[Linksys SPA-3102] Where is this products's page?

Hello
I went through all the sections in the Products list, but I still can't find the web page for this product. I'd like to download the latest firmware.
Does someone have the URL please?
Thank you.

this device has been migrated to the website of Cisco -- just go to www.cisco.com and type SPA3102 on the Search box at the top -- click on the first link and you should see an option to Download Software -- however, after you select firmware you will be asked to log in so you need to register first
| isolate! isolate! isolate! |

Similar Messages

200 OK message before call is established with linksys SPA 3102

I recently bought a cisco linksys SPA 3102 gateway to help me forward incoming VOIP calls to the PSTN network via the PSTN line. I also installed syslog to catch the sip trace. When i placed a call, after the SIP Invite and Trying, I immediately get a 200 OK reply from the PSTN LINE, just as soon as the calls is forwarded to the PSTN network for dialing. This 200 OK reply triggers the billing from the SIP side mean while the call has not yet been established.
Is there a way to stop this per-matured 200 OK reply from happening?
I will be very grateful for your help or hints.
Cheers
Emmanuel

I recently bought a cisco linksys SPA 3102 gateway to help me forward incoming VOIP calls to the PSTN network via the PSTN line. I also installed syslog to catch the sip trace. When i placed a call, after the SIP Invite and Trying, I immediately get a 200 OK reply from the PSTN LINE, just as soon as the calls is forwarded to the PSTN network for dialing. This 200 OK reply triggers the billing from the SIP side mean while the call has not yet been established.
Is there a way to stop this per-matured 200 OK reply from happening?
I will be very grateful for your help or hints.
Cheers
Emmanuel

Why does 9i AS have such lame EJB support? Where is this product going?

I installed 9i AS and really became disgusted with the product.
Oracle shows it's shameless corporate greed by hyping 9iAS as an EJB server -but 9iAS doesn't even support ejb 1.1 !! No entity beans! What a rip off...
Where is this product going? ejb 1.2 is already comming into the market....www.orionserver.com... BUT oracle doesn't even have 1.1 down yet....
What is the estimated development track for 9iAS?
As for me, I am ready to return my 9iAS copy...

Which version was giving you problems?
The latest version of O9iAS does in fact support EJB 1.1 (Session & Entity Beans with both Container Managed Persistence and Bean Managed Persistence). As well as Servlet 2.2, and JSP 1.1.
It does this through Oracle's Enterprise Java Engine, a highly scaleable reliable and available Java virtual machine. Unlike other JVMs, the EJE provides built in session isolation so programmers don't need to worry about thread collisions, concurrent garbage collection, or any of those things. Plus, you can compile the bytecode to native code before runtime for better performance.
That said... I agree the install could be a little simpler, but if you look at everything it's trying to install... you'll realize it's not trivial. Reading the documention is highly recommended.
-=< Nick >=

Linksys SPA 3102 not detecting hang up from Asterisk FreePBX

Hi,
I am forwarding calls from my pstn to freepbx using SPA 3102 (pstn to voip gateway). I have programmed asterisk to disconnect the call. It seems asterisk is working fine and disconnecting but pstn user still hear the call ringing. How can i make the call disconnect?
Thanks,
Rajeev

It's not be possible to solve issue described by rajeevraj22 in first message. According description, the incoming call has not picked up ("pstn user still hear the call ringing"). Unfortunately, POTS signaling protocol doesn't allow to reject the incoming call. SPA3102 can either pick up call or ignore ring signal on POTS line. In the second case the calling user is hearing ring back tone until he hang up or call setup timeout occur.
Despite your's (ildefonso_v1) problem seems not to be same (your call has been picked up first) there may not be solution available as well. CPC stands for "Calling party Control". CPC is signal sent from terminating PBX to called phone to indicate that the calling party has hung up. Terminating device (SPA3102 here) is recipient of CPC signal, not the source. E.g. CPC is related to opposite signaling direction that the one you are speaking about.
The call disconnect from end device to PBX is signaled by high impedance of end device.
I see when spa line is Idle the current Voltage is 52V, When a call is done the voltage is -7V. When I finish a incoming call from any internal extension or from asterisk the voltage is 52V again, but the call does not hang up in the other side.
OK. So idle line voltage is 52V. According your description, the SPA3102 disconnect properly from line on end of call (voltage rise to 52V). If the call doesn't disconnect, then it's not matter of SPA3102 but matter of terminating PBX.
The behavior like it is not so common in current phone networks, but older phone network switches allow no hangup from called side at all. Only caller is allowed to terminate the call.
All at all, you can't solve the issue from your side of wire. If terminating PBX is not willing to disconnect call immediately, you can do nothing with it. Ask your Telco operator for support. The behavior of particular line may be configurable. But don't put so much hope on it.
After 30 seconds I can see in the syslog POL REV -47 52 then the call hangs up in the other side.
It is the CPC signal from PBX to you. The PBX considered the call is over and is signaling it to you.
Rate helpful responses. It will help others to found solutions.

Disable calling name presentation on SPA-3102

Hi,
If I send a SIP INVITE to my SPA-3102, where the From header is like this -- (spaces inserted to stop the forum software treating it as an email address -- they're not there in the real invite)
From: Caller Name <01234567890 @ my.sip.server.net>;tag=as4b617ab1
-- the SPA-3102 generates a Caller ID spill on its FXS port with 'Caller Name' as the calling name, and '01234567890' as the calling number. That's all well and good.
If the From: header doesn't have a caller name, but is like this instead --
From: <01234567890 @ my.sip.server.net>;tag=as4b617ab1
-- the box sets the calling name to be 01234567890 as well.
Is there any way to turn that off, and have the SPA just not present a calling name at all?
If not, no bother! I'm just trying to get my box to behave a little more like BT with regards to caller ID presentation -- they don't ever send a reason for no calling *name*, but if the calling number is withheld or unavailable they will set the calling name to Withheld or Unavailable -- and set a reason for no calling number.
Many thanks!
Martin
Message was edited by: Martin Thorpe -- hopefully removed the auto-'email address' tagging! (Argh, no, it didn't. Bodged a different way.)

Hi Lindsey,
Thanks for the quick response. Here's a complete SIP invite -- I've changed the telephone number and put spaces around @ signs again, but everything else is unmodified.
INVITE sip:spa-line1 @ 81.2.113.115:5060 SIP/2.0
Via: SIP/2.0/UDP 81.187.239.177:5060;branch=z9hG4bK4062e0e9;rport
Max-Forwards: 70
From: ;tag=as75e22314
To:
Contact:
Call-ID: 445f75c33908fff74829a514159e9946 @ sentry.met24.net
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Mon, 29 Oct 2012 19:51:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286
So there is a contact field in there as well.
That's from a slightly patched Asterisk server, which doesn't put a calling name in if it's blank -- by default if you didn't set a calling name, Asterisk will also set the calling name from the calling number and you'd get this instead:
From: "01234567890" ;tag=as54c7bb08
I've done product management myself so I know one customer asking for it to work a little differently (as opposed to it doing something wrong!) isn't going to make a change -- that's no problem at all. If it were to be changed, I'd rather the ATA didn't generate a calling name field in the CLID spill at all, rather than 'Unknown'. But hey, that's just my opinion!
For the avoidance of doubt, the ATA is always generating the calling *number* field in the CLID spill correctly.
Thanks again!
All the best,
Martin

SPA 3102 Admin Guide

Where can I download the Admin-Guide for the Linksys SPA 3102 ?
I don't remember how many times I tried to find such a link and how many hours I spent so far with it. I could find answers to this question but the provided links were not valid anymore.
Thank you for your help.
Solved!
Go to Solution.

Finally I managed to find a working link in this forum. Why can this document not be made easier avalable? It would have been an advantage if it could have beeen found by using the search on the Cisco home-page.

HT5163 I need one Ipad 128Gb with Retina iPad (3rd generation) Wi-Fi + Cellular (Black colour), where should I approach in Delhi/NCR for this product? I am unable to find this in Croma and other retailers.

I need one Ipad 128Gb with Retina iPad (3rd generation) Wi-Fi + Cellular (Black colour), where should I approach in Delhi/NCR for this product? I am unable to find this in Croma and other retailers.
i need one very urgently for my day to day work. Can anyone suggest from where I can get a new piece(boxed) on or before 11 March 2013? As this is already launched in India in Jan month, but still not available with retailers, might be because of less demand of this very product.
Any help will be much appriciated.
Waiting for some clue.
Pawan Sharma

The 3rd Generation iPad does not come in 128GB capacity, only 16, 32, and 64GB capacities.
The 4th generation iPad does come in 128GB capacity.

And where did THIS document go: Portable Product Sheet

Until a few months ago (March 2011, I believe), there was a webpage entitled Portable Product Sheet. This page is very simple yet they contain very important details about the performance of Cisco appliances. Then sometime in March or April 2011, someone moved or deleted the webpage. All there is left is the router performance sheet.
Does anyone know where this very valuable page went? Can someone from Cisco please restore this?

Thank you for following up with your account manager. It’s important for them to know how much you value this information! Many of these docs may still be available, just not in the single location you found so useful.
There is a creative way that you can access the archived information. Go to the Internet Archive Wayback Machine, enter in the Portable Product Sheet URL, and click the Latest button. You can download the legacy documents from there. Keep in mind, of course, that the versions you get may not reflect the latest information.
Thanks again,
Janel

Where is the Help or User's Guide for this product?

I just installed Photoshop Album SE 3.2. Now I want to learn how to use it. I select Help from the menu, then select Adobe Photoshop Album Starter Edition 3.2 Help, and that takes me to the product download page; no user's guide, reference, or anything resembling them. Searching Adobe Support didn't get me anywhere. Is there help anywhere?

try http://www.adobe.com/products/photoshopalbumstarter/pdfs/psase3.2_help.pdf
Enjoy
CA
http://www.startereditionuser.com

SPA-3102 Line is disconnecting every 10 minutes and 40 sec

Hi,
I've a SPA-3102 connected behind a Motorola router, for any reason I'm not able to find, each time you call someone, after 10 min and 40 sec, the line cuts, and you have to call back.
Software Version: 3.3.6(GW) Hardware Version: 1.4.5(a)
SPA-3102 has a fix IP, and config has been done by my phone IP provider, does anyone have an idea which settings needs to be modified to avoid this cut?
Thanks for your help

I think under the “Regional” tab, you will have “Control Timer Values” and I belive thios is where you have to adjest the timer. I am not quite sure as well what parameters are needed to tweek in this section but I am sure that it should be one of them. Maybe try adjusting the “Reorder delay”. I suggest contacting Cisco Tech support to further look into your concern. I believe this unit belongs to the business series devices that Cisco is now supporting. Try to go to this link for the other business series devices and the site where you can get hold of Cisco for support:
http://www.cisco.com/web/products/linksys/index.html

How to trigger Linksys SPA 942 to send the info configured in Conference Bridge URL?

According to the Adminstration Guide for Linksys SPA 942, Conference Bridge URL field under EXT tab should be used to join into a conference call. However, when I press on CONF or confLx, I do not see the Conference Bridge URL gets sent in the INVITE message. Is there anything special I need to do to trigger the Conference Bridge URL to be sent?

Conference Bridge URL is the URL used to join into a conference call, generally in the form of the word “conference” or “user@IPaddressort”. By default it is blank so you need to fill the specific user and IP address with the appropriate port for the said settings to work. I further look into this matter and maybe you can be guided with this via this link:
https://www.myciscocommunity.com/thread/1457
Other than that I suggest contacting Cisco Tech support to further look into your concern. I believe this unit belongs to the business series devices that Cisco is now supporting. Try to go to this link for the other business series devices and the site where you can get hold of Cisco for support:
http://www.cisco.com/web/products/linksys/index.html

Extend telephone system extension to home - PAP2 and SPA-3102

Hi All,
I was wondering if someone could explain if this setup is easily implemented, or if it would be best left to a professional!
In short I am trying to extend one of my office telephone extensions to my home. I have found this product/setup on eBay which does exactly what I want it to but it carry's a rather large price tag.
http://cgi.ebay.co.uk/Voip-adapters-Extend-your-telephone-system-anywhere_W0QQitemZ280227103178QQihZ018QQcategoryZ34165QQcmdZViewItemQQ_trksidZp1742.m153.l1262
I already have the PAP2, and am planning on buying the other piece of hardware but I just wanted to check that this is something I could setup myself. Any and all advice will be gratefully received!
Kind regards,
Matthew
P.S. I am fairly familiar with networking but I have very limited knowledge on VoIP.

Hello,
look at this page http://www.provu.co.uk/support.html , check manuals named :
Linksys ATA Back-To-Back 1x SPA-3000 and 1x SPA-1001.(pdf)
Linksys ATA Back-To-Back 2x SPA-3000.(pdf)
Linksys ATA Back-To-Back 2x SPA-3000 and 1x SPA-2000.(pdf)
SPA-3102 has VoIP and PSTN part moreless identical to SPA-3000, just has additional build-in network router
SPA-2102 has VoIP part moreless identical to SPA-2000, just has has (like SPA-3102) build-in network router
PAP2T is similar to SPA-2000
PAP2 is very similar to SPA-2000
SPA-1001 is (except of dual voip registration) similar to one half of PAP2 - VoIP adapter with one port for analog phone
Thus using the mentioned info you shall be able to setup phone line VoIP extension.
Your knowledge of networking will come handy, the network setup is important part of this setup.
Message Edited by Scorpio-cz on 03-10-2009 10:11 PM

WRTU54G-TM/SPA-3102/Asterisk Disconnect Tone/Busy-Reorder tone?

I have a setup where I'm using the T-Mobile@Home Router (WRTU54G-TM) as a Trunk on my Asterisk system (PIAF). The WRTU54G (Phone 1 Port) is connected to the FXO (Line) port of the SPA-3102. I can making outgoing calls without any problems. However, incoming calls to my T-Mobile@home number once it hits the voicemail system on the Asterisk system and if the call hangs up before or after leaving a message, the "system" does not release the line and not do so unless I physically unplug the phone cord from either port (SPA-3102 or WRTU54G-TM). If I answer the cincoming calls and either party terminate the call, there is no disconnect issues; only when the call goes to voicemail. Is there any changes I can make to either the SPA-3102 or Asterisk, that will solve this problem/issue?
The problem seem to be related to:
a) CPC isssue and/or
b) Busy/reorder tone and/or
C) Disconnect Tones (does anyone know what the specs are for the T-Mobile system? Looks like this: 480@-30,620 @-30;4(.25/.25/1+2))
I saw on another site where an individual was able to do this:
..."Im running FreePBX on Asterisk and was able to use the busy/reorder tone by editing some lines in my zap channel config files. My solution was to simply program the PBX to detect that busy tone that T-mobile's @Home router makes after the call has ended, and use that as a signal to know when to hang up. Worked excellently, although the tail end of our voice mail message usually records a couple seconds of the busy signal... which I decided was not worth worrying about."..........
Not sure how I would implement a similar scheme, since I'm not using any ZAP channels or digium cards. Any help or suggestions welcome!

You could try to adjust this options on your SPA3102 PSTN Line. Under PSTN Disconnect Detection.
PSTN Long Silence Duration
This is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if <Detect Long Silence> is yes.
The default is 30.
Try to lower the values.
And Also PSTN Silence Threshold:
This parameter adjusts the sensitivity of PSTN silence detection. Choose from {very low, low, medium, high, very high}. The higher the setting, the easier to detect silence and hence easier to trigger a disconnection.
The default is medium.
Regarding for the 480@-30,620 @-30;4(.25/.25/1+2. basically this it the default settings for the US Disconnect tones. No need for you adjust.
Hope this help

SPA-3102 and Fax

Here's the problem: I am currently using a fax-switch that answers the incoming line, listens for a fax tone and, should it hear it, forwards the call to a fax machine. Without fax tone, the call is routed to the SPA-3102 and treated as voice.
This setup works nicely, but has one BIG disadvantage: All fax switches 'steal' the Caller ID. I am now trying to skip the fax-switch and use the SPA-3102 directly, by connecting the fax machine directly to the phone port of the unit. Since the SPA-3102 has the ability to recognize incoming faxes, is it able to route the call directly to the phone port? Without actually bothering the connected VOIP equipment?
I have tried to find a solution all over the Internet, but I seem to either be to blind to find anything, or, it might just not work. Thanks for your answers and suggestions.
Michaela

Thank you. I knew there must be a quick fix. Though ring thru would make the fax machine take all calls, which would make incoming phone calls be lost. If things were that easy, I wouldn't have bothered to ask. I was expecting somebody with actual Linksys knowledge to answer my question. Thanks again.

Connecting an SPA 3102 after the computer

I am moving to a house in a fairly remote part of Australia without a telephone connection. To get the phone connected will be very expensive and take considerable time. I have mobile phone and wireless internet connection available but only with the most expensive provider. I am not a big user of the phone but every now and then, I have to make one of those calls that go through adozen menus and then leave you on hold for an hour, theb type of call that can send you bankrupt using a mobile service.
My proposed solution is to have a prepaid wireless internet account, using a USB dongle connected to my computer as the internet connection. Then to connect either a SPA3102 or a PAP 2T between the computer and a handset and use a VOIP service such as Voipcheap to make calls. With this setup, I would have unlimited landline calls within Australia for around $50 per year, much less that something like Skype.
Can I do this and if so, how? I have both devices and they are both presently configured to work connected to my Linksys gateway. Do I connect the computer to the LAN or eithernet of the SPA 3102 and will it just require a normal LAN cable? How do I assign the various addresses required? Do I need any extra software for my computer?
John Adams

Hi John -- Thanks for participating in the Small Business Support Community. Please consider posting in the section dedicated to Australia/New Zealand here:
https://supportforums.cisco.com/community/netpro/small-business/international/australia_newzealand?view=discussions.
Thanks,
Stephanie Reaves
Cisco Small Business

[Linksys SPA-3102] Where is this products's page?

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