asterisk users - Feb 2004 - Mediatrix 1204 sip g/w now working

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

I've had one of these things working for ages, although I never set it up to
select which port to use on outgoing lines. I overcame the first-digit 
stripping by telling the 1204 to prefix outgoing calls not in my area code. I 
seem to remember it stripping leading zeroes (as in 011 for international 
calls). I should try your undocumented feature.

The other thing I've had troubles with is provisioning it via DHCP. All of
the
DHCP key-value pairs are recognised except the one for outgoing proxy. It's 
very annoying, and seems to be a firmware bug. So I've configured the
gateway
to use a static IP.

Anyway, once set up, it seems to work okay, though it of course suffers from 
the same hangup detection problems that afflict all users of loop start.

Thanks for the config tip to manually select outgoing ports; that could be 
handy.

Christian

On Wednesday 11 February 2004 14:51, Rich Adamson wrote:
> 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
>
>
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