ATA186 configuration.

The ATA configuration is well described by Cisco you will find pointers on the following page
Asterisk phone cisco ATA18x

For a 5mn quick start do the following :

Connect your ATA on the Ethernet, it will get an IP address via the DHCP. Look in your message log to see which IP adress has been allocated
Connect with a web browser on your adaptor http://the-ata-ip-address/dev to get access to the configuration.
Set UID0: to your-login-line1 ; This will be reuse later in asterisk configuration as the end point ID (Channel in sip.conf)
Set PWD0: to the associated passwd
repeat for UID1 and PWD1 for line 2    ; Note the UID1 MUST be different from UID0

Set UseLoginID: to 0 ; If you want a loginID different of your end point ID set to 1 and fill in LoginID0: and LoginID1:
Set GkOrProxy: to the IP address of your Asterisk server (or SIP proxy/gateway)
Save your configuration.

You can see my ATA186 configuration in the joined file sip.ata186.conf.pdf

/etc/asterisk/sip.conf

[general]
 context=internal              ; Default context for incoming calls
 allowguest=yes                  ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
 bindport=5060                  ; UDP Port to bind to (SIP standard port is 5060)
 bindaddr=0.0.0.0              ; IP address to bind to (a good idea to restrict that to what you really want)
 srvlookup=no                    ; Enable DNS SRV lookups on outbound calls
 pedantic=no                     ; Enable slow, pedantic checking for Pingtel; This section declares a Cisco ATA with 2 analogue lines attached.

; This section declare the Cisco ATA186 Lines 1 and 2

[ata1-line1]
 type=friend
 secret=ata1-line1
 qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal

[ata1-line2]
 type=friend
 secret=ata1-line2
 qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal

; This section declares 4 SIP phone runing on PC client (e.g kphone) Useful for test purpose

[phone1]
 type=friend
 secret=phone1
 ;qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal

[phone2]
 type=friend
 secret=phone2
 ;qualify=yes
 nat=no
 host=dynamic
 caninvite=no
 context=internal

/etc/asterisk/extension.conf

 
 [internal]                                ; As declared in /etc/mgcp.conf calls coming from the ATA will be presented with the context.
 ;
 include => emergency               ; ANY good numbering plan SHOULD have an entry for emergency services in any context.
 include => your-extra-context      ; Add here any context(s) that you need/want to

 ; Integrate a Cisco ATA186 configured in SIP mode.
 ;Integrate a SIP Cisco ATA 186 (see related conf.pdf)
 exten => 111,1,Dial(SIP/ata1-line1,45,o)                  
 exten => 112,1,Dial(SIP/ata1-line2,45,o)

 ;for info integrate a MGCP Cisco ATA 186 (see related conf pdf)
 exten => 121,1,Dial(MGCP/aaln/1@000e83e530ae,45,o)          ; dial end point aaln/1 in MGCP Channel (see mgcp.conf) 000e83e530ae, rings to to 45s , pass the CallerID as received in the context.
 exten => 122,1,Dial(MGCP/aaln/2@000e83e530ae,45,o)          ; dial end point aaln/2


You are done.

See Also

• More info on Sip configuration Asterisk config sip.conf
• HowTo Cisco ATA 186 Cisco ATA 186 MGCP and Asterisk (HowTo)
• Asterisk MGCP configuration Asterisk config mgcp.conf
• Asterisk phone cisco ATA18x