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. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
