www.xten.com
http://www.sjlabs.com/
- Original Message -
From: "James Moran" <[EMAIL PROTECTED]>
To: "Asterisk" <[EMAIL PROTECTED]>
Sent: Wednesday, April 14, 2004 8:52 PM
Subject: [Asterisk-Users] sip software
> Anyone have any suggestions on free sip phone software for windows??
> O
Paul,
http://www.voip-info.org/wiki-Asterisk
Wim
- Original Message -
From:
Paul
Tyreman
To: [EMAIL PROTECTED]
Sent: Saturday, April 10, 2004 6:47
PM
Subject: Re: [Asterisk-Users] Re:
Analogue telephone cards for the UK
Sorry to sound stupid, but whe
Maybe this??: http://www.grandstream.com/y-product.htm
- Original Message -
From: "Jason Miller" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 10:53 PM
Subject: [Asterisk-Users] Dial up adapter
I was wondering if anyone has used an adapter to dial up to a loca
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Wim Venneman
> > Sent: Tuesday, February 24, 2004 2:47 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] Unable to create channel of type 'Zap'
> >
create channel of type 'Zap'
> I had this after my last CVS update.
>
> A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Wim Venneman
> Sent: 24 F
IL PROTECTED]>
Sent: Tuesday, February 24, 2004 8:13 PM
Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
> Wim,
>
> What happens when you do a ztcfg -vv
>
> - Brent
>
> On Tue, 24 Feb 2004, Wim Venneman wrote:
>
> > Thanks Derek,
> &
Users] Unable to create channem of type 'Zap'
> Wim,
> Made one more change below in Zapata.conf
> It should be channel => 1
>
> -Original Message-
> From: Wim Venneman [mailto:[EMAIL PROTECTED]
> Sent: Monday, February 23, 2004 4:46 PM
> To: [EMAIL
own IRQ.
>
> - Brent
>
> -Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
> Sent: Monday, February 23, 2004 3:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
&
and after the modprobe.
>
> (You may also do a ztcfg -v to see whats configured)
>
> - Brent
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
> Sent: Monday, February 23, 2004 3:10 PM
> To: [EMAIL PROTECTED
Can anyone
help me, (after a two day search, also on the mailing list)
I have the
following situation:
Asterisk
works fine, until I added a FXO card. (Digium)
When I
tried to call to the pstn I have the following error
Executing
Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
NOTIVE[164
If this may be of any use:
I'm not an expert but I did the test with the FWD soft phone from X-ten and
iLBC & SPX don't work.
Asterisk wasn't between the connections. Just x-lite and fwd (who is an
Asterisk Server?)
The soft phone makes the connection but I can't hear any sound.
Wim
- Orig
Hi,
For our study we are searching for a (working) dial
plan that is used in a Cisco call manager or other PBX, to compare it with
Asterisk.
If anyone has a (simple) working example, can I
have a copy?
Thanks
Hi
all,Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the two sip
phones( hard or soft) passing through the asterisk server (on UDP
layer)
Isn't SIP
a protocol that (after that it has established the c
section of
sip.conf
that half fixes it for me calls between phones
work but talking to asterisk has some problems.
- Original Message -
From:
Wim
Venneman
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 2:29
PM
Subject
Hi,
I installed Asterisk an all works fine exept
for Grandstream.
When I call with a softphone (ex X-ten) to a
Grandstream (BudgetTone-100), I can make a conversation. = ok
When I call to a softphone with a Grandstream
I can pich up the call with the softphone but the Grandstream keeps ring
L PROTECTED]>
Sent: Thursday, October 30, 2003 9:02 PM
Subject: Re: [Asterisk-Users] Host unspecified ??
> Hi Wim,
>
> Citeren Wim Venneman <[EMAIL PROTECTED]>:
>
> > I changed the host to a fixed ip address (host1=192.168.10.12 and
> > host2=192.168.10.13) now the
Dear,
I changed the host to a fixed ip address (host1=192.168.10.12 and
host2=192.168.10.13) now the ip address shows up in the 'host' field = ok.
Try to call, no succes, nothing happens!
What's wrong?
Wim
- Original Message -
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: <[EMAIL PRO
Dear,
When I start asterisk -vvgrc and I
ask 'sip show peers', I don't get the ip adress in the 'Host"
field.
Name = phone1 and phone2
Host=unspecified
mask 255.255.255.255
port = 0
status = unmonitored
I can ping the two phone's and get a reply (also
from the laptop)
phone ip ad
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