[Asterisk-Users] First configuration
Germán Micale
gmicale at astrasoft.es
Wed Jan 19 03:51:41 MST 2005
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:password at sip.adiptel.com/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 6a4fea8451acdee4523ebd174a7fd379 at 192.168.1.2
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:tito at astrasoft.es>' failed for
'192.168.1.5'
I don't know how to solve it.
Thanks for your help
-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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/ _______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list