Patrick,
I think this is your answer:
https://www.nch.com.au/skypetosip/index.html
Regards,
Tigran
Patrick wrote:
Hi all,
In the past there was some discussion how to interface Skype to an
Asterisk box. I stumbled on this product:
http://www.nexotek.com/b300.html
There's a wmv video
The following page will give you a hint:
http://nerdvittles.com/index.php?p=73
Regards,
Tigran
Hello Sharon,
Saturday, July 1, 2006, 5:33:14 AM, you wrote:
anyone have information on how the call back features work with
asterisk? I means the dial plan or what so ever. thanks
--
Best
Mike,
If you feel afraid of the next power outage, why not install a more
powerfull UPS with a longer run time? Or, as it is in my case, a friend
of mine substituted the factory default battery in the UPS with a car
battery, that holds the Server for 4-5 hours. Add another battery and it
will
Hello,
VoiSmart GSM cards work with Asterisk. Though I have an issue with DTMF
detection. See the following pages for details.
http://open.voismart.it/index.php/VGSM
https://mailman.uli.it/pipermail/visdn-hackers/2006-June/thread.html
(Search for DTMF)
Regards,
Tigran
Woodoo People .pGa!
What exactly doesn't work?
Could you just paste the Asterisk debug output so that we could figure
out which part has the problem.
Maybe the forwarded-iax prompt does not exist, and it simply cannot
play it... :)
Regards,
Hohenzolern
Matthias Fechner wrote:
Hi,
I have configured asterisk
Hello,
Could anyone help me to figure out the following questions, please:
1. Whenever there is an incoming DTMF signal on the Zap channel, where
does the processing actually take place: In Asterisk?; or in Zaptel Drivers?
2. I'm having a problem of double (or sometimes tripple) detection of a
Hello Asterisk Community,
I'm using Voismart's GSM PCI cards to connect Asterisk to GSM cellular
network.
The problem I face is DTMF detection; that is, whenever I call to one of
the channels (SIMs) on GSM card through my Mobile phone, and dial DTMF
digits while in the call, the Asterisk
Try Ekiga,
It works as and is stabil as well.
Wednesday, June 21, 2006, 1:53:33 PM, you wrote:
I had problems with sjphone ... same version as yours.
Finally, i managed to solve it by:
- in sjphone, media channels settings: untick Use remote codec
preferences and Open audio streams after
It should do the job!
In my setup, I call from an IAX phone to an h323 Gateway, and all is
fine. The opposite direction also works fine.
Though this is an IAX setup, SIP should perform likewise.
REgards,
Hohenzolern
Gary Richardson wrote:
Nope, asterisk does the bridging. Asterisk can talk
Tigran Kocharyan osszedobalt bytejaira:"
1. Customer Calls the outgoing number which is a PSTN line connected to
my Zap channel
2. Asterisk captures the Caller ID and calls back the customer.
3. As soon as the customer picks up the phone, asterisk plays a promt to
enter the Destin
Thanks a lot for responding.
I did what you recomended, and it works now. At least I can make simple
calls out. Did not try the incoming part though.
Now it is still unclear :
- how to make the Dial application choose the first available channel?
- or how to get CID out of the interface? Does
Dear Forum Members,
I just purchased two VoiSmart GSM cards. Tried to install one of them on
my Fedora Core 5 system, The compilation was not smooth, as expected,
but after a small fix, it went through.
Then I put two SIM cards in the card's slots.
Then I loaded the modules.
Then I started the
Dear Asterisk Comunity,
I'm thinking about developing a callback application based on the
following scenario:
1. Customer Calls the outgoing number which is a PSTN line connected to
my Zap channel
2. Asterisk captures the Caller ID and calls back the customer.
3. As soon as the customer
for DTMFs? or should I park the call or should it be joined to a
conference? How to Bridge the calls?
Thanks for your reference anyway.
Jean-Michel Hiver wrote:
Tigran Kocharyan a écrit :
Dear Asterisk Comunity,
I'm thinking about developing a callback application based on the
following
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