I've been using voiptalk.org for about two years now and have never had
any problems. I've been using them for my outgoing business calls for a
year and am starting to use them for some incoming calls, which is some
indication of my comfortableness with their service.
My reluctance to move
I'm extremely pleased with the ZyWall 35 (ZyXel) - well worth it at
about GBP350. The Linksys and Netgear dual-WAN routers are a pile of
rubbish.
The ZyWall has good management tools and a lot of options. Rock solid
in use.
Todd- Asterisk wrote:
I've been looking for this as well.. I
With respect, I don't think you understand the dynamics of growing a
business.
If we are all to benefit from the continued development of Asterisk then
it is in our own best interests for Digium to succeed, because their
success is for our benefit. Your posting is unfortunate as it
I've spent some time now trying to find information on the changes made
to the Asterisk config files. I want to upgrade an old installation to
the latest version, which of course now uses phone.conf. If anyone
could point me in the direction of a set of upgrade notes so that I can
work out
Most modern installations/buildings are wired with RJ45, as are the patch panels. RJ12 is a real pain - I had to chop up patch leads and put RJ12 sockets on the end. Very messy and a waste of time.
On Sun, 6 Nov 2005 22:04:48 -0500, Andrew Kohlsmith wrote: On Sunday 06 November 2005 21:46,
Take this silly argument off-line please.
On Sun, 27 Nov 2005 16:28:27 -, Senad Jordanovic wrote: Seshu, So, now you are not "Seshu Kanuri" any more but "Pbxware Swithware"? Since you are not working or associated with our company I need to ask you not to use "Pbxware, Switchware" in your
/ipDialogSipToneIIISpecs.pdf;
On Mon, 4 Apr 2005 21:05:47 +0100, George Gardiner wrote:
I've just purchased a Netgear FS108P switch which has PoE (802.3af
standard) on 4 ports. According to some websites the SipTone II
supposedly is compliant with 802.3af - IPDialog just says that it
supports power over ethernet
I've just purchased a Netgear FS108P switch which has PoE (802.3af standard) on
4 ports. According to some websites the SipTone II supposedly is compliant
with 802.3af - IPDialog just says that it supports power over ethernet.
Powerdsine list the SipTone II as 802.3af as well.
Can I get it
On Mon, 14 Feb 2005 15:13:37 -, Patrick Lidstone (Personal E-mail) wrote:
Hi there
Just a general question, has anybody experienced any problems
with any Digium telephony cards in the UK, specifically with BT
(British Telecom) lines. I just want to make sure there are no
compatibility
I've received my Com-on-Air DECT card this morning and while I have configured
it (both the DECT registration and SIP to Asterisk configuration), I am running
into a Call missing Call ID from error message. I can call from a DECT
handset to an extension, but not the reverse.
Not being an
I've been scratching my head for a while and I expect it is my mediocre
knowledge of Asterisk which is holding me back. If anyone can assist me with
some pointers I'd be grateful.
Basically, I've hooked up a Viking intercom at the front door. It hooks into
an fxs as a phone. Up till now
On Thu, 18 Nov 2004 14:37:15 +1100, Adam Goryachev wrote:
Just wondering if anyone has used either of these motherboard with
a TE405p. My current board is causing problems, and I'm looking to
replace it...
gigabyte GA-7NF-RZ
gigabyte GA-7N400 Pro2
Thanks,
Adam
Not a direct answer I'm
I have read the the various views on top/bottom posting and it seems to me that
the proper thing to do is:
FIRSTLY, snip as much of the original e-mail as you can,
SECONDLY, reply in-line so that your answers/points are immediately below the
original questions/points,
THIRDLY, having snipped
I must admit I live in perpetual fear of forgetting to switch of html or rtf
formatting (useful for work) and top posting. I can understand the issue with
the former but can see absolutely no reason why top posting is such a problem.
In fact I'd far prefer it. I get to my e-mail in batches
On Wed, 3 Nov 2004 13:10:57 +, [EMAIL PROTECTED] wrote:
On Tue, Nov 02, 2004 at 10:58:39PM +, StrUK wrote:
snip other information
I guess my question is: does anyone have polarity reversal hangup
detection working on a BT line with an fxo module in a TDM400P?
Testing with my fxo
Sometime ago I posted a query about CallerID not working on one of my analog channels
on my BT Home Highway ISDN. It subsequently turns out that I only had CallerID
enabled on one of the analog numbers, not both. Entirely my fault for not checking
both lines.
Now that CallerID is enabled on
I would be grateful for any pointers in the right direction. In short, I get CallerID
to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101
to display anything other than the phone's own number.
I've been through the Wiki and have followed (I think) the
Thanks to everyone for their help.
I sorted out my CallerID problem - I had a stray fromuser=101 command in my sip.conf
which was overwriting any CallerID info. It was a process of elimination (on my part)
helped by all the comments I had back.
Regards,
George
I'd be grateful for some help on this. I've been following the various e-mails on the
UK CID issue, particularly the last posting in bug fix 9.
It seems that all I need now is to apply ast-UK-and-DTMF-pol-CID.diff.
I apologise in advance for what is probably a very simple question, but how do
Impedance setting in the UK!? OK, I've clearly missed something along the way. A
search on the Wiki says something about Zcomplex impedance but I have absolutely no
idea where or what this is. If someone could point me in the right direction I'd be
extremely grateful.
Many thanks,
George
I am new to Asterisk so can I start by apologising if this question has been asked and
answered already.
I'm in the UK using BT for two incoming lines, one on Wildcard TDM400P and the other
on Wildcard X100P. I also have a SIP connection to voiptalk.org.
Incoming calls via SIP/broadband
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