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Sticky:
Group info uk.telecom.voip - monthly posting by
Ivor Jones
This is a monthly posting giving information about the Usenet group
uk.telecom.voip.
If you are a new user, or even if you've been around for some time, please
take a few minutes to read through the charter and bear it in mind when
posting.
Charter for uk.telecom.voip
===================
| | 0 | 115 |
| | Linksys wirelss bridge for ATA by
Roger Barrett
Anyone else got one of these (WBP54G) ? I'm trying to use it with MAC
filtering enabled on my router but the adapter wont connect unless I disable
MAC filtering (and yes I have entered the MAC addresses of the bridge and
the SPA in my router table).
Thanks
Roger.
| | 5 | 21 |
| | VoIP call quality falls - Survey by
KnowingAbout.com
VoIP call quality has taken a nosedive over the last 18 months, with
nearly 20 percent of VoIP callers experiencing unacceptable voice
quality, according to a survey done by Brix Networks.
The results come from Brix Networks' TestYourVoIP.com site, a free
VoIP quality-testing portal. Brix claims that nearly one million VoIP
phone tests have been conducted at the site since its March,...
| | 0 | 21 |
| | [03 geographic range] Ofcom plans future of UK Telephone Numbering by
Solario
Paul Cupis wrote:
Ofcom plans future of UK Telephone Numbering
http://www.ofcom.org.uk/media/news/2006/07/nr_20060727
*******
Ofcom will introduce new UK-wide 03 numbers from early next year. Calls
to 03 numbers will cost the same as calls to geographic numbers, and be
included as part of any inclusive call minutes or discount schemes for
geographic calls. This will apply to calls...
| | 0 | 18 |
| | Free SIP Softphone with IM features (MSN/Yahoo/ICQ/AOL etc) by
www.tpad.com
Free Softphone from www.tpad.com with Instant Messaging Features which
allow you to chat to mulitple networks (MSN/YAHOO/ICQ/AOL etc)
Other features include:
Free Calls - PC to PC
PSTN Breakout to International / UK Landlines - very cheap rates
Free SIP Telephone Number
Send Multimedia Files to other contacts
Conference Chat
| | 0 | 33 |
| | Problems with IPKall..? by
Ivor Jones
Anyone else having problems with incoming calls from an IPKall number this
morning..? One of my two is working ok but the other is reporting that my
end is busy, which it isn't..!
The annoying thing is callers get charged to hear the busy message so
hopefully it will be fixed soon, but I'd be interested to know if it's
just me or not.
Ivor
( 1 2 ) | | 11 | 35 |
| | Asterisk Hardware by
Dave {Reply Address in.Sig}
Any pointers to cheap FXS/FXO hardware for an Asterisk system? I'm
basically after something for home use (i.e. playing and learning) that
can handle a couple of incoming PSTN lines and four extensions. I assume
that for VoIP lines I just need my existing broadband and ethernet
harwdare. Digium stuff seems expensive even on eBay, at least when I've
looked, so pointers to other manufacturers would...
| | 9 | 28 |
| | | | 2 | 17 |
| | Skype and Talk Talk by
Sacred
Hi All
I have just been told by Carphone Warehouse that their Talk Talk 3 (free
broadband) offering is not compatible with Skype.
Can anyone advise if this is fact? Amd is it therefore not compatible with
any voip products?
Many thanks
| | 4 | 21 |
| | ATA3000 drops inbound PSTN calls by
Martin
I have a Linksys/Sipura ATA3000, set up so that I can use the existing
house phones to make or receive both PSTN calls and VOIP calls.
Recently, I've found that when people call me on the PSTN number,
calls get dropped after 2 to 6 minutes. I suddenly get an
unobtainable tone, while callers get a click followed by nothing.
It's always possible that it's a BT problem - but a quick test call...
|
July 26th 06 10:15 PM
by Martin | 2 | 35 |
| | SIP & playlists by
kael
Hi,
Is it possible to manage a playlist on a SIP server from a SIP client ?
Thanks.
--
kael
|
July 26th 06 02:43 PM
by kael | 0 | 25 |
| | Asterisk@home 2.8/freepbx has stopped creating extension by
linker3000
Hi folks,
All of a sudden, our AAH 2.8 server has started to sulk and if we add a
new extension in freepbx, it shows on the list but not in the flash
panel and SIP phones cannot register to it. Looking at the extensions
and sip conf files, the right entries aren't being made, but I can add
extensions manually, although I presume that if I subsequently use
freepbx to change anything I will...
| | 3 | 21 |
| | Wi-Fi phones, any good? by
divoch
I am not using VOIP yet so please bear with me.
I came across NETGEAR SPH101 Skype WiFi Phone, UTStarcom UTS F1000 WiFi IP
Phone Wi-Fi phones. Use of Wi-Fi phones seems to me potentially useful
development but I am rather unsure about
a] compatibility of these or any other similar devices with current VOIP
services. What do I need to look for to ensure compatibility or would these
work...
( 1 2 3 ) | | 28 | 38 |
| | Pulse-to-DTMF converter for UK (Linksys ATA3000) by
Martin
Hi,
Does anyone here happen to know of a UK supplier for a pulse-dial to
tone-dial adaptor?
I like the idea of originating VOIP calls with a 1930's vintage phone,
but of course the Sipura/Linksys SIP ATA only understands DTMF.
Thanks for any info!
|
July 23rd 06 10:16 PM
by Martin | 4 | 23 |
| | | | 9 | 31 |
| | Asterisk advice and costs by
Les Desser
We are a volunteer emergency first responder group and currently share a
Siemens switchboard with a company, that provides us with the features
below (except recording)
We need to set up our own independent switchboard. To replicate the
existing one, even with a reduced configuration is likely to be very
expensive and so we are looking for alternatives.
Our needs a-
( 1 2 3 ... Last Page) | | 33 | 70 |
| | Voip PBX,Private Phone Systems,PBX Telephone Systems, Business Phone Systems by
broadbandera@gmail.com
Introduction,Distributed Workforce,Virtual Organization,Field Offices
for Larger Concerns,Virtual PBX,Case Study-Centract,Knowledge Worker
Impact Quotient,Conclusions
http://flying-rugs.com/virtual-pbx/
| | 4 | 23 |
| | Dialing Plan Help by
PhilÅ
lol
I am useless at this..
I had some great help setting this up about a year ago.
Current is...
(0:0044x. | :0044151xxxxxxS0)
( 1 2 ) |
July 23rd 06 09:34 AM
by news | 14 | 49 |
| | Choose To Refuse by
PajaP
The one thing I am likely to miss most now I am moving to VoIP as my
only telephone access is BT's 'Choose To Refuse service'.
I have a PAP2T and accounts with Gradwell and Sipgate.
Is there any way I can implement a similar set-up?
--
Thx.
PajaP
|
July 23rd 06 12:11 AM
by Jono | 9 | 111 |
| | Sipura SPA1001 by
Jono
I've been successfully using the following dialplan on Line 1 of the
SPA1001
|:01274xxxxxS0|0xxxxxxxxxS0|xx.)
Line 1 is configured for voip.co.uk, Line 2 is Sipgate.
Basically, I want calls to 123456 & 123457 to go via Sipgate (as they
would be free at all times), hence the @sipgate elements.
|
July 22nd 06 10:58 PM
by Jono | 2 | 25 |
| | Dial Plan by
Woggle
I've setup a separate gateway (voipstunt) on my SPA 3000 which I want
to be able to use only when I make a call starting:
00 27
Which would be the call from the UK to South Africa.
I currently use voip.co.uk but want to use voipstun for calls starting
00 27
| | 2 | 39 |