Hello,
On Di, 2006-01-03 at 16:31 +0100, Giovanni Miano wrote:
> Use meetme app
Unfortunately meetme is no solution for me. If nobody can help me, is
there at least anybody who has the same problem?
As far as I can see there are lots of people using the HFC PCI card, is
nobody using Three-Way-Ca
Use meetme appCheers, Giovanni Miano2006/1/3, Henry Margies <[EMAIL PROTECTED]>:
Hello,I'm using a TDM 400 together with two HFC PCI Cards in my Box. Three WayCalling with a SIP or analog Phone is working perfectly.But if I try to do Three Way Calling with my ISDN Phone I get an error
message: "Fac
Hello,
I'm using a TDM 400 together with two HFC PCI Cards in my Box. Three Way
Calling with a SIP or analog Phone is working perfectly.
But if I try to do Three Way Calling with my ISDN Phone I get an error
message: "Facility Name requested on channel 0/2 not in use on span 1"
I use bristuff wi
hi
can i make sip three way call on asterisk
i meen one person call one time to two another
and when they answer this 3 person speak with each other
as in confereces
i cant use
meetme becouse i need send dtmf
--
Oleh Mukha
IClub
380322722738
www.ic.lviv.ua
___
Hi guys,
Does anyone know of a way where I can bring a third person in on my
conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM
and am speaking to someone now I want to quickly bring another SIP or IAX
extension into this call so the three of us can speak to each other.
I
How do I setup three way calling with Asterisk and a Cisco 12SP+
telephone? I would like to be able to three way Voipjet numbers as
well as IAX calls.
Thanks
BlakeOPS
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com
> There is a crack available:> http://www.regnow.com/softsell/nph-softsell.cgi?item=9054-12
You're suggesting eyeBeam ?
Do you Yahoo!?
Yahoo! Small Business - Try our new resources site! ___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Vyom A wrote:
In X-Lite, the "Conference" button is disabled, but that can probably
be done in X-Pro (from the XPRO_users_guide.pdf)
There is a crack available:
http://www.regnow.com/softsell/nph-softsell.cgi?item=9054-12
___
Asterisk-Users mailing list
In X-Lite, the "Conference" button is disabled, but that can probablybe done in X-Pro (from the XPRO_users_guide.pdf)
Do you Yahoo!?
Yahoo! Mail - You care about security. So do we.___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http:/
Try SJPhone. (www.sjlabs.com)
--On Tuesday, March 15, 2005 3:00 PM + Chris Blunt
<[EMAIL PROTECTED]> wrote:
Hi All,
Does any one know of a way to make a three way call from Asterisk using
X-Lite.
I need the ability to be able to call someone on the PSTN using my IAX
provider then bring ano
Hi All,
Does any one know of a way to make a three way call
from Asterisk using X-Lite.
I need the ability to be able to call someone on the
PSTN using my IAX provider then bring another person from a local extension
into the call if needs be?
I believe most three way calling
Hello All,
I have a customer site that is using * for ACD. In comming calls are eventually
routed to a support rep via a queue. For new accounts the agent needs to be able
to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial
the number of an authentication center and th
Title: RE: [Asterisk-Users] Dial Plan Format Strings
I
currently have about 40 users up on Asterisk and it is working great. One issue
I have though is the inability to conference calls/3-way calling on my SNOM
200 phones. Whenever I press the CNF/TRAN button on the phone, it just
drops the
Hello,
It's posible to implement three-way-calling with Asterisk. I would like to invite
other participants to an established current call. Should i use a dynamicly created
meetme room?
Thank you very much.
Gustavo García
___
Asterisk-Users mailin
I'm having a problem getting 3 way calling to work correctly using two
outside lines and one extension. The two outside lines are connected
to the X100P's and a standard model 2500 phone is connected to the
TDM10.
When I dial the first outside destination 9xxx, the call completes
correctly. W
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nome de Martin Pycko
Enviada em: quinta-feira, 28 de agosto de 2003 13:44
Para: [EMAIL PROTECTED]
Assunto: Re: [Asterisk-Users] Three way calling on outgoing FXO line
Press flash on your phone (asterisk will intercept
Press flash on your phone (asterisk will intercept that) and then when you
have a dialtone press *0 then asterisk will send the flash to PSTN line.
regards
Martin
On Thu, 28 Aug 2003, Carlton J. O'Riley wrote:
> I was wondering if anyone is able to use the three way calling features from
> their
I was wondering if anyone is able to use the three way calling features from
their telco on the incoming FXO lines to transfer a caller back out to say a
cell phone. I am currently moving from a Talkswitch to the Asterisk PBX and
one nice feature they have is after 4 rings or so I can have the cal
Thomas Dingermann wrote:
Ok, if this is not working with sip or h.323, maybe it does with mgcp ?
I tried to get ATA and Asterisk working with MGCP, but nothing worked!
Any Howtos available about MGCP/ATA186/Asterisk?
I just try two ATA with asterisk with that configuration files :
;
; MGCP Configu
Pavel Zheltouhov wrote:
Ok, if this is not working with sip or h.323, maybe it does with mgcp ?
I tried to get ATA and Asterisk working with MGCP, but nothing worked!
Any Howtos available about MGCP/ATA186/Asterisk?
Thomas
___
Asterisk-Users mailing lis
Thomas Dingermann wrote:
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk as pbx. I need feature called as 'three way calling' or
'transfer with consultation'. Registering,calling and 'blind transfer'
work fine.
Same here - and if you hang up, the call is not transferred...
Pavel Zheltouhov wrote:
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk as pbx. I need feature called as 'three way calling' or
'transfer with consultation'. Registering,calling and 'blind transfer'
work fine.
Same here
Is this feature provided by sip clients or by asteris
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk as pbx. I need feature called as 'three way calling' or
'transfer with consultation'. Registering,calling and 'blind transfer'
work fine.
Is this feature provided by sip clients or by asterisk itself ?
What I have to configur
23 matches
Mail list logo
