[Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual

Jamie Carl geek at j-code.net
Sun Sep 5 17:49:43 MST 2004


Thanks to everyone for their help and comments on this.  You've all been 
very helpful.  I've actually got outbound calls working on it fine right 
now without having to change the configuration on the Mediatrix box at 
all, as I don't have the Unit Manager Software at the moment.  Outbount 
seems to work well but without inbound it means I can't put it in place 
for general use.  I have my 'reseller' tracking down the software for me 
right now so hopefully he'll be able to find it for me. :)

Asterisk doesn't seem to have any issues working with the APA III-4FXO 
at all as yet. 

Thanks again guys.

J




Gonzalo Gasca Meza wrote:

>     Here is my configuration for MEdiatrix 1204, by default the 1204
>     strips one digit, so it is not necessary to use:
>
>     To dial OUTSIDE
>
>     EXTENSIONS.CONF
>
>     [locales]
>     ;ignorepat => 9
>
>     exten => _9XXXXXXXX,1,Dial(SIP/${EXTEN-1}@Mediatrix
>     <mailto:SIP/$%7BEXTEN-1%7D at Mediatrix>)
>     exten => _9XXXXXXXX,2,Congestion
>     exten => _9XXXXXXXX,102,Congestion
>
>     To receive calls
>
>     [from-pstn]
>     ;Incoming calls from Mediatrix 1204, the 1204, sends an invite to
>     1111 at 110.10.200.2 <mailto:1111 at 110.10.200.2>
>
>     exten => 1111,1,Dial(SIP/100,20)
>     exten => 1111,2,Voicemail(u100)
>     exten => 1111,102,Voicemail(b100)
>     exten => 1111,103,Hangup
>
>     *******************************************************************************************************
>
>     SIP.CONF
>
>     ;Mediatrix Telecomm 1204
>     [Mediatrix]
>     type=peer
>     host=110.10.200.10
>     mask=255.255.255.255
>     context=from-sip
>     qualify=yes
>     canreinvite=yes
>     disallow=g729
>     nat = yes
>
>     In MEdiatrix 1204 use a program called Unit Manager Network a
>     Configure the first port as extension 1111 for port 1, in option
>     SIP. as user agent. also edit registar an dproxy SIP as the IP
>     address of Asterisk.
>
>     Works VERY GOOD with one line, although i have seen some scenarios
>     with more than 1 line which experince problems.
>
>
>
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