Configuring SPA 3102 with Ooma Telo Problems

Does anyone have any experience with configuring the SPA 3102 with the Ooma Telo devise. I have been around and around on this with no luck. I have been able to get into my 3102 ip address fine, but I need a proxy server to configure the 3102 with my existing Ooma device. Ooma doesn't seem to have one to dish out. Ooma is also saying that they don't have a portal for configuring external devices to the Ooma.
Has anyone ever tried to connect the two? I'm just trying to cut my losses here and move on if this is futile.
Thanks anyone!
Duncan

Wrong forum, post in "small business voice - SPA phones". You can move your post using the actions panel on the right.

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Configure SPA 525G with 2911 ISR Routers

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I see discussion where people have already installed similar modules. And hence the question. Called up cisco, but they were helpless and asked me to buy new phones.

I have it working now. Thank you very much to both Dragan and Cyril for your help and guidance. It turned out the problem was a communication error between the ISP and me. We have two connections supplied by the same ISP, one for internet, one for telephony. When talking to their support we were mistakenly talking about different connections — the PPPoE credentials I was given related to the telephony connection, whereas I was trying to configure the internet connection. Once the confusion was identified I was given the correct connection info (which is not PPPoE after all), and I've got basic connectivity up and running quite quickly after that...
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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

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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

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 3102 : Setting up Voicemail

Hi Guys,
I have this SPA 3102 ATA and wondering how to setup the ATA configuration to use my phone voicemail instead of my ISP's voicemail.All incoming callers hear my ISP's voice message and are redirected to their message box. I want to use my Phone's message box instead. Thanks in advance.
-R

If you have a phone attached to the SPA with its own voicemail, there is usually a setting that you can make on the phone for the number of rings before the voicemail answers. This needs to be shorter than your voip provider's voicemail. Better yet, many (most?) voip providers have a way to disable voicemail at the account level and then you don't have to worry about their voicemail. This also solves the problem of a call going to the provider's voicemail when your phone is busy and you not realizing that there is a message there waiting for you.

SPA 3102: loosing connection intermittently

Hi,
I am using SPA 3102 and D-Link wireless router together. Connection is Cable Modem to SPA 3102 to D-Link DIR-615.
Both the phone and wirless works fine in this set up. The main issue is when I use utorrent, SPA 3102 keeps resetting (loosing connection intermittently). All the three lights goes of for a second and then comes back. End result is, I have to keep resetting the routers (power off for 10 secs and switch on). This is really annoying, I have to keep doing it atleast 3 or 4 times a day. I guess the problem might be with the heavy downloading/uploading with utorrent. Any way out ? Help appreciated.
aj

Ooops, sorry My first post was a mistake. Here is my question:
I have modem connection to a remote site.This connection must go from wire to Ethernet. The modems are 2-wire, not dial-up, and are embedded in a device, so there is no access to the serial port of the device.
I have two SPA3102. On the remote site the modem is always connected to the FXS port. This port is configured with SAS Enable=Yes and SAS Inbound RTP Sink = $IP. On the local site when modem is connected the conection is automatically established. With Wireshark I can see that RTP audio stream is flowing in both directions but signal can not be heared on the server side. What should I do to hear the inbound stream? Is it possible with SPA3102?
First I tried with SPA112 as SAS on remote site. The signal passes in both directions and can be heared on SPA112 port. Same settings. But SPA112 causes delay of 140ms and timing is critical for my devices. SPA3102 has only 80ms delay with RTP Packet Size = 0.02 and modems work with two SPA3102 configured as hot line. The problem with hot line is that in case of connection fail I have to go to the remote site to disconnect the modem and connect it again. Not very convinient.
Thanks
Svilen

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.

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.)

Receiving fax trought spa 3102 asterisk 1.4 and spa2102

I have some problem receiving fax from a pstn line trought spa3102 - asterisk and spa 2102. I have updated the firmware of the latest version for spa3102 and 2102 and I have setting the port of spa 2102 were is attached the fax machine as described in the CISCO Small Business Administration guide (page nr. 55 - using a fax machine) but i cannot receive fax at speed upper to 2400 b.
My setup is:
pstn line -> spa3102 - asterisk 1.4.29 -> spa2102.
The spa3102 is configured to pass all control to asterisk. When a call is answered and it is a fax, this call is transferd by operator to the internall extension of spa2102 where is attached the fax machine.
I don't know as configure the pstn port of spa3102 for fax receiving.
Can you help me ?
thank in adavnce.
e.

Thank you for your reply.
Ok, I report my solution because I think that can be userfull for other users.
I have resolved the problem in this manner:
1) on the SPA 3102 I have set the PSTN Line, audio section:
a) preferred codec G711u;
b) use pref. codec: YES;
c) silence supp. enable: NO;
d) echo canc enable: YES;
e) echo canc. adapt enable: YES;
f) echo sup enable: NO;
g) FAX CED Detect Enable: YES;
h) FAX CNG Detect Enable: YES;
i) FAX Passthru Codec: G711u;
j) FAX Codec Symmetric: YES;
k) FAX Passthru Method: reinvite;
l) FAX Process NSE: YES;
m) FAX Disable ECAN: no
2) on the SPA 2102 on the line were is attached the fax machine, audio section:
I have followed the setup as described in the document that I have mentioned in my first message that is:
a) Preferred Codec: G711u;
b) Use Pref Codec Only: YES
c) Silence Supp Enable: NO
d) Echo Canc Enable: NO
e) Echo Canc Adapt Enable: NO
f) Echo Supp Enable: NO
g) FAX CED Detect Enable:YES
h) FAX CNG Detect Enable: YES
i) FAX Passthru Codec: G711u
j) FAX Codec Symmetric: YES
k) FAX Passthru Method: reinvite
l) FAX Process NSE: YES
m) FAX Disable ECAN: NO
n) FAX Enable T38: NO
o) FAX Tone Detect Mode: caller or callee
With this configuration I can receive and send fax to 14.400 on PSTN line acros the asterisk and the SPA 3102 / 2102 device.
The phone attached on the line 1 port of the spa 3102 have not echo when answer or call on the PSTN line as the other phone that have access to PSTN Line.
The phone of the fax macchine have echo, but usualy is not used as phone.
E.

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

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.

Configuring SPA 3102 with Ooma Telo Problems

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