[Asterisk-Users] Mediatrix 1204 SIP FXO 4-port gateway review

[Rich Adamson]

Product Review

Mediatrix 1204 4-Port SIP FXO Gateway
Firmware: v2.4.10.69 - US Version
US Retail: ~$750, Street Price: ~$450.

The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks
and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn
lines in either Loop Start or Ground Start mode, handles incoming CallerID,
and generates either Dial Tone (back towards the incoming pstn caller) or
redirects the call to a specific pre-programmed sip proxy extension. The
1204 can only be programmed through an SNMP (Simple Network Management
Protocol) manager. A Windows-based SNMP manager is supplied with the unit;
but no Unix-based manager. (And, no telnet, no web.)

To use the 1204 with Asterisk, each of the four pstn lines "must" be redirected
to an Asterisk extension. In this eval case, port 1 was redirected to x3091,
port 2 to x3092, etc. The 1204 detects the incoming call, and about midway
through the second ring, sends a sip Invite with the CallerID (if available)
to the defined sip proxy server (Asterisk). (After Asterisk completes the call
to another sip phone and the pstn caller hangs up, the Asterisk sip phone
will continue to ring for at least two-to-four additional ringing cycles.)

The firmware version tested did not support the sip "register" function even
though parameters were provided to enter the IP address of a registrar. As
a result, no userid/passwords or other security features are available. 
Sip.conf security entries are limited to "host=<ip address>" and context=
<MyPstnContext>". All incoming port 1 calls are directed to an extension.conf
construct similar to exten => 3091,1,Goto(my-ivr) contained within the
<MyPstnContext> section.

Again since the 1204 does not support the sip "register" function, outgoing
pstn calls from Asterisk can only be sent to the 1204 with commands similar to:
 exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:1}@mediatrix-1204)
where the 1204 selects one of the non-busy four pstn ports at random to initiate 
the pstn call. The reviewer could not find a way to direct specific Asterisk
calls to specific 1204 ports, and believes Mediatrix needs to fully implement
the "register" command on a per-port basis to aid in this requirement.

Echo cancellation and transmission levels were excellent on all inbound and 
outbound calls. Ringback tone and Music on Hold (MOH) were extremely choppy
until the Port1DspVoiceActivityDetection = 0 parameter disabled this function.
NAT is supported according to the documentation, however I did not test this
to see if it actually worked.  The standard rtp redirection (canreinvite=yes)
appeared to function properly.

As mentioned, the only way to configure the 1204 is via an SNMP Manager. There
is no way to change/secure the SNMP-v1 community string, therefore this box 
should never be exposed to the Internet. The *.pdf documentation files are
very verbose and good (Admin = 196 pages); however there are no references to 
Asterisk, leaving the reader to guess at how some functions actually 
inter-operate, etc.

Opinion:
It would appear the 1204 is oriented to inter-operate with another 1204 across
the Internet, creating essentially a virtual pstn line extension to some 
distant point. The box is available with either H-323 or SIP images, but not
both. One can only assume the incomplete SIP implementation is the result of
retrofitting the 323-based box into the SIP world. Since much of the *.pdf
documentation and files were dated March/April 2003, it does not appear 
that SIP advancement is high on Mediatrix's list of priorities. Support for
the unit is limited by Mediatrix to "resellers only", therefore obtaining
any relevant support data in a timely manner is 100% dependent on how well
your reseller will support you.

Trouble shooting is limited to the SNMP manager only. The manager can be used
to view configuration data, however needed dynamic operational statistics are
limited to mib2 definitions only.  For example, when trying to determine the
souce of choppy MOH sound, I wanted to check the Ethernet port speed. There
was no mib variable defined for this purpose.

Overall, the 1204 functioned very well for what has been implemented, however 
a more complete sip implementation, better technical support, and limited 
trouble shooting access will delay my decision to purchase this unit.