cisco spa3102 voip ata u0026 router manual

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cisco spa3102 voip ata u0026 router manual

The SPA3102 also supports one PSTN FXO port to connect to a Telco or PBX circuit. The SPA3102 includes 2 100BaseT RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to a broadband modem or router. The SPA3102 FXS and FXO lines can be independently configured via software controlled by the service provider or the end user. Compact in design, the SPA3102 can be used in consumer and business VoIP service offerings including a full-featured IP Centrex environment. The SPA3102 uses international standards for voice and data networking for reliable voice and fax operation. We have configuration guides for each of the 3 ways you may want to configure your device. We offer prepaid phone service and International DID numbers using our voice over IP system and an analog telephone adaptor (ATA). The solutions are designed for home phone service, business phone service, call shops, telemarketing firms and cyber cafes. InPhonex is proud to support Internet telephony equipment (IP Phones) including Sipura 2000, Sipura 3000, Cisco 186, Linksys PAP2 and other SIP phone adaptors. We also support Asterisk PBX, Trixbox and offer turn-key VoIP Reseller business opportunities to let entrepreneurs and businesses resell voice over Internet (VoIP) under their brand name. In this scenario, NAT will be disabled and only WAN port (blue) will be used. LAN port (yellow) will be used only during initial configuration. Default IP address of Linksys is 192.168.0.1, so you will need to set up IP address of your Ethernet card in the same range. Once you have verified this, please continue configuring your device with the instructions below. However, Linksys SPA3102 will send all VOIP traffice to its WAN port! Recommended is AVT (according to RFC 2833). If there is a problem with detecting disconnect tone, you can detect silence as well.

Both are dB of digital gain (or attenuation if negative) to be applied to the signal sent from the PSTN side to the SPA of from SPA to the PSTN side. The range is -15 to 12. The SPA is using line voltage to check if line is in use or not, thats why it is essential to set right values. Otherwise line can be considered as in use and you will get a busy tone. At this page you can check what is voltage of your line when it is On-hook or Off-hook. Set this value about 2V lower then Line Voltage Off-hook. We use peer-to-peer connection, so string in field Number 1 will look like: You can set up another buttons to dial different numbers. You have to enable switch and then you can pick which output you want to control. Also you can set a code here. Want to use your normal household phones to make VOIP calls whilst still receiving incoming calls on your normal BT number?It doesn't matter if your actual phone service is provided by someone else such as your ISP, the main consideration is that you have a BT-compatible master socket with an 01 or 02 telephone number. For the purpose of this document we will consider your provider to be BT.At minimum you have an existing, working, separate broadband router with a spare RJ45 LAN socket.At this point I would like to remind all readers that I love my wife Melissa very, very much.Maplins code AR34M. You can use the RJ11-BT adaptor to convert the RJ11 cable that came with your SPA3102. Note that RJ11 is a different size from RJ45. If you are using an old modem lead, be aware that some are 2-wires straight through (may work), some 4-wires straight through (will work) and some 2-wires crossed over (rare, but will not work).It stores a small amount of charge which is released in two short bursts to create a ring-ring sound. Originally, this burst was enough to cause an electric motor, or a lever attached to an electromagnet, to strike a bell.

UK BT telephone lines have remained with this system, and BT compatible telephones will not ring unless they receive this burst of charge- even though their ring is electronic rather than a motorised bell. Your BT master wall socket contains a capacitor, but the effect of this is lost when you go through the SPA3102, because your phones are no longer directly connected to the master socket at the wall- the SPA3102 is a little telephone exhange in its own right. So you have to add a capacitor between the SPA3102 and your BT handsets in order for them to ring. Without the capacitor, you can still make and receive calls, but the phones will never ring.We are going to do this bit with it turned off.Maplins code AR34M).Maplins code VD36P) into the SPA3102's PHONE socket. You should now have a BT socket hanging off your SPA3102.When turned off, the SPA3102 connects the LINE to the PHONE socket directly, and we are going to test this connection.You should hear a dial tone. Try calling a telephone number, such as your mobile or 0800 144 144 (BT chargecard automated service). If this doesn't work, your cabling is wrong.Your landline phone should ring as normal. If this doesn't work, most likely you are using a secondary adaptor (no ring capacitor) instead of a master adaptor (with ring capacitor). If you're desperate, you can use a spare ADSL filter to convert a secondary adaptor into a master adaptor.If this doesn't work, your cabling is wrong. Due to echo cancellation on modern BT exchanges, you may have to tap the receiver or do some heavy breathing (ooh-er) in order to hear yourself.Until this is fully completed, which may take about half an hour, you won't be able to make or receive calls through the SPA3102 in the manner you'd normally expect. Therefore:Plug your normal phones back into the BT master wall socket as if you didn't have an SPA3102. Check that you can ring in and get a dial tone as normal.

I am going to assume this is not the case for your domestic situation. I am going to assume that you already have an existing, working, separate broadband router that you wish to continue using.By default this only works for the Ethernet (LAN) socket. If you are not using another existing broadband router, changing this is unsafe, and could give everyone access to your VOIP router, including allowing hackers to make calls on your phone bill and allowing hackers to get access to your computer network. Since like one in twelve white males, I am colour-blind (red-green colour vision deficient), I will not be referring to these colours.You can do this either directly, using a crossover cable (easiest way) or indirectly, through your existing network.Don't ask me what colour.If not, your SPA3102 may be using a different IP address and netmask - see the documentation in the box to find out what.Change the Static IP, Netmask, Gateway and Primary DNS to the correct values. For instance, on my networks all computers are 192.168.0.something and the broadband router is 192.168.0.254, so I use 192.168.0.252 as my Static IP, 255.255.255.0 as the netmask, 192.168.0.254 as both the gateway and primary DNS.Otherwise use any Stratum 2 UK NTP server from this link. This will mean that the log files and last call details have the correct date and time. Don't worry about this otherwise.Change your IP address back to whatever your LAN normally uses (eg. DHCP). Check that you can still access the Internet.Yes, you are plugging your LAN into the WAN socket, as I explained in my assumptions above.You should see the configuration page again.Well, the bad news is there are lots of other differences, such as the pitch of the dialling tone, and the SPA3102, despite being shipped with a British mains plug, has factory defaults of the USA. So we'll be spending some time reconfiguring it to sound and act like a British telephone line.

The SPA3102 will generate its own ring tone, its own dial tone, its own call waiting tone, everything. It'll only connect to your BT line when it needs to, such as an incoming call. So in order for your existing BT telephone handsets to work, and more importantly, your wife not to complain about the phone doing odd stuff, we need to make sure it acts as much like a BT line as possible.If you intend to use the SPA3102 to do more advanced things such as voicemail or call redirection, you'll probably want this to be Yes, but that's beyond the scope of this document.If you intend to use the SPA3102 to do more advanced things such as voicemail or call redirection, you'll probably want a shorter value, but that's beyond the scope of this document.The SPA3102 should now route all incoming and outgoing calls via the BT line even when switched on. If you've unplugged the SPA3102 from your phoneline and handsets, plug it back in.You should hear a dial tone. If this doesn't work, your cabling is wrong.Your landline phone should ring as normal.Due to echo cancellation on modern BT exchanges, you may have to tap the receiver or do some heavy breathing (ooh-er) in order to hear yourself.If it works when the SPA3102 is off, the problem is with your configuration. If it doesn't work when the SPA3102 is off, the problem is with your cabling.This means that lots of companies can all use the same protocol. This is also the protocol that your SPA3102 uses. Skype, at the time of writing (Jan 2008) does not use SIP and you will not be able to use Skype with your SPA3102.Most of them will have rates far cheaper than BT. For example, my wife makes a lot of 01 and 02 calls, but I occasionally call my sister in Holland.Many VOIP companies offer bundled, inclusive or free landline minutes for a period if you spend a certain amount, so pick one of those.Some providers may take this in Euros or US Dollars. I currently use WebCallDirect, so I will be using this in my configuration examples.

These can usually be found on the FAQ or technical support pages. WebCallDirect's SIP configuration details are here. This is required in order for your name to show up as the caller when you ring your friend's mobile phone, for instance. If you do not do this, your Caller ID will probably be withheld, and people you ring won't know that it is you who is calling.For example, WebCallDirect require you to download and run their MS-Windows application and use the account settings options to confirm your phone number with a test call. Once you've done this, you don't need to use the downloaded application ever again - which is good as far as I'm concerned, as I use Linux most of the time.This is not the case for us, as we want to route 100, 151, 999 and 0800 calls via BT still. Also we may, at a later date, want to route different calls via more than one VOIP SIP provider (one may be cheaper for UK calls, another cheaper for international calls). Therefore ignore any instructions that tell you to fill in the Proxy and Registration section, as this will route all calls via that provider by default!Gateway Zero is your BT line, also known as PSTN or POTS. Gateway One is your VOIP SIP provider. Time for another test!For example, if my UK mobile number is 07711 456789, then I would dial 0044 7711 456789. After a moment, it should ring. Talk to yourself, tap the receiver or do some heavy breathing to confirm it works. Don't worry, we're going to add in some rules so you don't have to dial the full international number, later.This should fail and thus prove that your calls are going via the Internet because there is no operator on the VOIP system. Don't worry, we're going to add in some rules so that operator and local calls work as normal, later.A common problem is that you may need to enable NAT Mapping, depending on how strict your broadband router's firewall is (some broadband routers are shipped with very strict firewalls by default).

So:On some Belkin routers there is a hidden SIP ALG page at: I'm going to give you a call plan which routes operator, faults, emergency and freephone numbers over the BT line, routes local, national, mobile and international calls over the internet, and bans directory enquiries and premium rate numbers.Change the bit at the end, that says 00441242, to match your local home dialling code. I live in the Cheltenham area, so my local code is 01242 and the international version is 00441242. If you live in Birmingham, your local code is 0121 and you'd change it to 0044121. If you live in London, your local code is 020 and you'd change it to 004420. Make sure you don't change the chevrons or colon around it.Basically the only difference is that we use gw0 instead of gw1 for 01, 02 etc. numbers.There's also a lot of discussion on the Voxilla.com forums.If you have BT paperless billing, you can also check on BT.com that you aren't being charged for calls on BT.It only takes a typo to change gw1 to gw0 and your calls will run up an expensive BT bill.Leave this call going. Then, at the same time, use your mobile to call your landline number. If it is engaged (busy), then you've done something wrong - it's calling via BT. Yup, the SPA3102 generates its own Call Waiting system!The SPA3102 cannot generate a busy tone for incoming calls on the BT line, if you are already on a VOIP call over the Internet, because busy tones are generated at the BT exchange. Instead, when you are on a VOIP call and another call comes in on the BT line, you will hear quiet beeps in the background. You can either ignore them, or you can put down the handset and it will ring with the new incoming call. If you have an R (Recall) button on your handset, you can use this instead to switch betwen the two calls whilst keeping both of them going!If you then press 3 to return the call, this will also route via BT and you will be charged at BT rates instead of VOIP rates.

The SPA3102 will then try to call the number every 30 seconds and will ring when it is available. It will give up after 30 minutes.If your internet connection isn't too reliable, and you're worried about consequently unknowingly running up a large BT bill, on the Line 1 page, under VoIP Fallback To PSTN, change:Codecs are methods by which the sound data is compressed, and there is a setting available to change this on the SPA 3102 (Audio Configuration - Preferred Codec). I strongly recommend you don't do this because the SPA 3102 does not have much processing power and it does not handle complex compression in a timely fashion; you will get garbling, echo or delay. Quite often you can end up in a situation where you can hear the call fine but the person you're calling can't understand a word you're saying, because it takes more processor power to compress data than it does to uncompress it. Leave the codec set at G.711 unless you are absolutely desperate for bandwidth. If you do change the setting, check that people you call can understand you; don't just assume that because you can understand them, it must be alright the other way around; most likely it isn't, and they're too polite to tell you. I'm not going to recommend anything other than G.711 so don't ask what I think is second best; I found that second best basically made me incomprehensible to the person I was calling.You can use it to get cheap international calls from your mobile, by configuring it to recognise your mobile caller ID and answer those calls with an international dialling tone. You can use it to interface to an Asterisk software telephone exhange and create your very own voicemail hell. You can use it to store your favourite phone numbers and access them with a shorter sequence. Plus lots, lots more - read the manual and forums for more ideas.Corrected duplicate Disconnect Tone.Thanks to Theo Markettos.Added section on codecs and why you should stick to G.711. Please try again.

Please try again.In order to navigate out of this carousel please use your heading shortcut key to navigate to the next or previous heading. In order to navigate out of this carousel please use your heading shortcut key to navigate to the next or previous heading. In order to navigate out of this carousel please use your heading shortcut key to navigate to the next or previous heading. Full content visible, double tap to read brief content. Please try your search again later.SPA3102 users will be able to leverage their broadband phone service more than ever by automatically routing local calls from mobile phones and land lines over to VoIP service providers and vice versa. If power is lost to the unit or Internet service is down, calls can be redirected to a traditional carrier via the FXO interface.To calculate the overall star rating and percentage breakdown by star, we don’t use a simple average. Instead, our system considers things like how recent a review is and if the reviewer bought the item on Amazon. It also analyzes reviews to verify trustworthiness. Please try again later. The Telco connecting FXO on this little box provided the bridge to the PSTN for that backup. I didn't have anything hooked to the FXS port that would be used for analog phone equipment. Since then I got tired of running my own PBX and the SPA3102 went on my shelf. In the last year I became interested in moving my home phone to a voip service. In my case Phonepower. It's fine and, yes, they provide a free adapter. Phonepower will support this adapter to connect phones to the FXS port if updated to the most current firmware. But they will support using up to two ports on an adapter. If you want a solution as a SIP gateway Phonepower or similar services the It also has 2 FXS ports to connect 2 phone lines for up to 2 concurrent calls. If you need one FXO port for telco connection like I did years ago the SPA3102 is probably the most widely compatible and economical choice.

The latest firmware appears to work really well (not so true 5 years ago). If you just need one or two FXS (for phones) ports get the newer models. The SPA3102 filled a need for me very well in it's original telco bridge role and continues to work a SIP bridge with its FXS port hooked into my home phone wiring. For being flexible and reliable it has earned 5 stars from me.I previously had tried the PAP2T, but the PAP2T only converts the 'sound' of the fax signal to digital and back over the line, which often introduces enough distortion to corrupt the fax transmission enough that it never goes through. The SPA3102 supports the T38 protocol, which detects that a fax is going through, and then digitizes the 'image' of the fax and sends the image to the voip provider, which then reproduces a correct and clean fax signal, preventing issues with distortion. In short, you can send faxes with this thing, which is what I was looking for. The voice gateway is set up with an almost indentical interface as to its older PAP2 cousins it is based off of, but with more options and customizability. I'm pretty satisfied with this gizmo.So far I have only used the FXO port, so my review is based on that part of the device's functionality only. The unit works well. The documentation included on the CD is a fairly complete technical reference, but don't expect much in the way of explanation or tutorial. If you are not comfortable reading manuals or online documentation, this is not the gizmo for you. I give it only 3 stars for ease of use because, well, it's a complicated device with lots of configuration options. The options aren't hard to use, but knowing (guessing?) what options to change is the tough part. The FXO interface works well. I found that I needed to change the POTS to VoIP gain to 3 dB for my weak phone line. Also I changed the answer on ring delay to 3 seconds (default is 16 seconds). Most of the out-of-the-box settings work just fine.

There are tutorials on the web for how to set it up to use with an Asterisk system. It's mostly a matter of getting the SIP parameters, digit dialplan, and the right codecs. The only negative thing I have to say is that the unit runs hot. I recommend putting it in a place where air can circulate around it for cooling. This unit is a great way to interface an Asterisk system to POTS lines and equipment. There's no software drivers needed either, all you do is edit the Asterisk sip.conf file to add support for the unit.It is true this product has a lot of features once you dig into the extensive technical manual that can be found online (but not easily from the supplier). The biggest con is that it has a lot of difficulties dealing with a not so stable internet connection. The unit does not recover well from these interruptions and I need to restart the modem and the SPA and be lucky if it registers itself again. Or it may take hours to re-register at the VOIP provider. I had a lot better experiences with the AVM Fritzbox back in Europe. I would not recommend this product to most users. Only if you have extensive technical skills and a very stable DSL connection it is worth a try. Connecting seems more stable for the last 24hrs.I wanted a lower phone bill and multiple Direct Inward Dial numbers so family in different cities can call us without long distance charges. This was relatively easy to configure as a novice. If you have some experience setting up a wifi router, you can configure this device. Once I had my service provider configured, I had it up and running in minutes using the basic mode configuration options. The adapter includes a large number of advanced configuration options if you have a specific need. It also includes a NAT style router feature, so you can connect it directly to your modem and have your home network behind it, or you can configure it to work behind most modern consumer routers without any special changes.

One nice feature, which I haven't used but I can see the value in, is the analog phone line pass through. If the VoIP service isn't working or you want to call a number not accessible through your VoIP provider (like 911), then this VoIP switch will automatically connect you to the regular phone service. Combining this with a regular cordless handset is cheaper than most dedicated VoIP handsets and has more features (like support for regular analog phone lines). With a decent commodity VoIP provider, you can dramatically lower the costs of having a household phone.Sorry, we failed to record your vote. Please try again Documentation was scarse. When I got it to work, I was a happy camper, until it started heating up and acting up, a few months after (so no more warranty, of course:) ). I switched to Grandstream products and haven't looked back since. They're cheaper and easier to configure (granted, to be fair, I also have more experience, but still.). Good day.Sorry, we failed to record your vote. Please try again Sorry, we failed to record your vote. Please try again. Simplified Users Guide. Version 1.1a. In Progress:) A Step by Step Introduction. Written by Jason from. JMG Technology. Thank you for purchasing Linksys SPA 3102. Page 1 of 3. VoIP Setup Guide for Linksys (Cisco) SPA3102. 1. Connect a telephone handset to the Phone 1 port. Check the IP Address of the phone adapter: 2. Linksys SPA-3102 download manual. Linksys SPA-3102 User Manual. This page contains the user manual in PDF form for the Linksys SPA-3102 router. Setup your Linksys SPA3102 for InPhonex VoIP phone service using Internet Telephony to make Free VoIP calls. Linksys SPA3102 is a high quality device that Cisco Small Business ATA Administration Guide. 3. Contents. Chapter 1: Introducing Cisco Call Forwarding to PSTN Gateway (SPA3102 and SPA8800). 98 Setup tab. SPA3102. Voice Gateway with Router. C. E Proceed to the appropriate instructions for your Internet. Connection Type: DHCP.

Static IP, or PPPoE. No, I couldn't find it either. Thankfully, I didn't need it. If you haven't read it yet, Jasons guide is pretty good, and all I needed to get my SPA3102 View and Download Linksys SPA3102 user manual online. Voice Gateway with Router.Reload to refresh your session. Reload to refresh your session. This article SPA3102 users will Figure 1 shows how to connect devices to Cisco SPA3102.In addition, when The SPA3102 FXS and FXO lines can be independently configured via software controlled Compact in design,The SPA-3102 uses international standards for. The Sipura products have literally hundreds of configuration options and can be quite daunting to configure. But, it only goes as far as v.2.0.9 of the firmware. Many of the newer parameters need to be documented but Sipura does not plan on releasing any new documentation. Download to a local computer and name the file spa.bin All company, product and service names used in this website are for identification purposes only, and do not imply endorsement.By continuing you are giving consent to cookies being used. I have tried to google for document how SPA3102 work in Singapore environment, but without success. I managed to find information on how to setup SPA3102 with Freepbx, but document was long and not very easy to read and follow. So I decided to write this document, hoping anyone like me, being task to setup VOIP, can follow and understand. Purpose of the Lab setup Objective 2 is off course to allow incoming call from analog line, to go to an Interactive voice respond menu, and select the option, and forward the call to the selected extension or softphone. Rest of the FreePBX feature, is not in this lab scope, and you should be able to find a lot of information on asterisk feature. There are also a lot of document covering on SPA3102 to connect to SIP service provider (In Singapore, service provider like hoiio), this document will not cover those too.

Setup step (Summary) This will provide the basic administration configuration version of the interface. 6. Now click on the “advanced” link on the top right hand side of the page. This will provide the advanced administration configuration version of the interface. 7. If you are able to access to the above, put the hardware one side, and setup the FreePBX. I have only configured to dial 9, to call an external number. Example, on the softphone dial 991231234, will put a call to PSTN line “91231234”. If you dial normal extension number, it will still go to extension number. Look under RTP Parameters and check the RTP Packet Size. Linksys has set this to 0.030 by default, which is not correct for use on ulaw (G711u codec) connections. Change it to 0.020 instead (or 0.02 on older Sipura devices). If you don’t do this, you may experience strange problems with “choppiness” or random clicks on some calls but not others, and you may also experience problems when playing Asterisk sound files. 3. Next go to PSTN Tab Keep it at 15 characters or shorter.Note that this must exactly match the DID number in your FreePBX Inbound Route setting for this device. If the number here and in the Inbound Route don’t match exactly, you won’t receive incoming calls. PSTN CID Name Prefix: (Leave Blank) 11. Leave everything else in this section blank. In rare situations you may need a slightly longer delay (5 should be more than enough). If you live in Australia, Canada, the United States or most other countries with modern telephone systems you probably won’t have to change anything except perhaps the gain levels, so we’ll only deal with them for now. But just so you know, here’s some information on those settings: If it’s set too high, the people on the PSTN side of the connection will be more likely to hear echo (they may hear their own voices echoed back from your end).

Also, any echo that has been reflected back to you will be heard at a higher volume level, and will therefore be more objectionable. If you have actual test equipment available you can fine-tune the volume settings for best results. Jeddah 23425. You may have to register before you can post: click the register link above to proceed. To start viewing messages, select the forum that you want to visit from the selection below. We have 6 Cisco Linksys SPA3102 manuals available for free PDF download: Administration Manual, Provisioning Manual, User Manual. Linksys SPA-3102 is a cool little device that can convert your analog phone line or fax machine into a SIP trunk into your trixbox or Asterisk PBX system.An ALG is created in the s. These adaptors do not support dialing via the Web UI or Ethernet, so only call monitoring is available. Press Scan to look for the next VoIP adaptor and fill in the IP address found if any. Once an IP address is specified, pressing Login will attempt to open your web browser to the device status page verifying your login credentials were accepted. You will need to know the IP address of your VoIP Telephone Adaptor if Phone Amego doesn't find it for you. Some adaptors provide a built-in router and 2nd Ethernet port, so you may be connected to the LAN port, while other configurations place the adaptor behind your existing router so you will be connected to the WAN or Internet port. Press 1 to enable WAN access to the administration web server. Using Notification By default, Phone Amego will poll your VoIP adaptor every 5 seconds to retrieve the call status. On a switched Ethernet LAN, the overhead is modest and responsiveness is comparable to landline caller ID which is sent between the first and second ring. If you have administrator access to your VoIP adaptor, you can configure it to notify Phone Amego when the call status changes.