SPA 3102 Voice router and gateway

Hello, I was told to get this device but i do not know what is does and I did not find information on the Linksys website about its purpose. I am setting up a desktop PC as a webserver thru a cable conection that does not have a static IP. My understanding is that this device will cause it to behave like a static IP. Is this correct?
I followed the instructions to connected it but I don't know how to fine the "DNS" numbers to imput into the field
I am running Windows Vista Home edition as the OS. The server will run a voice response system.
Any suggestions.

First of all, you can get the information for the device on this site.
http://www.cisco.com/en/US/products/ps10027/
This unit is part of Cisco-Linksys Small Business Voice Gateways and ATAS.
By the way, static IP has nothing to do with the SPA3102. If ever you'd like to setup a
webserver, it requires you to set a static IP on the server itself (IP of the PC) and at least a
Static IP for your WAN connection. WAN Static IP will be provided by your ISP and you need
to subscribe that IP from them.If in case your server requires you to open a port number for
the Server then that's the time you forward it to the router.

Similar Messages

High time SPA 3102 supports TLS and SDP

There have been absolutely no firmware updates in years for SPA 3102
Its a pity that SPA 3102 doesn't support TLS + SRTP with SDP (which is the default
and widely used key exchange mechanism as supposed to some proprietary method
or MIKEY that no one supports).
Grandstream has greatly overtaken with TLS +SRTP in their HT.503 that also sells much cheaper.
I have about 20 boxes of 3102 and can't throw them. I need to buy about 12 more and wish to
have uniformity. Main problem I see with 3102 is TLS and use of SDP for SRTP key exchange.
Otherwise is a great product.
Please have your engineering team make this a high priority and try to provide firmware update
for all the 3102 hw versions.
Thanks,
Anz

> does not have the support for SRTP.
This is contrary to what a lot of information out there says. Are you sure ?!
I would like to be able to use a gateway that can talk to a 3102 using SRTP and mini-certificate without using anothe 3102. There was a mention of this by your representative himself.
I am not really interested in your upcoming ATA that is similar to SPA 3102.
Are you going to replace all my SPA 3102 for free ?!!
Why can't you just do a firwmare upgrade for SPA 3102.
Please provide accurate and proper answers, that would be appreciated.

RV082 Router and Gateway?

Is there a way with the RV082 to configure routable IPs as additional subnets on the LAN site and have the device route to these from WAN to LAN in gateway mode? It obivously works fine in router mode but then private IPs don't function.
I have a server that requires a full public IP for SIP trunking (on a secondary NIC). I've tried 1-1 NAT and I get unexpected behavior like loss of call control anf one way audio.
I was thinking this could be accomplished by directly editing the routing table but with the public entry configured the table looks accurate automatically but doesn't route.
For now I have an RVS4000 behind the RV082 and have the RV082 in router mode with my block of IPs. I want to simplify the setup and see noticable speed increases when the RVS4000 is out of the equation.
-trevor

Thank you all for your help. I appreciate your answers.
I want to mention something I discovered that I did not pick up from the documentation. When setting up my Vonage telephone adapter for a static IP address, I thought I had to configure the adapter as well as the RV082. It turns out that I could leave the Vonage TA in DHCP mode and still enter an IP address (on the RV082 in load balancing mode) for a given mac address. This had the effect of reserving that IP address and assigning it to the Vonage TA every time it comes on line. Then I could use that IP address to select which WAN to use and what priority or speed. Very cool... :-)
Poul
^..^

Voice Routing and Normalization Rules

Hello All,
I am very weak about Lync ENT voice. I want to configure the Lync 2013 for below scenarios.
PABX extensions – 2xx, 3xx.
Lync extensions – 4xx
Access code – 9 for external access (This will be added from the AudioCodes gateway)
What should i configure ?

please may i know the Build Numbers of latest patches ?
i have Cu4 on of Lync 2013 Server.
Should i want to add below line ?
$x = New-CsClientPolicyEntry -Name "ShowExtensionInFormattedDisplayString" -Value "True"
$y = Get-CsClientPolicy -Identity Global
$y.PolicyEntry.Add($x)
Set-CsClientPolicy -Instance $y

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

Syslog not being sent by spa-3102

I can't seem to get any syslog to come out of my spa-3102. I've enabled syslog putting the ip address in both the syslog server and the debug server in the "System" tab. I set the debug level to 3. I have my system in a "double NAT" configuration: internet -- spa-3102 -- NAT router -- LAN (incl. syslogd) I don't know how the SPA chooses which interface to send its syslog UDP to. Since my syslog server is on my LAN, I assume that I have to specify my router's external IP (assigned by the SPA-3102), and then I configured my router to forward the packets to my LAN machine running syslogd. I reconfigured my linux syslogd to accept remote syslog and to log local0 and user. Kind of complex I know, and multiple points where it could be failing. But I want the SPA in front so it can do QOS and so I don't have to worry about NAT for its SIP support. Any suggestions on what I might do.

Just responding to myself -- it seems it was some problem with my particular syslog server, nothing to do with the SPA. I set up an alternative network syslog on a different LAN machine, and it is working fine through the double NAT and forwarding of UDP 514. -mda

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.

WRT54GC Operation Mode: Router Or Gateway

The WRT54GC has two modes: Router and Gateway. What is the difference?
I'm trying to do the following and I'm not sure what mode I need for that
/-->LAN1 (192.168.1.*)
2WireModem/Router
\-->WRT54GC-->LAN2 (192.168.2.*)
Note 1: LAN1 and LAN2 have different IP ranges.
Note 2: I tried router mode connecting the WAN port of the WRT54GC to a LAN port on the 2WireModem. The 2WireModem then complained that there is another router behind it. Why does it care?

As indicated in the graph I have changed the IP of the WRT54GC to 192.168.2.1. But what mode do I have to set it (router or gateway) to allow two separate LANs?
Also, the computers connected to the WRT54GC need to have internet access through router 1. Somebody told me I need a static IP for this. Is that correct?
Thanks

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.

HELP: SPA-3102 Gateway Setup Question

Hello,
I would like to set up the SPA-3102 to do the following:
1. The adaptor is registered to one SIP account
2. The adaptor can make outgoing calls through two or more SIP accounts which are not registered.
3. Calls to local numbers and emergency numbers are routed to PSTN
4. Calls to SIP phones are routed to the registered SIP account
5. Calls to long distance and international numbers are routed to the registered SIP account.
6. If we dial with a prefix, calls to long distance and international numbers are routed to the 2nd SIP account which is not registered.
Currently, we have 1, 3, 4, and 5 working. But 6 is not working. Is 6 possible? If so, could someone help me with an instruction of how to set it up?
Thanks,
AVS

You setup the second sip account in one of the gateway fields. Let us assume you are using gateway 1, the sip provider is voipbuster, and your userid is avs. Gateway 1:
[email protected]
GW1 Auth ID: avs
GW1 Password: your_password
GW1 NAT Mapping Enable: (same as you have on Line tab)
In the example above the provider's sip proxy is sip.voipbuster.com. In the example above avs and your_password are the userid and password for a specific account at the provider.
You put a prefix element in the dial plan, let us assume you put #8 for the prefix and wish a 2d dial tone after you dial #8:
|<#8,xx.<:@gw1>|
You dial #8 and you get a new dial tone, you dial the number and the number is sent to the gateway provider for call termination. The sip provider that you use must allow you to make calls without registration. Some providers don't allow making calls without registration.
Message Edited by hw on 06-12-2008 11:45 AM

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

SIP phone and SPA 3102

Hi,
I don't have an ipPBX or call signaling server. Can I register a SIP phone on the remote SPA 3102 then call the remote number. SPA 3102 is on remote site, the FXO port is connected to a phone line. My SIP phone is on local site, connected to Internet.
Thanks

yytellmey wrote:
Hi,
Can I register a SIP phone on the remote SPA 3102 then call the remote number. SPA 3102 is on remote site, the FXO port is connected to a phone line. My SIP phone is on local site, connected to Internet.
Yes you can do that. You need to know the ip address where you are calling. This is called direct ip dialing. You can call the SPA3102 and have the attached phone ring, or you can call the SPA3102 and have it dial a call out the pstn line. It all depends on what you want to do.
Initially you can get it working with whatever ip address you have at the moment. For the long term, if you don't have a static ip address you can get a symbolic address from someone like dyndns.com and then when your ip address changes you setup some means, either thru your router that supports dynamic dns, or with a pc program to keep your ip address updated at dyndns for your symbolic address.
You almost always have to forward the sip signalling port in your router to the SPA3102. You may also have to forward the spa's rtp ports in your router to the SPA3102.
There are a couple of ways to configure the SPA3102 when you want to bridge the call out the pstn line. The simple way is to just return a dial tone to the caller and then the caller enters the pstn number they want to dial. A more complicated way is to send a sip invite to the SPA3102 and have the SPA dial the number. The latter method is more reliable because you don't have to send dtmf signals over your voip link.

Verizon quantum gateway router and 2 Xbox ones nat type open?

Hello, so Verizon sent me there new router quantum gateway, I have 2 Xbox ones. Upnp is enabled and I have both xbox ones set to static ip. I have one xbox in dmz and the other xbox has all the ports forwarded udp 88, 3074, 53, 500, 3544, 4500, tcp 3074, 53, 80. Both Xbox ones in settings / network show nat type open but in call of duty advanced warfare multiplayer/ xboxlive screen. what ever Xbox one turns on first says nat type open and the other says moderate. Hopefully someone here can help cause I can't find any information anywhere and Verizon technical support has no clue
Thanks in advanced

Hi you need to do two things
Step 1: You have to program the router with port forwarding rules for your xbox one specifically
See instructions here Network Settings on Xbox One - Xbox Support
Step 2: You have to program the router with port forwarding rules for your CoD AW
see instructions here: ports used for Call of Duty games.
if you're not terribly familiar with port forwarding, then there are some good guides online including some youtube video's.
You might not technically find a whole lot on the quantum gateway router specifically, but its identical programming as compared to the Actiontec REV I router
This site has the specific ports that the xbox wants
Getting Configured: A guide to configuring your NAT settings.

Remote Provisioning for SPA 3102 and PAP 2 Series

Hi,
can some one help me with SETTING up Provisioning server for SPA 3102 pap2 etc using TFTP servers (I want a detailed explanation ) please it is urgent

These urls might help:
HTTPS Based Remote Provisioning with the SPA2102, SPA3102, and SPA9000:
http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a33b6a.shtml
http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa3102/release/notes/SPA3102_RN_V5-2.pdf

Introduce second default gateway into policy-based routing and optimization

Questions:
1) How to get the second PBR_DEFAULT_GATEWAY address 10.20.20.3 into the policy-based routing for redundancy?
2) Any optimizations as more and more traffic (policy-based routed and otherwise) goes through interface Gi1/0/1?
Address range A.B.0.0/16 represents assigned Internet-routable addresses.
Network also uses 10.0.0.0/8, 172.16.0.0/20, 192.168.0.0/16.
DEFAULT_GATEWAY router participates in OSPF and injects the default routes 0.0.0.0/0 10.10.10.1 and 0.0.0.0/0 10.20.20.1 into OSPF.
PBR_DEFAULT_GATEWAY router participates in OSPF but filters out default routes injected by DEFAULT_GATEWAY router.
ROUTER_A participates in OSPF and receives default routes injected by DEFAULT_GATEWAY router.
ROUTER_A contains the attached policy-routing configuration that allows the subnet A.B.30.0/24 to route anywhere on the network and uses PBR_DEFAULT_GATEWAY as the way out.

Ok I will see if I can run out to work and try this today..
After thinking about this, If I need to get to local ip addresses (192.168.1.0 and 192.168.128.0), I might have to change my route map to include those ranges in an ACL, then assign the 172.20.200.1 as the gateway to get to those networks, with the last statement being the traffic to be sent out the firewall
for instance
# Access to one of my local networks
access-list 101 permit ip 172.20.200.0 0.0.0.255 192.168.1.0 0.0.0.255
# Send Internet traffic to ASA/PIX
access-list 172 permit ip 172.20.200.0 0.0.0.255 any
route-map pix-172-20-200 permit 10
match ip address 101
set ip next-hop 172.20.200.1
route-map pix-172-20-200 permit 20
match ip address 172
set ip next-hop 172.20.200.2
and so on?
I know I need to be in front of my switch to test the change from set ip default next-hop to set ip next-hop...
I wantto make sure I can still get to the local networks I need to get to.
I appreciate all your help, and I will test this later on today..
Thanks
Don Hickey

SPA 3102 Voice router and gateway

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