ATA186 configuration.

The ATA configuration is well described by Cisco and 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 mgcp.ata186.conf.pdf

attachment id=656

/etc/sip.conf

; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;

Useful CLI commands to check peers/users

; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
; reload chan_sip.so Reload configuration file
; Active SIP peers will not be reconfigured
;

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'
                               ; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=141.169.121.142 ; IP address to bind to (0.0.0.0 binds to all)
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/extention.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

• HowTo Cisco ATA 186 Cisco ATA 186 MGCP and Asterisk (HowTo)
• Asterisk MGCP configuration Asterisk config mgcp.conf