asterisk users - Mar 2004 - Calling one local SIP user from another (using X-Lite)

I'm having trouble (using a recent CVS Asterisk) calling one local SIP user 
from another (using X-Lite as the SIP phone).  My "sip.conf" file is
as
follows:

;;;;;;;;;;;;;;;;;;;;;;;;
[general]
port = 5060                     ; Port to bind to
bindaddr = <IP-address-of-Asterisk-machine>          ; Address to bind to
context = default               ; Default for incoming calls
disallow=all
allow=gsm
allow=ulaw
allow=alaw

[user1]
type=friend
username=user1
secret=<password-for-user1>
host=dynamic
context=default

[user2]
type=friend
username=user2
secret=<password-for-user2>
host=dynamic
context=default
;;;;;;;;;;;;;;;;;;;;;;;;

I have X-Lite running on two client computers.  On the first computer, 
X-Lite is registered (successfully) as "user1"; on the second, X-Lite
is
registered (successfully) as "user2".  From either computer, I am able
to
successfully call the 'demo' extension, and voice mail.  However, if I
try
to call "user2" from user1's X-Lite - or vice-versa - I get a
"404 Not
Found" error.

Is there anything obvious that I'm doing wrong?  (In particular, do I also 
need to add entries to "extensions.conf" for "user1" and
"user2"??)

	Ross.

Try to put this line to your user config in SIP.CONF

[USERID]
username=xxxx
secret=xxxx
dtmfmode=rfc2833
....

Ignace

Ross,

>From: Ross Finlayson <finlayson@live.com>
>Subject: [Asterisk-Users] Calling one local  SIP user from another (using 
>X-Lite)
>Date: Sun, 14 Mar 2004 18:21:20 -0800
<snip>
>voice mail.  However, if I try to call "user2" from user1's
X-Lite - or
>vice-versa - I get a "404 Not Found" error.
>Is there anything obvious that I'm doing wrong?  (In particular, do I
also
>need to add entries to "extensions.conf" for "user1" and
"user2"??)
>
>	Ross.
Try adding something similer to this in the default context of your 
extensions.conf:
exten => <your exten>,1,Dial(SIP/user1,20, tr)
exten => <your exten>,1,Dial(SIP/user2,20, tr)

Regards, Girish

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>>voice mail.  However, if I try to call "user2" from
user1's X-Lite - or
>>vice-versa - I get a "404 Not Found" error.
>
>>Is there anything obvious that I'm doing wrong?  (In particular, do
I
>>also need to add entries to "extensions.conf" for
"user1" and "user2"??)
>>
>>         Ross.
>
>Try adding something similer to this in the default context of your 
>extensions.conf:
>exten => <your exten>,1,Dial(SIP/user1,20, tr)
>exten => <your exten>,1,Dial(SIP/user2,20, tr)
Girish,

Thanks for the tip.   Yes, the following additions to
"extensions.conf" did
the trick:
         exten => user1,1,Dial(SIP/user1,20, tr)
         exten => user2,1,Dial(SIP/user2,20, tr)

(Why does the Asterisk configuration file syntax have to be so damned 
baroque?  Sigh...)

         Ross.

>>>Try adding something similer to this in the default context of your 
>>>extensions.conf:
>>>exten => <your exten>,1,Dial(SIP/user1,20, tr)
>>>exten => <your exten>,1,Dial(SIP/user2,20, tr)
>Always remember that device names - peers, users, friends - in any channel
are
>not the same as extensions. You call extensions (defined in extensions.conf)
>and the Dial application connects an extension to a device on a channel.
>
>If you're used to a SIP proxy, this is not very easy to understand. I
confuse
>myself regurlarly in this area.
>
>You have to remember that Asterisk is a multiprotocol (VoIP and PSTN)
>PBX where we have to define a protocol-neutral layer to connect everything
>together. This layer is the dial plan in the file extensions.conf.
>
>Believe me, you will appreciate it when you get the whole picture.
Olle,

Yes, I get it now.  (I'm a computer scientist, so I'm used to dealing
with
layers of abstraction :-)

(Nonetheless, it would be useful if there were a comment added to the 
default "sip.conf" file explaining that, for each SIP user (or friend
etc.)
defined in the "sip.conf" file, there needs to be a corresponding 
'extension' defined in "extensions.conf", with an example of
how to do this.)


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>