Cisco ATA 186 SIP and Asterisk - HowTo

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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

Image




/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


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

Image




/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


Created by: dominig, Last modification: Sun 20 of May, 2012 (19:54 UTC) by admin
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