rai

Pitty, I'm also looking for this. I've some Siemens GIGASET S450IP phones and an old GIGASET 4000 confort. I now have to keep the old basestation up and running because the voicemail is in that box and this one does flash the light if there is a new message. With IVM I can't do this.

What also would be great is if Uplink can forward the name or number of the person that is calling you on Skype. Because now if I have a missed call on my SIP phone, I have to look in my server with Skype <-> Uplink running to find out who was the caller.

Did you also solve the problem that in the log-window of AXON every 11 minutes an error line is coming by to tell you that the SPA can't register? If not, how did you solve this. In the link you are providing http://wiki.pbxnsip.com/wiki/index.php/Lin...PA-3102_FXS/FXO , the writer is talking about firmware version 3.4x and 3.6.x. If I look at the linksys server, the only firmare for the 3102 is 3.2.10(GW). Which version are you using?

With the EXPRESS software I'm having a ringtone, but with the SJphone from SJLabs it is a continues tone. If I connect direct to my SPI provider both programs are working fine. I guess that AXON is not provinding ringtone options.

Can't you use the logging option in AXON? This will create a log.csv file on the server. If you open this file, you can see all in and outgoing calls, including time of call and duration.

With me it doesn't make a difference if I put it to YES or NO in the section: Make Call Without Reg: no Ans Call Without Reg: no But I have to add also the IP_address and the portnumber of the SPA. Only problem I'm still having, is that in the AXON monitor window, I'm getting every 11 minutes a message that the external line can't register. Not realy important, because everything is working, but it would be nice if you could suppress that message somehow. Anybody an idea?

I know it works from the Softphone, but is there perhaps a *xx combination that makes it possible to do this from a phone that is connected to an FXS?

Everything works fine in AXON, but since I've installed AXON on my new Windows 2003 server, I can't delete the logfiles by using the browser. I can read them and everything else in AXON I can change, add or delete without a problem, but not the logfiles. It doesn't matter if I try to do this from my XP PC or on the 2003 machine direct. I only can delete them by using the LOG option in the AXON program itself and select delete from the OS. PROBLEM SOLVED, FIXED IT MYSELF.

SPA3102 = SPA 3000. All works now.

Well the magic one is the "PSTN CID For VoIP CID". THis has to be set to YES. Int eh SPA 3102 it's set to NO be default.

Somehow I got it now working. Can make calls from the handset connected to the SPA-3102 (should be identical als the SPA-3000) and even from my softphone on the PC. Also get in call's from the outside world en every handset is ringing. So far so good, but if someone is now calling me I get an caller-id on the handset not the number of the one that is calling me, but the info from what has been filled in by "PSTN Line" from the section "Subscriber Information". It is showing on the handset" "SipuraFXO (that's how I called the SPA) and as telephonnumber it's showing 111 (that's the user-id). My guess is that in the "PSTN-To-VoIP Gateway Setup" part (were I have set "no" to "PSTN Ring Thru Line 1") you have to fill in something in the "PSTN Caller ID Pattern" field. Is this realy the solution, is someone having htis same problem and how to fix it?

I had to buy a SPA-3102 because the SPA-3000 is not EOL. Plus here in the Netherlands ABEL Telecom had bought every SPA-3000 available to start a VOIP telecom provider. And those SPA-3000 they are selling are now overprotected and can't be rest to factory default anymore. So I had to buy a SPA-3102. I'm now trying to get this 3102 working with AXON. The FXS part is working, but I can't get the FXo part to work. I've been reading al the articles on this forum but it still doesn't work. 1 major thing I'm not understanding is how you get the old PSTN line available as external line. If you follow the rules, then with an FXO you must leave the proxy field empty. But then it want be available as dailing-rule in your rules. So I will never get the old outside line up and running. What I've got to work is the incoming line now, but outgoing is still impossible.

I've got it to work. In Axon I've added in the dailing plan a rule that if the number is 501 it's removing 0 digits and the prepend is the name of the person you want to call in Skype plus an @. So if you want to call echo123, then it will be: echo123@. The "dail on line" is then the external line you have created for your uplink line. In uplink use the option "When SIP calls Use the dailed number". AXON will now translate the 501 to echo123@. Uplink will now pass the echo123 to Skype.

I'm also having the problem that UPLINK is freezing SKYPE. This is only happening if I'm trying to use SKYPE itself on that PC. So if someone is calling me on SKYPE and UPLINK is transfering it to a SIP phone or v.v. it's OK. But if you answer it on SKYPE then the problem is starting.