[Asterisk-Users] Problems configuring Asterisk with SIP

[Felix]

Hi everybody

Could someone give a tip  on how can I configure asterisk to use 2 ATA's 
186 to communicate each other using SIP with asterisk. I know this most 
be a very simple task, however this is the very first aproach I have to 
asterisk. I set the following config but I don't get dial-tone when I 
off-hook the phone from any of the two ATAs. Can some one tell what I'm 
missing in the configuration??

sip.conf  file

[general]
port = 5060                     ; Port to bind to
bindaddr = 192.168.0.254        ; Address to bind to
context = default               ; Default for incoming calls
tos=lowdelay
;tos=184
maxexpirey=3600         ; Max length of incoming registration we allow
defaultexpirey=120              ; Default length of incoming/outoing 
registration
;notifymimetype=text/plain      ; Allow overriding of mime type in NOTIFY
;
;register => 9873 at 192.168.0.2   ; Register with a SIP provider
;register => 9874 at 192.168.0.2
;register => 2345 at 148.243.196.14/1234 ; Register 2345 at sip provider as 
1234 here.
;allow=g729
;
;[cisco]
type=friend
username=9873
secret=pwd
;nat=yes                        ; This phone may be natted
host=dynamic
canreinvite=no                  ; Cisco poops on reinvite sometimes
qualify=200                     ; Qualify peer is no more than 200ms away
defaultip=192.168.0.5
mailbox=9873

;[cisco2]
type=friend
username=9874
secret=pwd
nat=yes                        ; This phone may be natted
host=dynamic
canreinvite=no                  ; Cisco poops on reinvite sometimes
qualify=200                     ; Qualify peer is no more than 200ms away
defaultip=192.168.0.10
mailbox=9874


extensions.conf
I added this at the end of the extension.conf file:

exten => 9873,1,Dial(SIP/cisco,30,tr)
exten => 9873,2,Playback(new/nbdy-avail-to-take-call)
exten => 9873,3,Voicemail(u9873)
exten => 9873,4,Hangup
exten => 9873,102,Voicemail(b9873)
exten => 9873,103,Hangup


exten => 9874,1,Dial(SIP/cisco2,30,tr)
exten => 9874,2,Playback(new/nbdy-avail-to-take-call)
exten => 9874,3,Voicemail(u9874)
exten => 9874,4,Hangup
exten => 9874,102,Voicemail(b9874)
exten => 9874,103,Hangup

And that's all I did. However I'm not sure If I have to configure 
something else?? I also have a SIP proxy server(Not asterisk) and I 
pretend to send out calls through this proxy server, but thisonce the 2 
ATAs can call to each other behid asterisk.

Can someone give a hint on this??? Any tip would be appreciated.

I'm actually using Redhat 9. The ATAs are using the 2.16 firmware. The 
ATA's are pointing to the asterisk bind address 192.168.0.254 in the 
sip.conf

Thanks in advance!!

Kind Regards!!

This is what I have:

ATA 1, UID0=9873
192.168.0.5-------------
                                    |
                                    |----------Asterisk 
BOX---------------------SIP-Proxy Server
ATA 2, UIDO=9874  |             192.168.0.254                           
192.168.0.2
192.168.0.10 -----------



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