similar to: Linksys 3102 - Call Waiting

Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7

I have two old Sipura 3000 original (green) boxes and they are running perfectly faxes going through, no echo. Firmware 2.0.13(GWg), hardware 2.0.1(1813) Does the number in the bracket means anything? In addition I have Linksys 3000 running the same firmware and same hardware (except the number in the bracket). I can not get to work correctly. The unit has a lot of echo, can not get rid of it,

I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). -- Joseph

Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk)

I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next to fix it. Any ideas? -- Alan Chandler http://www.chandlerfamily.org.uk

Sorry, I am resending this, I tried earlier, but I couldn't see it appear on the archives - apologogies if it appears double! -------------------------------------------------- My Sipura 3000 ATA died on me this morning. I had a Linksys SPA 3102 available which I would like to use as a replacement. Unfortunately, the SPA3102 is not able to register with the asterisk server - I am

This is probably better suited to a cisco forum, but thought i'd drop it in here also. Using a 7960 with *, and have a very specific need. If the end user is currently on a call on the 7960, and a new call comes in, i need the phone to: show a visual indicator of the call (pref flash the call light) emit a ringing tone *NOT* play a call waiting beep inline on the current call. currently

Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack

Will the DAHDI with OSLEC have any effect one ECHO in Linksys or Sipura adapters? I know Dahdi is a replacement for Zaptel. And Zaptel is mostly for internal cards not external units like Linksys. I have one Linksys 3K that with so echo so bad that it is pretty much useless. -- Joseph

Hi I am setting up an Asterisk System in an office environment, Incoming and Outgoing calls are working ok, but i am having a few strange problems regarding call waiting. With the SNOM 200 (firmware 2.02t) phones, if you are on a call and a 2nd call comes in, the call waiting beep is played and the light flashes, but if you hang up the 1st call, instead of the phone ringing, it connects the

So far no one I have talked to has either had this issue or does not know a answer. I currently run asterisk 1.0.9. I have two issues to deal with. 1. The caller waiting caller ID does not show on the uniden hand held that is hooked to a sipura spa 1000 or my Polycom ip500. 2. When a caller is calling in and I hear the caller waiting beep on the line when talking with somone, is there no way

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the "disable call waiting" in asterisk as well, but I have not been able to find any

I'm having an issue with call park on my new Linksys 922. It has soft menu keys for doing call transfer (which I always think is a good idea because it's amazing how every phone has a different xfer interface and people always get confused). However, I can't get a good call park working on it. It doesn't respond to the use of "#" for transfer (nor should I want it to,

Hello I successfully hooked up a Linksys 3102 SIP gateway (http://www.voip-info.org/users/683/21683/images/716/SPA3102_lrg.jpg) to an Asterisk server, but since it's connected to a PSTN line, I must make sure it cannot be used by unauthorized users from the Net. Actually, even legit users with an account on the Asterisk server shouldn't be able to use it (outgoing calls should go

I've tried to ask this before but didn't want to waste a lot of bandwidth. However, I have to give the params in order to make the exact question understood. We have 2 ZAP FXO connected to two phone lines(2xX100P) 3 phones connected to a TDM400P with FXS modules Usually a caller will be on ZAP/2 and the first outbound line in the g1 group is on ZAP/1. There are two (main) call waiting

Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines

I'm faxing from stand alone fax machine via linksys SPA3102 but most of the time only half or quarter page goes through. Did anybody have any experience like this? -- #Joseph

Hi All, I just experienced a weird issue and though I'd share. I have a pretty standard business PBX setup for a business customer, local extensions, Linksys phones, call comes in and rings local extension exten => 101,1,Dial(SIP/101,20,tr) the physical phone has call forward enabled to the users home, Time Warner residential line service. Intermittently all seems to work except when the

I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using standard telephones. I've been running them for the better part of this year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost and especially the ease of provisioning. In an effort to inexpensively bridge our office PBX (InterTel Axxess) to our VoIP network, we've opted to connect