[Asterisk-Users] IAX Softphone

2005-01-26 Thread Germán Micale
Hi,

Does someone know an ActiveX IAX softphone?
I need a free softphone to connect with Asterisk from a web page.

Regards 


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[Asterisk-Users] Mobile Callings

2005-01-24 Thread Germán Micale
Hi,

Does someone knows what kind of device I need to call from my pc to the
mobile network?
In Spain VoIP prices are very similar to call to a mobile than do it
from an other mobile. So, I want to plug some device to the PC and get
out the call throught it, but I dn't know what kind of device I need.
Thanks in advance


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RE: [Asterisk-Users] Mobile Callings

2005-01-24 Thread Germán Micale
Thank you Andrew,

The network is gsm



-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Andrew
Thompson
Enviado el: lunes, 24 de enero de 2005 17:24
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Mobile Callings


Germán Micale wrote:
 Hi,
 
 Does someone knows what kind of device I need to call from my pc to 
 the mobile network? In Spain VoIP prices are very similar to call to a

 mobile than do it from an other mobile. So, I want to plug some device

 to the PC and get out the call throught it, but I dn't know what kind 
 of device I need. Thanks in advance

look up: cellsocket

There are other similar devices, but the names now slip my mind. What 
type of network is your cell phone on? (cdma, gsm, tdma, etc)


-- 
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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[Asterisk-Users] Asterisk and SIP Communicator

2005-01-21 Thread Germán Micale
Hi,

I have a problem calling with SIP Communicator.
Connecting directly from SIP Communicator to the VOIP Provider, the
audio doesn't work.
Connecting throught Asterisk, the audio doesn't work.
Connecting throught Brekeke, all works correctly.

I need to use Asterisk and SIP Communicator. Does someone know what can
be happening?

Thanks in advance.


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[Asterisk-Users] softphone

2005-01-20 Thread Germán Micale
Does someone know a free SIP softphone which can be used from a web page
and with Asterisk?
Thanks in advance


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RE: [Asterisk-Users] First configuration

2005-01-19 Thread Germán Micale
I'm trying to do the following:
Several validated users of a web page makes their calls. The call arrive
to Asterisk and is redirected to sip.adiptel.com , where I have only one
user account. 
All the callers will arrive to Asterisk with their own user and password
(web validation), and Asterisk must change it to the parameters of my
only account at tha SIP Provider.

To do this, I compiles with 'make samples' and, after that:

sip.conf:

[general]
port = 5060
bindaddr = 0.0.0.0
context = sip

register = user:[EMAIL PROTECTED]/1000

[astrasoft.es]
type=peer
host=192.168.1.2
fromuser=us
secret=pwd
fromdomain=astrasoft.es

[sip.adiptel.com]
type=friend
secret=password
username=user
host=sip.adiptel.com

[phone1]
type=friend
host=dynamic
defaultip=192.168.1.2
username=tito
secret=tito
dtmfmode=rfc2833
mailbox=1000
context=sip
callerid=Tito 2124

extensions.conf:
--
[sip]
exten = 1,1,Dial(SIP/phone1,20,tr)
exten = 1000,1,Dial(SIP/Phone1,20,tr)



Error messages:
-
Jan 19 11:15:57 WARNING[2728]: chan_sip.c:685 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Request)

Trying to connect using Xlite:
Jan 19 11:17:41 NOTICE[2728]: chan_sip.c:7531 handle_request:
Registration from 'tito sip:[EMAIL PROTECTED]' failed for
'192.168.1.5'


I don't know how to solve it.
Thanks for your help


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Andrew
Thompson
Enviado el: martes, 18 de enero de 2005 19:56
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] First configuration


Germán Micale wrote:
 Hi everybody,
 
 I'had install asterisk, but I can't configure it to validate with my 
 VOIP provider.

Perhaps you could tell us who your provider is? Also, my telepathy is 
not working this week, so you'll actually need to send us the relevant 
sections of your config files.

 What I need is recieve our costumer's calls and redirect it using 
 allways a unique user and password.

Receive calls from where? Redirect them to where?

 Could some one help me?

It's very likely that someone here can, but right now, we know nothing 
about your specific configuration, or your problem. Error messages are 
helpful, too!

-- 
Andrew Thompson
http://aktzero.com/ ___
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[Asterisk-Users] First configuration

2005-01-18 Thread Germán Micale
Hi everybody,

I'had install asterisk, but I can't configure it to validate with my
VOIP provider.
What I need is recieve our costumer's calls and redirect it using
allways a unique user and password.
Could some one help me?
Thank you


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