-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: 19 January 2005 14:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Router Recommendations Please
On Tue, 18 Jan 2005 20:01:51
Hi,
I am looking for some affordable IP Phones. Any experiences with the
SipToneII by ipDialog?
What about soft phones? Any recommendations there (for Windoze and Linux)?
Thanks,
Yiannis
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Hi,
before you start throwing stones to me let me tell you that I am a bit new
to Linux. I downloaded Asterisk from the cvs server on Wednesday 15 July
2004, as described in Andy Powell's Getting Started with Asterisk
(http://www.automated.it/guidetoasterisk.htm). Thanks Andy! I read
The extension of an incoming call through the X100P is s. So,
[incoming]
exten = s,1,Answer
exten = s,2,Dial(SIP/200)
exten = s,3,Hangup
[outgoing]
exten = _9.,1,Dial(ZAP/g1/${EXTEN,1})
You need to dial 9 from your SIP phone to get an outside line and then the
number you wish to dial.
g1
Hi,
I know that this issue has been discused guite a lot, but I haven't managed
to get a definite answer. Is those two values supposed to be floats (e.g.
3.5) or integers with the percent symbol (e.g. 20%)?
Thanks,
Yiannis.
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Asterisk-Users
Can I contact you off-list?
Please provide email address.
Yiannis Costopoulos.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of SeshKanuri
Sent: 22 September 2004 22:41
To: Asterisk Users Mailing List - Non-Commercial Discussion; 'Shaun
Ewing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andy Powell
Sent: 25 September 2004 16:27
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
On 25/09/2004 at 14:31 Arik Funke wrote:
Hello together,
I
Hi,
I have a couple of IpDialog SipToneII phones and although I understood that
they had a choice of 5 ringtones, it turns out that it is Distinctive
Ringing. I contacted IpDialog support and sent me this.
---snip--
The phones you have support 5 different ringtones and 4 call
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Sent: 15 October 2004 02:41
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Looking for recommendations for a low-cost FXO
toIP gateway.
At first I thought the X100P was what I was looking for,
Well,
assuming that some of these CODECS do error correction and drop any
information that hasn't come through instead of doing error detection and
request to re-transmit the lost information, is somewhat expected. Are there
any Fax over IP protocols?
Yiannis.
-Original
Yes, it's true! Connect the card to a phone line and the Red Alert
disappears. I don't think it draws power from the phone line, but it gives a
red alert if the phone line is not there. I have experienced it myself!
Yiannis.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi All,
I am investigating the deployment of VoIP/* in Eastern European areas
where
there is no PSTN infrastructure. As you can understand DSL/Cable connections
are a dream. The only option is satellite.
Does anyone know of any satellite providers that have low enough/acceptable
delays
No it wouldn't. It doesn't run on his laptop.
I would suggest two FWD accounts with two SIP softphones.
Unless you really want to go the Asterisk way and use IAX softphones over
NAT that works slightly better.
Yiannis.
-Original Message-
Skype would do you the best.
On 6/2/05,
Hi,
the best thing to do is get a Sipura 3000 that has 1 FXO and 1 FXS port.
You won't need to bother with IRQs and echo problems that at least here
in UK we have with FXO cards.
Yiannis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Logan
Sent:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Russ Price
Sent: Wednesday, November 16, 2005 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk hobby box
Funny you would say that - I have a box running with a pair
Personally, I believe it's a good thing. It gives more choice.
Look at other products: IPCop (Linux based firewall) is a fork derived from
Smoothwall. They made such a nice job that Smoothwall were playing catch-up
with IPCop for quite some time. I don't know the current situation.
GPL allows
Hi,
I have an Asterisk box with two P100X connected on two UK (BT) lines. When
I make a call, sometimes, I get too much echo and the system gives me a Red
Alert. After that the line is busy. If I disconnect the line from the X100P
and I call the line, I get ringing tone. When I reconnect
Have there been noticed any differences in echo from distro to distro on the
very same hardware?
I mean install a distro compile and run *, then replace it with another
distro on the same box and cards.
That could be intersting.
Thanks,
Yiannis.
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I am the IT guy at a small startup based in UK. At the moment we have 3
analogue (PSTN) lines and we will be adding another 2 or 3 soon. Later on we
should be changing to ISDN30.
One of the partners mentioned getting an analogue PBX now, and when we move
to ISDN, then get a
PROTECTED] Behalf Of Chris Bond
Sent: 27 June 2004 16:11
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Multiple X100P in Asterisk box?
You can use the new digium TDM cards with 4x FXO modules you'd only need 2
then.
-Original Message-
From: Yiannis Costopoulos, Web2Net Solutions
Hi,
for SIP account you can use this: http://www.freeworldialup.com/
for a UK number try this: http://www.calluk.comthe numbers are
free.
Regards,
Yiannis
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Johannes van
HulstSent: 14 July
Hi,
I am thinking of using * with IP phones instead of a hardware PBX.
The situation is like this. I have 3 different companies having an analogue
line for each of them. I want to make sure that when a call comes in, we
have an indication on the IP phone which line the call comes from.
Hi,
I think that the problem is with the codecs. Search the
Wiki and the list archives (through Google) to find what settings in sip.conf
you need for Budgetone and Sipura. The settings you need are *allow* and/or
*disallow*.
Yiannis.
-Original Message-From:
[EMAIL
Hi,
is there a definite answer if asterisk can pass calls between SIP and h.323
protocols?
Thanks,
Yiannis.
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