SPA 3102 Port Forwarding

I am seriously stumped.....I have placed my SPA 3102 between my cable modem and my apple router. I would like to forward some ports from my spa 3102 to my apple router. I've gone into the admin console and defined the settings I want on the "Application" tab. Yet there seems to be no difference when I submit all changes. The darn SPA just doesn't seem to actually forward anything. After several hours banging my head against the wall I'd greatly appreciate any assistance. Thanks, -Chris

I agree the ideal solution is one where the Sipura is behind the Apple router. In fact I had previously been in the setup with my previous Netgear router. Currently there is a bug in the Apple Time Capsule that prevents the defining of a range of ports (ie. 10000-20000) therefore I was originally unable to get the setup to work. I have managed to get the Sipura to work behind the Apple router by enabling NTP-PMP on the Apple router. I'm a little concerned about this setup as I'm not familiar with how secure this setup is. I had at one point tried to set the Apple router in the DMZ on the Sipura and it always amounted to nothing. The darn Sipura just wouldn't ever allow for ports to be forwarded. In any case, thanks for the input.

Similar Messages

SPA-3102 - phone port dead (FXS?)

I have a SPA-3102 that one day stopped providing dial tone to the connected telephone. I can still see it on the network, configure it, and all looks fine there, but the connected phone gets no dialtone, no voltage, touch tones don't work, no IVR, etc.... I've tried several phones, so I assume it's the adapter. I've tried to find out how to get it RMA'd and sent to Linksys for repair, but every avenue I've tried to pursue tells me that Linksys doesn't support it and i have to go through my reseller... The reseller says to go the manufacturer. Can someone please tell me: A) Am I missing something simply on this problem? B) WHO to contact at Linksys to get a replacement. Thanks! Steve

for one, calls made to the PSTN line form the SIP or internet will definitely not ring the FXS port. You must have the 2nd account dedicated to the call going to the PSTN only,
these are the call flows for the SPA3102
incoming VOIP
VOIP to FXS
VOIP to FXS hop over to PSTN
VOIP( THRU PSTN line) to FXO( PSTN)
Out going VOIP
FXS to VOIP
FXS to GW0
PSTN ( FXO) to VOIP
PSTN to FXS ( PSTN RINGTHRU LINE 1)

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.

Create line extension between two SPA-3102

I`m having problems to create a line extension between two SPA-3102
I have one SPA-3102 connected to an analog PBX system with IP 192.168.0.201, and the other SPA-3102 with analog phone and IP 192.168.0.200
I succesfully setup them to make a call from the first to the second
But I couldn`t setup them to make a call from the second (192.168.0.200) and give me the dialtone of the PBX connected to the first SPA-3102 (192.168.0.201).
I could setup a hot line on the second SPA-3102 (192.168.0.200) and call to 192.168.0.201, but it doesn`t take the line to hear the pstn dialtone.
I saw many answers about this problem, but no one resolve the problem, i have the latest firmware. please, anyone could help me and if it`s possible to work please send me all the configuration needed.
Thanks again

Hi Jeremy,
I have a similar problem, I have one PSTN line (say Line1) with free minutes to mobiles, so its good for outgoing calls. The other line (say Line2) which i have is acually VoIP but it comes with its own hardware (magicJack if you have heard) so I can't use a SIP client and have to use the supplied Hw client, but it does give me an option to connect any normal phone to this magicJack (i suppose that would make it a fxs port). Now this magicJack is cheap for other people to call me.
I want to find a solution so that all the calls I receive on Line2 get forwarded to my mobile number via Line1. And if I receive any calls on Line1 they should be treated normally (my home phone rings). Do you have some idea how I can achieve this with minimal spend? Thanx
Atif

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

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.

Two SPA 3102 connection problem

Hi All!
I have 2 3102 (2 location: my location, other location). I'd like to call the other location's 3102 wtith my 3102 through the internet:
Caller: my spa's line1 fxs port
Called: other spa's pstn line's fxo port
All the 2 locations: the 3102's behind a nat (fli4l linux router). I use dyndns service on all the 2 locations.
Other settings (either 3102):
- 5060-5063 tcp+udp ports forwarded tho the spa's lan ip
- 16384-16482 tcp+udp ports forwarded tho the spa's lan ip
- NAT Mapping Enable:YES
CALLED SPA's PSTN:
- SIP Port:5061
- Register:NO
- User ID:any
- Dial Plan 1:xx.
- VoIP Caller Auth Method:none
I think this setting is correct
CALLER SPA's LINE1:
DIALPLAN:
(xx.:@other.spas.dynipname:5061)
I don't know what is the problem
I tried all the SIP tab's 'NAT Support Parameters'. No result
Please help me!
Thanks!

calibra wrote:
CALLER SPA's LINE1:
DIALPLAN:
(xx.:@other.spas.dynipname:5061)
If you have a userid on the distant spa (you show "any" ) then you need to include the userid in the address. If you have no user id then you don't need to include it. You can have the address in the dial plan or you can set it up in a Speed Dial. For a dial plan you would have something like this:
(S0<:any@dynipname :5061> )
For a speed dial you would have something like this
any@dynipname :5061
without the extra spaces which are here because of the forum syntax. If you put the speed dial in number 2 then you would dial it by 2#. In this case, your basic dial plan needs to allow single digit dialing.
You could also consider using HTTP digest authentication. The benefit would be that you can dial the distant pstn number directly on the calling spa when you make the call. You don't have to worry about dtmf digit transmission over the internet, the called number goes out in a sip invite.
To use HTTP digest authentication you setup the distant spa with that type of authentication. On the distant spa, under VoIP Users and Passwords (HTTP Authentication) you setup an AuthID, Password, and Dial plan. On the Calling SPA you put the distant spa dynipname : port as the proxy, the AuthID and Password (that you setup on the distant SPA) as the userid and password. Of course, you setup Register NO, Make and Receive calls without reg YES on both spas. In this http digest authentication setup your dial plan in the calling and the receiving spas would be (xx.).
You can only use the HTTP digest authentication if you are not using the calling spa for other calling purposes that would conflict with the proxy setting.

Crazy thing: No dialtone from a SPA 3102 inside the same network.

Hi folks: We have a couple of SPA-3102 connected to two POTS Bellsouth lines. Both 3102 are registering fine with our SIP server and also register the phone line.
The crazy thing is this: If we try to get dialtone calling from an IP phone (SPA922) LOCATED IN THE SAME LOCAL NETWORK (I.E: In the office) to the 3102 number, we get no dialtone.
If i try co call the same 3102 from a softphone (x-lite) also in the office, we still get no dialtone.
But if i try to call the spa from other network than the office, (I.e: My home), i´m getting the dialtone!!! I tried it with both the 922 IP Phone and the x-lite softphone and works fine.
Also, we tried to call the 3102 from other IP phone in our office in Mexico, and also works fine.
Our network topology in the office: Motorola´s Netopia 3347-02 with Bellsouth xDSL internet service. Two Linksys EZXS16W swithes, and a mix of laptops, PCs, servers, IPPhones (SPA922 and 942) and printers connected to these switches
Any idea ??
Thanks in advance.
Martin

If the in-office X-Lite is accessing the SPA-3102 on its private IP, try the public IP, or vice-versa.
I'm presuming that the SPA-3102 has a static private address, the Netopia is forwarding some UDP ports to the 3102, and that you are accessing the 3102 directly (not through a server such as Asterisk). In this case, access from the LAN via the public IP may fail, because many routers don't handle 'hairpinning' correctly. Access via the private IP may fail, because the client is being confused by the NAT mapping being applied by the 3102. You can use Wireshark on the machine running X-Lite to see what's happening in each case.
If your situation is other than the above, please provide the details: Accessed via server? If so, is server on LAN or outside? What software? Do you have administrative control over it? Does your router have a SIP ALG? Is it being used in this path? What port forwarding or other special setup on router? What NAT mapping parameters on SPA-3102? What have you seen with Wireshark? What remedial actions have you tried, with what results? Do you have multiple static public IPs? If so, is there a spare one?
Message Edited by Stew on 05-05-2008 10:48 AM

SPA 3102- STUN SERVER NOT REACHABLE

Hello,
Following are Line 1 settings of an SPA-3102
Nat mapping enabled=yes
Nat keep alive enabled=yes
Proxy server:correct proxy server specified
Outbound proxy server: Correct outbound proxy server specified
UserID and passwords are configured properly
SIP TAB:
RTP port min=16384
RTP port max=16390
Handle via received=yes
Insert via received=yes
Substitute via addr=yes
STUN enable=yes
STUN Server=aa.aa.com (correct STUN server configured)
Hadle via rport=yes
Inser via rport=yes
Send resp to SRC port=yes
STUN test enable=yes
TURN server=not configured
STUN test enable=yes
EXT IP=not configured
NAT keep alive INTVL=15
Problem:
On the REGISTER message I am receiving following warning
Warning: 399 spa "STUN Server Not Reachable"
User-Agent: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Would you please explain why I receive STUN not reachable warning? It is a reachable STUN server. SPA 3102 devices are placed behind ADSL modems of ITSP

I just noticed that you have Handle VIA received = yes, so if the proxy is responding with a 'received' tag, then the SPA would learn its external IP, without needing to contact the STUN server. I have a system that is behind NAT but does not need STUN or symmetric RTP, because it uses the IP from the received tag in the SDP. (In my case, I have the RTP ports forwarded to the SPA, so no port number translations are needed.)

Connect back to back 2 SPA-3102

Hello everyone,
I would ask a little help from any expert in order to fix the following connection:
(analog phone)--(SPA-3102)--(Access Point)--> <--(Access Point)--(SPA-3102)--(analog phone).
First of all is it possible to do it ? And also do I need any special setup in the SPA-3102 ?
The story is to transfer an analog line through air in another place near my home which doesn't have any analog lines at all.
Thank you in advance
vangellis

There are probably several ways you can configure it. I will outline one way that will ring the distant spa on an incoming pstn call and will return a pstn dial tone to the distant spa when the attached handset goes off hook.
The distant ata is a SPA3102. As an alternate you could also use a different but similiar Linksys adapter such as a PAP2.
On the SPA3102 attached to the PSTN line.
PSTN Line Tab
Line Enable: YES
NAT Mapping Enable: yes or no depending on the router used
NAT Keep Alive Enable: yes or no depending on the router used
Register: No
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
UserID: spaone
VoIP-To-PSTN Gateway Enable: yes
VoIP Caller Auth Method: none
VoIP Default DP: 2
VoIP Caller Default DP: none
none will give you the actual PSTN dial tone when the VoIP-to-PSTN gateway answers the call
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN Caller Default DP: 2
Dial Plan 2: (S0<:spatwo@ipaddress: 5060> )
The dial plan will automatically dial the distant SPA3102 Line 1 when the PSTN-to-VoIP gateway answers the call
Where spatwo is the userid on the Line1 tab of the distant SPA3102 (spatwo)
Where ipaddress is the ip address of the distant SPA3102
Where 5060 is the sip port number of the Line 1 tab on the distant SPA3102
VoIP Answer Delay: 0
PSTN Answer Delay: 2
Where 2 seconds is long enough to receive an incoming caller id on the pstn line (if you care about it)
Line-In-Use Voltage: 30
Where 30 is about half way between the on-hook and off-hook voltage of your pstn line (polarity disregarded)
Line 1 Tab
Enable IP Dialing: YES
On the Distant SPA3102 Line 1 tab
Line Enable: Yes
NAT Mapping Enable: yes or no depending on your router
NAT Keep Alive Enable: yes or no depending on your router
Register: No
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
UserID: spatwo
DialPlan: (S0<:spaone@ipaddress:5061> )
Where spaone is the useid on the pstn line tab of the SPA3102 attached to the pstn line
Where ipaddress is the ipaddress of the SPA3102 attached to the pstn line
Where 5061 is the sip port number on the pstn line tab on the SPA3102 attached to the pstn line
Enable IP Dialing: YES
Note: Depending on your network you may or may not have to have NAT Mapping Enable. If you do have NAT Mapping Enable you may also need to configure a STUN server depending on the router that you are using.
If the two ip addresses are not on the same local network you will need to forward the sip port in the router to the SPA3102 and you will need to use the external ip address.

SPA 3102 Possible MTU issue

I have a customer using the SPA3102 that is having issues connecting to netbanking sites, swap to another router and the service works fine (except the customer looses VoIP) this screams to me MTU issue,
Is their somewhere in the SPA 3102 that I can drop the MTU size for a PPPoE session?

there's no MTU setting on the SPA3102...since there is no port triggering on the SPA3102 as well try either DMZ or port forwarding (for port forwarding just forward port 443 which is for secured sites) -- this is under Router> Application tab
| isolate! isolate! isolate! |

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

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

SPA-3102s back-to-back

Hello, we are trying to use two spa-3102 boxes in a back to back configuration to provide phone lines to a construction site (via a Cisco wireless bridge) like this:
(PSTN)--(SPA3102 FXO PORT)--(Wireless Bridge)--(SPA3102 FXS PORT)--(Phone/Fax)
We have voice calls passing correctly in both directions, and the phone/fax machine at the construction site is able to receive faxes most of the time, but they are unable to send outgoing faxes. When the fax machine attempts to dial, you can hear the machine pick up, dial the number, and then there is a few moments of silence followed by a fast busy signal. Any ideas?
Thanks!

EvilFetus wrote:
When the fax machine attempts to dial, you can hear the machine pick up, dial the number, and then there is a few moments of silence followed by a fast busy signal. Any ideas?
Thanks!
If the dialing waits until the interdigit timeout and then gives a fast busy it usually means that the digits dialed are not correct. If the dialing is "back-to-back" as you say, you probably are dialing some code that sends a sip uri for direct ip dialing. There are a number of variables involved.
You should be able to view the last dialed number on the INFO tab.

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

SPA 3102 Port Forwarding

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