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Cisco ATA 186 MGCP and Asterisk - HowTo
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
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; 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=18.104.22.168 ; 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
[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.
- Asterisk MGCP configuration Asterisk config mgcp.conf
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