[Asterisk-Users] Mediatrix 1204 sip g/w now working

[Rich Adamson]

For those that might have the Mediatrix 1204 4-port FXO sip gateway or
for those that might have an interest, finally got it to work the way
one would expect when interconnecting to analog pstn lines.

Configuring the box for incoming calls was rather easy and worked 
shortly after installing the box.

Configuring it for outgoing pstn calls has been at least a two week effort
interacting with the reseller multiple times. The issues:

Port Selection:
---------------
The 1204 does not provide any documented method to "select" which of the
four ports will be used for outgoing calls. The manufacturer assumes all
four ports are the equivalent of a trunk group.
Fix: 
In extensions.conf, add something like:
 exten => _6X.,1,SETCIDNUM(1111)                                                 
 exten => _6X.,2,Dial(SIP/${EXTEN:1}@201.111.193.101)
 exten => _6X.,3,Congestion
and in the 1204:
 set gatewayPort1NetToPstnSourceFilter = 1111
Since the callerid that is set in asterisk never gets forward out the pstn
line, the above mechanism works fine for selecting port 1. (Use 2222, 3333,
4444 for the remaining ports.)

Outbound calls dropping first digit:
------------------------------------
The 1204 automatically drops the "1" when calling any long distance call
such as 1-800-555-1212.
Fix:
on the 1204, set countryCountryCode = 2
This is an undocumented item, but essentially stops the 1204 from stripping
leading digits.

Summary:
--------
The limited testing conducted thus far indicates the 1204 is working very
well. There is no noticeable echo at any time. Seems to work very well with
canreinvite=yes although I've not tried it with a remote nat phone.

One of the nice things about the box is you can locate it at your demarc
and not have to provide 2-wire pstn connections to the asterisk system.

Rich