Hi again. I've used the zapata.conf and the extensions.conf files at the
adrees you sent (showed below). I didn;t use the sip.conf file because
it seems to be appliable only to sip devices... and here comes my first
doubt:
At the "Getting Started" link included at www.asterisk.org
(http://www.automated.it/guidetoasterisk.htm) says that if you want to
use a FWD you must include some information in sip.conf:
<>
------------------------------------------
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sip ;default Default for incoming calls

register =>fwdnumber:[EMAIL PROTECTED]/1000

<><>[fwd.pulver.com]
type=friend
secret=mypassword
username=my fwd number
host=fwd.pulver.com

---------------------------------

I have this sip.conf file (using the real FWD number-485445 and pasword,
of course), and when I start asterisk:

linux# ] asterisk -vvvc

I get a lot of messages (I'read this is normal) but then, when the CLI
prompt appears, I get warning messages:

NOTICE[229390]: chan_sip.c:3922 sip_reg_timeout: Registration for
[EMAIL PROTECTED] has timed out, trying again
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 590 __sip_xmit: sip_xmit of 0x81445dc (len
366) to 192.246.69.223 returned -1: Bad file descriptor
WARNING[229390]: chan_sip.c: 673 retrans_pkt: maximum retries exceeded
on call [EMAIL PROTECTED] for seqno 104 (Critical Request)

What is wrong?

When configuring XLite I saw many parameters I dont see anywhere here,
like stun server, sip outbound proxy....

I hope you can help me,

RODOLFO

Eric Wieling wrote:
See sample configs in www.fnords.org/~eric/asterisk

On Sun, 2004-09-12 at 15:42, Rodolfo Grave wrote:


OK. Thanks a lot. I'll change my first set up then...:)

I want to be able to make a PSTN call to the line connected to asterisk, and that asterisk answer that call and ask for a sip number to dial.... is this also simple? can you give me a simple setup for this?

Thanks a lot for your help.

Eric Wieling wrote:



The Phone port wired to the Line port so you can still use a phone
plugged into the card when the server is down or powered off.  Let me
repeat this: You cannot plug a phone into the X100P and expect it to
work with Asterisk.

On Sun, 2004-09-12 at 13:09, Rodolfo Grave wrote:




But I see there is a plug for LINE and another for PHONE..... and also I'm almost sure I read you could connect a normal phone to the x100p...

Eric Wieling wrote:





The X100P is for connecting to phone LINES (telco lines) not for
connecting to phones.

FXO = expects to RECEIVE dialtone and ring voltage
FXS = expects to PROVIDE dialtone and ring voltage.

On Sun, 2004-09-12 at 12:59, Rodolfo Grave wrote:






Hi.

I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly:

I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal PSTN calls...

I suppose that's possible (I find it very simple for the things you can do with asterisk), but I'm a bit lost in the configuration options and I want to use this simple settings to go in there and understand all that better.... I do have read the documentation, but I'm still a bit lost.

Thanks to everybody, you're a helpful community.

RODOLFO


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