asterisk users - Jun 2010 - originating a sip call from the CLI

Hello again!
   I just got a SIP account and it seems - from a config on the net -, that 
I've configured it correctly. But I get no call to the outside. Registration
was OK.
   I tried:
channel originate sip/1/echo at iptel.org Application ...
   I see the channel active for a while, but no call gets established.
   In my config I have defined the section [iptel] for the outgoing call and I 
tried [iptel_in] for incoming calls, but commented it out again. My 
extensions.conf context is called sip-in.
   I didn't find an originate syntax on the web. I found a dial syntax, but
I'm
not sure it is the right thing.
   Can anyone please help.
   Kindest regards
            Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: =======http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: ======http://www.juliencoder.de

Hello again!
   So I tried again, experimented a bit more and got this:
channel originate sip/echo at iptel.org
[Jun  5 12:14:29] WARNING[8537]: chan_sip.c:17882 handle_response_invite: 
Re-invite to non-existing call leg on other UA. SIP dialog 
'3b39b40240b6126a61c7ad16108bee74 at 91.58.24.59'. Giving up.
   Below you can find a condensed version of my sip.conf.
*** /etc/asterisk/sip.conf ***
[general]
context=sip-in                 ; Default context for incoming calls
allowguest=yes                  ; Allow or reject guest calls (default is yes)
match_auth_username=yes        ; if available, match user entry using the
allowoverlap=yes                 ; Disable overlap dialing support. (Default is
yes)
allowtransfer=yes               ; Disable all transfers (unless enabled in peers
or users)
udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to
(0.0.0.0 binds to all)
tcpenable=no                    ; Enable server for incoming TCP connections
(default is no)
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0
binds to all interfaces)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
disallow=all                   ; First disallow all codecs
allow=ulaw                     ; Allow codecs in order of preference
allow=ilbc                     ; see doc/rtp-packetization for framing options
mohinterpret=default
mohsuggest=default
language=en                    ; Default language setting for all users/peers
relaxdtmf=yes                  ; Relax dtmf handling
useragent=J's Asterisk         ; Allows you to change the user agent string
sdpsession=J's Asterisk        ; Allows you to change the SDP session name
string, (s=)
sdpowner=juliencoder                  ; Allows you to change the username field
in the SDP owner string, (o=)
videosupport=no               ; Turn on support for SIP video. You need to turn
this
alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be
rejected,
dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering
shrinkcallerid=yes     ; on by default
rtptimeout=60                  ; Terminate call if 60 seconds of no RTP or RTCP
activity
rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP
activity
rtpkeepalive=50            ; Send keepalives in the RTP stream to keep NAT open
hash_users=32
hash_peers=32
hash_dialogs=16
recordhistory=yes              ; Record SIP history by default 
allowsubscribe=no              ; Disable support for subscriptions. (Default is
yes)
callcounter = yes              ; Enable call counters on devices. This can be
set per
registertimeout=20             ; retry registration calls every 20 seconds
(default)
registerattempts=10            ; Number of registration attempts before we give
up
localnet=192.168.220.1/255.255.0.0
localnet=192.168.220.105/255.255.0.0
localnet=192.168.220.106/255.255.0.0
externip=my_networks_external_ip_adress ; JPC IP goes here
directmedia=yes                ; Asterisk by default tries to redirect the
rtcachefriends=yes             ; Cache realtime friends by adding them to the
internal list
rtsavesysname=yes              ; Save systemname in realtime database at
registration
rtautoclear=yes                ; Auto-Expire friends created on the fly on the
same schedule
domain=iptel.org,sip-in
allowexternaldomains=yes
[authentication]
secret=password_for_iptel.org
remotesecret=password_for_+iptel.org_again
transport=upd,tcp
nat=yes
language=en
[iptel]
type=friend
host=iptel.org
username=juliencoder at iptel.org
fromuser=juliencoder at iptel.org              ; how your provider knows you
fromdomain=iptel.org
remotesecret=password_for_iptel.org ; The password we use to authenticate to
them
secret=password_for_iptel.org_again                ; The password they use to
contact us
callbackextension=S            ; Register with this server and require calls
coming back to this extension
transport=udp                ; This sets the transport type to udp for outgoing,
and will
busylevel=2
port=5060
; only templates and examples after this line
*** END of /etc/asterisk/sip.conf ***
   so this is it? Could you give me some hints, tips to get me going?
   Kindly yours
           Julien
--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: =======http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: ======http://www.juliencoder.de