asterisk users - Oct 2009 - Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?

I'm trying to setup sipgate on 1.6.1. Following the instructions on the 
site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,

I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:

[sipgate]
type=friend
secret=  ;;SIP_PASSWORD
insecure=port,invite
defaultuser= ;; SIP-ID
fromuser=      ;;SIP-ID
context=sipgate_in
fromdomain=sipgate.com
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
canreinvite=no

which seems to take:
sip show user sipgate
asterisk*CLI>

   * Name       : sipgate
   Secret       : <Set>
   MD5Secret    : <Not set>
   Context      : sipgate_in
   Language     :
   AMA flags    : Unknown
   Transfer mode: open
   MaxCallBR    : 384 kbps
   CallingPres  : Presentation Allowed, Not Screened
   Call limit   : 0
   Callgroup    :
   Pickupgroup  :
   Callerid     : "" <>
   ACL          : No
   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Sess-Min-SE  : 90 secs
   Ign SDP ver  : No
   Codec Order  : (ulaw:20)
   Auto-Framing:  No

[sipgate-in]
exten => s,1,NoOp(Context: sipgate-in)
exten => s,n,NoOp(CALLERID(all))
exten => s,n,NoOp(${SIP_HEADER(To)})

But:

chan_sip.c:18667 handle_request_invite: Call from '7xxxxxxxx' to 
extension '7xxxx' rejected because extension not found.

7xxxxxx is the SIP-ID.

I've tried using _7! in sipgate-in, but no change.

Thanks for any help.

sean

sean darcy wrote:
> I'm trying to setup sipgate on 1.6.1. Following the instructions on the
> site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,
> 
> I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:
> 
> [sipgate]
> type=friend
> secret=  ;;SIP_PASSWORD
> insecure=port,invite
> defaultuser= ;; SIP-ID
> fromuser=      ;;SIP-ID
> context=sipgate_in
> fromdomain=sipgate.com
> host=sipgate.com
> outboundproxy=proxy.live.sipgate.com
> qualify=yes
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> nat=yes
> canreinvite=no
> 
> which seems to take:
> sip show user sipgate
> asterisk*CLI>
> 
>    * Name       : sipgate
>    Secret       : <Set>
>    MD5Secret    : <Not set>
>    Context      : sipgate_in
>    Language     :
>    AMA flags    : Unknown
>    Transfer mode: open
>    MaxCallBR    : 384 kbps
>    CallingPres  : Presentation Allowed, Not Screened
>    Call limit   : 0
>    Callgroup    :
>    Pickupgroup  :
>    Callerid     : "" <>
>    ACL          : No
>    Sess-Timers  : Accept
>    Sess-Refresh : uas
>    Sess-Expires : 1800 secs
>    Sess-Min-SE  : 90 secs
>    Ign SDP ver  : No
>    Codec Order  : (ulaw:20)
>    Auto-Framing:  No
> 
> [sipgate-in]
> exten => s,1,NoOp(Context: sipgate-in)
> exten => s,n,NoOp(CALLERID(all))
> exten => s,n,NoOp(${SIP_HEADER(To)})
> 
> But:
> 
> chan_sip.c:18667 handle_request_invite: Call from '7xxxxxxxx' to 
> extension '7xxxx' rejected because extension not found.
> 
> 7xxxxxx is the SIP-ID.
> 
> I've tried using _7! in sipgate-in, but no change.
> 
> Thanks for any help.
> 
> sean
Now I see it. sipgate_in vs. sipgate-in!

sean