[Asterisk-Users] Re: VoIP gateway (2 FXO, 2 FXS)

[Stewart Nelson]

> Does anyone know a good (and stable) voip gateway product with 4 ports
> (2 fxo and 2 fxs), with the following requirements:
> * being able to connect analog phones to the FXS ports, and communicate
>   over SIP with an REGISTRAR/PROXY server (SER in our case).
> * being able to connect the FXO port to local office PSTN network, and
>   dial to that office pstn number and getting an internal dialtone, or
>   forward ability to the SIP gateway.
>   So employees can call to the local pstn number, and enter an
>   international phone number which is routed over the SIP gateway (SER).
>
>
> The following are results with 2 products I tried, without any success.
>
> I used http://www.voip-info.org/wiki-VOIP+Gateways to order the following,
> * Ovislink VoIP-422
> * Welltech 3702A
>
> I've tested them, and came across the following problems,
>
> * Ovislink product
>
>   - Problem #1
>   adding sip accounts worked like a charm, they register perfectly with
>   our sip gateway (SIP Express Router).
>   But when we make a call from an analog phone (connected to a FXS
>   port), the SIP packets (INVITE, etc) do NOT include the authentication
>   details (SER sends 'Proxy Authentication Required'), the DIGEST
>   username is just blank and From is elite@ (no idea where that came from,
>   probably hardcoded).
>   I've tried linking a callerid/name with that FXS port, without a difference.
>
>   The same problem arises when we call the office pstn number (pstn
>   port connected to FXO port of ovislink box).
>   We get an internal dialtone (of the ovislink), and when the enter a
>   number, it also doesn't send the auth details in the SIP INVITE packet it
>   sends to SER.
>
>
>   - Problem #2
>   As a 'quickfix' I configured SER to NOT look at the auth details,
>   and just process the call anyway.
>   When the call is answered, and SER sends the  SIP/2.0 200 OK, the
>   Ovislink does NOT send the ACK (but I can see the incoming OK packet
>   in the ovislink console).
>
>   Quite buggy indeed.. or i'm misconfiguring the device, but i'm sure
>   I got everything right.
>
>   Anyone else with some experiences ?
>
>
> * Welltech product
>
>   Dialplan issues, I created the necessary routes to route everything
>   over IP.. but it still sends incoming PSTN calls (FXO port, LINE1),
>   to the analog phone connected on the FXS port (TEL1).
>
>   Calls made from the analog phone are routed over the LINE1/FXO port.
>
>   I specifically changed all the reference to FXO to IP, and STILL it's
>   sending the calls over the FXO port.
>
>
> Anyone got some luck with either of these products, or has another
> product that fullfill our needs ?
>
>
> Thanks in advance.
>
> --__--__--

Take a look at the Planet VIP-450
http://www.planet.com.tw/product/product_dm.php?product_id=195&menu_id=3 .
We use the VIP-400 (H.323 version), which has lots of flexibility
in the dial plan, IDs, etc.  You can download the complete manual from
the Planet site.

Pros:  Inexpensive, good voice quality, doesn't crash, excellent hardware
       reliability, good support for configuration problems.

Cons:  Many minor bugs and shortcomings (more subtle than your Ovislink
       and Welltech problems), no support at all for getting these fixed,
       unless a big customer of Planet happens to experience the same trouble!

Good luck,

Stewart