11 september 2005 09:31
Hi!
What hardware is required for Asterisk to handle 30 simultaneous
calls? All sip to sip.
Regards
Anders Svensson
CTO
BoBas Communication
Glimminge 2045
S-280 60 Broby
Sweden
Phone: +46 (0)40 608 22 50
Cell: +46 (0)703 17 13 06
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED
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