PAP2, SPA3102, WIP330 questions

Hello, I dont know pretty much in voip telephony, so please clear for me some issues =) 1) Are pap2 and spa3102 doing the same thing - transferring phone call from internet connection to regular home phone? Whats the difference between them? 2) can I use those devices without PC, just with internet connection? I wanna use voipstunt provider and they give me just username, so nobody will be able to call me from say, cell phone, just from the same application (voipstunt) from PC? 3) If I configure WIP330 with my voip provider, will I be able to make calls from anywhere where a hot-spot is present without PC? Do i need to get phone number from my voip provider? Thank you.

you can pretty much use the devices without a PC< but you will need a PC to configure the PAP2 and the SPA3102
the WIP can be used on a hot spot, and when registered to a voip provider, can pretty much give you IP phone access.
you may need to pay extra to have your regular calls routed to your voipstunt account.

Similar Messages

  • QUESTION ABOUT PAP2

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    from line 1. The first call will use g729r8, and the second call will use
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    Hello,
    I have seen this problem on the LinkSys RT31P2 product. Exactly the same issue. The RT31P2 has two FXS ports, but only one can use G729 at a time. May be the system is designed to work this way. I also would like to get answer and solution to this problem on the RT31P2.
    I started using now the SPA2002 but I did not test this issue till now.
    Message Edited by SamirG on 09-25-2007 02:07 AM
    Message Edited by SamirG on 09-25-2007 02:08 AM

  • SPA3102 questions

    Hey all, I just read the admin guide for the SPA 3102 but I wanna make sure I didn't misunderstand anything so, could someone confirm that I can do all the stuff below, and note I'm in the UK.
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    Message Edited by pepsi_max2k on 06-01-2008 04:29 AM

    thanks for that so basically, yes, it does everything i asked other than pstn fallback when voip fails. i can deal with that well, there is the no ssh thing that sucks a bit. other than multi voip accounts it then means my speedtouch is just as good (actually better as it has telnet access and logging, although only for voip calls).
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    Message Edited by pepsi_max2k on 06-03-2008 08:34 AM

  • Linksys Phone Adapter PAP2 question

    Can i connect two Linksys Phone Adapter PAP2 to one linksys router without creating any networking issues ?

    chojna wrote:
    Can i connect two Linksys Phone Adapter PAP2 to one linksys router without creating any networking issues ?
    Yes you can connect them.  Each will get a different local ip address and the router is supposed to keep the port numbers straight.  It is easy to change the sip port numbers on one of the PAP2 adapters and to also change the RTP port range on that adapter.  Then you don't have to worry that the router can't keep those port numbers separated and you more or less have to make this change if for some reason you need to forward the port numbers in the router.
    Message Edited by hw on 12-28-2009 08:49 AM

  • Is my pap2 faulty

    Hi
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    I think I'm right that the unit has to log into the SIP server to get a dial tone but I should still be able to dial 4 stars for the IVR menu?
    When I plug my phone into the unit it is dead not even any induction. I have checked and tried different cables and different phones but non have any sign of life.
    Thanks
    Alan

    In the manual I have for the PAP2 the power LED is either green or red.  I don't see anything for a blue LED.  Are you able to access the PAP2's ui?  If so try to flash or reflash the latest firmware if you are able to.  Also did you try to reset the PAP2 by using the phone and dailing 80# then hang up?  That is supports to act just like the reset button on a router.  In the manual I have it said if you have the wrong power adaptor connected to the PAP2 the power LED maynot light up or be red but nothing about blue.  I think if the reset or firmware are a no-go it may be defective.
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  • P2P SPA3102 setup with no SIP server/service provider

    Description:
    - Peer-to-peer ATA connection with no service provider or SIP server
    Location A (my location):
    - Linksys SPA3102
    - PABX analog device (Panasonic)
    - Public and static IP
    Location B (my partner location):
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    - Public IP
    Questions:
    - Can B ATA (D-Link) call to A ATA (Linksys) being different brands?
    - Can I pick up the B ATA attached phone and directly to have PABX internal dial tone at A, for other extension or external calls?
    - From PABX other extensions, can I call to SPA3102 extension number and directly redirect the call to B ATA?
    I was read many .pdf documents about SPA setup, and try differents configuration, but I'cant make the communication with my partner location happens.
    I think my problem is understanding and writing the correct PSTN LINE and LINE 1 Dial Plan parameters in my SPA3102.
    Can I make it happens?
    Thanks in advance,
    Bitman
    Message Edited by Bitman on 07-28-2007 05:37 AM

    Good day! I just hope this is not one of those “.pdf documents” that you have already read but basically this should work if you’re setting up two SPA3000 or SPA3102  since they have exactly the same Voice configuration:
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  • SPA3102

    I have some questions as below listed:
    1. Is SPA3102 can handled more than one call session at moment?
    2. How can i share the VoIP account in the LAN for other users?
    3. How many IP phones can connect to one SPA3102 for concurrent call sessions? (we have only one VoIP carrier account from lowratevoip.com)
    4. please let me have a solution for multiple users and concurrent calls in the LAN with one VoIP carrier account?

    1. Is SPA3102 can handled more than one call session at moment?
    the SPA3102 can handle 2 lines for line 1 , 1 for main call and 1 for call waiting.
    from the analog phone, you can also setup a dial plan for dialing out to 4 other gateways. the PSTN line has a 1 gateway call.
    2. How can i share the VoIP account in the LAN for other users?
    if you have for that one ATA, you can set those others to make call without regsiter to yes and answer call to yes, and regsiter to no.
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    this one is not possible. the SPA3102 cannot be used as a voip gateway or PBX
    4. please let me have a solution for multiple users and concurrent calls in the LAN
    with one VoIP carrier account?
    consider the SPA9000 PBX system that can can have mutiple sessions depending on the trunk line provided by the VOIP provider for you.

  • SPA3102 Help needed

    I have two questions on PSTN connections with the SPA3102. I tried the Linksys Support chat, but the agent there said he wasn't qualified to help and couldn't pass me on to someone who could help - very disappointing!
    Here are my questions (I'm in the UK):
    Q1 - Can I set the SPA to pass through the telco's PSTN dialling code, instead of presenting me with its own tone? This is so that I can hear the interrupted dialling tone that indicates a message waiting.
    Q2 - The SPA seems very slow to pick up an incoming PSTN call. I have not found any setting that will adjust this delay. Is there one?
    If someone here could help me I would be very grateful.

    RG99 wrote:
    Q1 - Can I set the SPA to pass through the telco's PSTN dialling code, instead of presenting me with its own tone? This is so that I can hear the interrupted dialling tone that indicates a message waiting.
    I have seen some discussion about how to do this if you are calling from the phone attached to line 1 but I don't think that technique will work when you come into the adapter over the pstn tab. The adapter really was designed to have you enter the dialing digits, then the adapter edits the digits with the dial plan and then sends the digits. Maybe someone knows how to circumvent SPA's dial plan, however it wasn't the way the adapter was designed.
    Q2 - The SPA seems very slow to pick up an incoming PSTN call. I have not found any setting that will adjust this delay. Is there one?
    The principal adjustment for the timing for the pstn-to-voip adapter to pickup an incoming pstn call is the PSTN Answer Delay. The principal adjustment for the timing for the voip-to-pstn adapter to pickup an incoming voip call is the VoIP Answer Delay.

  • SPA3102: Can I route a POTS line to a SIP phone?

    I need to temporarily get POTS service where there is, well, nothing. My idea is to use an SPA3102, a wireless link and a SIP phone (all of which I have) to solve the problem. At the required phone location, I would just have a SIP phone connected to a WiFi link.
    At the other end of the link would be the SPA3102. It's only connections would be to the WiFi link radio and the POTS line. No internet involved.
    First question: Can I do this? If yes, second is how as in how do I configure the SPA3102? If would be great if picking up the SIP phone got POTS dialtone but not mandatory. But, it clearly is mandatory that an incoming POTS call would ring the SIP phone.

    There is no network beyond the pieces I listed. The two ends look like this:
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    The radio link is point-to-point and effectively can be ignored for the configuration. That is, a POTS line, SPA3102 and SIP phone hooked together would look effectively the same and this is how I will test the system before deployment.
    As there is no external/internet connection, the IP addresses are arbitrary. Clearly, adding a computer with web browser to the system for configuration purposes is needed but, once again, the IP address is arbitrary. My guess (and it is just that) is that the computer can be plugged into the Ethernet port and the SIP side into the Internet port so no Ethernet hub would be needed but that is a guess.

  • [PAP2] Configuring Multiple Service Providers for Outgoing Calls

    I'm trying to configure my PAP2 with Gizmo project (for incoming calls) and VOIPCheap (for outgoing calls).
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    (xx.<:@voipcheap.host.xxx:5060; usr=myvoipcheapuserid; pwd=myvoipcheappassword; >)
    But this doesn't seem to work. But when I configure the line for the VOIPCheap provider, then I am able to make calls through that account.
    Can someone help me configure my PAP2 adapter please?
    Thank you very much in advance.
    Sekhar.

    I shall have found this post a bit earlier... sik! 
    I have 2 PAP2T 'listenning' to 3 different providers (1 is common); that works... but to be optimal, both ATAs should allow using a 4th provider for outgoing calls.
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    VoIP_Addict, I guess that if you did find some time to 'play' with this, you can only confirm what has been said in this post, right?

  • Newbie - Setting up the SPA3102

    Hi,
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    I bought a SPA3102 and it was delivered yesterday, I plugged the DECT phone base station into the telephone port and connected the line to the wall jack.
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    There is no reason to connect both internet and ethernet ports, unless I missed something setting up mine.
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    Michael
    Message Edited by Swiftnets on 09-03-2007 01:45 PM

  • Spa-3102 Feature Question

    Hello
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    I didn't read your questions carefully enough the first time.
    The EXT IP: address is the external ip address of the SPA3102 itself, not the opposite one.  

  • Hi,, Sorry because maybe this is a dumb question... I wan...

    Hi,, Sorry because maybe this is a dumb question...
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    I want to do this using PAP2 and i need to know what am I need to know in order to programm the equipment,,,
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  • Quick question from a newbie

    I want to set up VOIP trunk between two fixed location. Location A is home while location B is work. There is a VPN between the two, they are on different subnets. I want to access the PSTN line at site B (work) from Site A (home) using VOIP. In other words I want to use the PSTN at work to make voice calls when I work from home.
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    Solved!
    Go to Solution.

    I found teh answer in the excellent "SPA3000 bac to back document". It seems this question has been asked many times apologies, for those new to VOIP it as much knowing what question to ask and what to actually search for!
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  • SPA3102 + Telfort NL

    My ADSL provider recently changed their voip system from ritstele.com to telefoniedienst.nl.
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    Session expires: 1800 seconds
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    My best guess (I could be wrong) would be:
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