Bill H

Jordan, I am happy to hear that it is working now. You can check your Firmeware Version on the Status page of the GWX-410x The reason your unit would work hit and miss was due to time it takes the Dial Tone from the PSTN to begin after you go Off Hook. With your original setting of 100 (milliseconds not seconds) your unit began to dial before the PSTN was ready to accept any digits. There are many Registers in a PSTN where the Dial Tone originates from. Some are a little faster and some a little slower. You were on the edge with 100 ms.. I suggested a 500 millisecond (1/2 second) delay before dialling to remedy the situation. With your setting of 750 ms. you are even safer. If you go too high, then the call will take longer to process after you finish dialling. The Current Disconnect value of 300 means that if the PSTN sends a Break in the Loop (PSTN Line) for 300 ms. the Gateway will release its connection and the call will be disconnected. This is the Calling Party Control (CPC) in PSTN talk. (Similar to a SIP "BYE") Well now you can try using and learning a Grandstream GWX-400x analog adapter. In some ways it is similar to the GWX-410x and in other ways it is 100% the opposite. With it, you will be able to add a cordless telephone and other non-SIP telephones or devices to your PBX. Bill Hayhurst

Try this. Add a 6 in front of the number to be dialled in the "Call Forward All Calls to:" Option Box for Redirection for Extension 61. Or skip the Redirection and simply put 6 plus the number to be dialled in the "Cell phone number:" Option box under the General tab. Also set "When calling the extension: to "Immediatley" I did exactly what you are doing, except I used 8 instead of 6 and it works OK. Bill H

You are missing a CPC signal from the Panasonic telephone system. Most Panasonic telephone systems do not extend the CPC (Calling Party Control) disconnect signal to the analog station ports. The CPC is a brief (Wink) open in the loop current when a disconnect has taken place. You discovered this when you stated that the disconect only worked when you unplugged the line (PBX extension) from the CS 410. If you plan to use it behind a PBX then you would need a CPC emulator device. If you use the CS 410 with "normal" PSTN lines you get the CPC OK. If you are technical to the extent of using an LED, connect one in series with the line (circuit) going to the CS 410. It should light when the circuit is closed. (talking). If not then reverse the LED in the circuit. When the call is completed you may be able to see a quick WINK OFF of the LED before the disconnect takes place. Bill H

I have a GXW-4004 and it is registering directly to Callcentric.com. Mine works fine. No echoes and you can adjust the gain for both receive audio and transmit audio. Make sure you update it to the newest firmware. It adds a few more nice controls. I have not used any other brands of FXO or FXS gateways, but for the money I believe it is a good unit. However, the online documentation (PDF) is not current as of 6/16/07 for the latest .44 firmware. The existing documentation is good, but it could be better by being more in detail. This adds to the steep learning curve. The Grandstream free tech support so far has been by Email and runs about a day or so delayed. That's OK if you are setting up PBXnSIP for yourself and have time to spend. If, however, you are in the business of selling and setting up IP Telephone Systems, then its a different story. Be prepared to spend several hours learning the unit, its worth the investment. Also, the first unit I ordered was bad out of the box. It would not pull a DHCP IP address. Stayed at 0.0.0.0. VoipSupply (a good company) where I bought it wouldn't even talk to me about the trouble. They have a "No questions asked return policy" for returns. I guess that also means I can't ask any questions too... They did replace it, but it took about a week in transit. Still, a good unit. I would buy more of them... Bill H

What version of the firmware are you using? Just a few general pointers on the Grandstrream FXO that I discovered after many hours of testing and observing. Some make sense and some don't. I guess its that way until you & I understand the unit better. Whenever you re-boot the Grandstream make sure it does re-boot by watching the LEDS. If they all don't go out and then back on again, it did not re-boot. (even if they do, it still may not be updated - at least thats what I found to be the case) I set up a Syslog Server (KIWI) on another computer and it says something like "channel: 0 is busy, user req of reboot is postphoned for 10s.". So if you don't see the LEDS go off in say 10 seconds, then do a power on power off cycle to actually re-boot it. Othewise the changes you made are not actually made in the unit. Anyway, I have updated mine to the .55 version. When you set up the trunk in pbxnsip, make sure it is set as a SIP Gateway. Add the IP address of the Grandstream only in the Domain field. Make CO Lines co1 co2 co3 co4 etc... I don't think that has anything to do with the operation of the lines. The Password doesn't matter since we are not registering the gateway. In the Grandstream there are several items to change. In Profile 1 put the IP address of the pbxnsip in the SIP Server field Also, change SIP Registaration to NO. Thats it for that page. On the Channels page: Put a number (1,2,3,4,5,6,7,8) for each actuall working PSTN Line that is going into the gateway in the CHANNEL(s) Field. Then set the Profile ID to 1 for working PSTN Lines ONLY. Set others (unused PSTN Lines) to Profile 3. Thats it for that page. On the FXO Lines page: Set Wait for Dial Tone to NO for all channels Set Stage Method to 1 for all channels Set Min Delay Before Dial PSTN to 500 for all channels Leave everything else on the page in default settings. That is how I set mine up. And it did take about a day or so to figure it out since I was learning as I was doing it. If you are in the USA and have toll free dialling you can call me. Email first. If you set up a FREE SIP Account with WWW.Callcentric.com you can call me on 1 777 238-8032. Bill Hayhurst

I have found that a call that is made directly to an extension with Cellphone ringing will work OK. However, if the call is directed through an IVR, ACD, Hunt Group or Auto Attendant it will not ring the Cellphone. I think the message you are receiving is due to the PBX trying to connect to an EXTENSION NUMBER that is really your Cellphone Number. Bill H