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VOIP, Discussion, questions and answers

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  • redpin_2
    redpin_2 Posts: 159 Forumite
    edited 3 May 2009 at 7:42PM
    hi all, im looking for something that i can use that doesnt require a USB connection so i dont have to have my pc on all the time to make and recieve voip calls. is there any devices that one can recommend

    i assume this would do the trick; http://www.probsolvexpress.com/products.asp?partno=SPA1001 but is there anything similar / newer / better available now
    Savings Target: 100K by 2015

    Current Savings: £81,429,04 (Since starting my job as a postman - October 2008)
  • tonyhamm
    tonyhamm Posts: 221 Forumite
    :eek:Well that lasted all of 3 weeks!

    Apparently a common problem with these boxes. I notice Zoom Inc's share price is a tenth of what it was last year I suspect this has something to do with it.

    (PSU dead, but when I stuck it on my bench - being an electronic engineer, opened the box up, and connected a 12V regulated PSU to the PCB, the box powered on, but a component burst into flames a few seconds later).

    Its a real shame as the box itself was great when working, just the crappy build quality lets it down.:confused:

    Getting a refund and a LinkSys PAP2 instead.
    so says another ordinary mug fighting the 1% who own the political machine grinding them down from on high...
    :A
  • full-stop
    full-stop Posts: 101 Forumite
    redpin said:
    i assume this would do the trick; [link removed] but is there anything similar / newer / better available now
    The SPA1001 and PAP2 are still cutting edge ATA's and I am not sure you will find better at the price. Not that easy to set up, so may not be for the faint hearted.

    My preference would be the PAP2 simply because it can handle simultaneously 2 different voip providers.

    Good Luck
  • Adhoc_2
    Adhoc_2 Posts: 9 Forumite
    AVM Fritz.box are a good box of tricks - they can register 10 different providers for incoming and outgoing. I've got the 7050 an earlier model (with WLAN). It provides 3 anaolgue phone ports and an ISDN port (1 of each port can have a pbx connected, although the box comes with pbx capabilities). The ISDN is not dependent on having ISDN as the call can be routed over one of the 10 VOIP providers or the fixed line. There are later models with various changes in set up and improvements, but the 7050 provides an easy user interface and a bundle of features.
  • tonyhamm
    tonyhamm Posts: 221 Forumite
    Well got this Linksys PAP2T and I must say I am NOT impressed.

    It seems you can only setup a single voip provider service account for each line.
    So if you use 2 service providers like I do your faced with having to buy two phones!

    On the Zoom 8251 box I could have both outgoing and incoming different voip services setup on the same line, any incoming call could ring both lines etc..
    All was so much better, except for the 3 week lifespan before it blew up.
    so says another ordinary mug fighting the 1% who own the political machine grinding them down from on high...
    :A
  • BexTech
    BexTech Posts: 4,772 Forumite
    tonyhamm wrote: »
    Well got this Linksys PAP2T and I must say I am NOT impressed.

    It seems you can only setup a single voip provider service account for each line.
    So if you use 2 service providers like I do your faced with having to buy two phones!

    On the Zoom 8251 box I could have both outgoing and incoming different voip services setup on the same line, any incoming call could ring both lines etc..
    All was so much better, except for the 3 week lifespan before it blew up.

    The quality and configuration of the PAP2 is excellent, however it is a base model in the range. If you want more than one provider on one line then you can do this via MySIPSwitch, I have several accounts on line one via MySIPSwitch and Line Two is now set-up for the kids.
    It's PAC not PAC Code, it's MAC not MAC Code, it's PIN not PIN Number, it's ATM not ATM Machine, it's LCD not LCD Display, it's DVD not DVD disc... It's no one not noone, It's a lot not alot, It's got not gotten... Panini is the plural of panino - there is no S!!
    (OK my English isn't great, the sciences, maths & IT are my strong points!)
  • Graham1
    Graham1 Posts: 445 Forumite
    If anybody has been frustrated that they couldn't give out a SIP address for their sipgate.co.uk account for direct IP dialling from users on other VOIP providers I have found that a SIP address like:

    44XXXXXXXXXX@sipgate.net where XXX etc. is the rest of your sipgate.co.uk number including the area code

    does work for incoming calls where

    44XXXXXXXXXX@sipgate.co.uk etc. does not

    This is probably a bit esoteric for most users. I haven't seen it posted elsewhere, nor is it in their help centre so it might save some users some time. Of course there is no guarantee that this will always work.
  • tonyhamm
    tonyhamm Posts: 221 Forumite
    edited 21 May 2009 at 2:23PM
    Thanks Bextech!

    Was quite involved to setup, but now using mySIPswitch to hold my justvoip account for outgoing calls and using another provider for incoming calls.

    Had a LOT of trouble getting justvoip working on mySIPswitch, it didnt work at all - until I registered for a basic account with voxalot.com, and then setup it with my justvoip account, and used mySIPswitch, to use voxalot for outgoing calls. No caller ID showing, but at least can call!

    A great side benefit of mySIPswitch is the ability to write script to replicate a normal phone, so that 7 digit local numbers can be called just as you would with a normal phone, instead of dialling country code+local code+7 digit number etc..
    BexTech wrote: »
    The quality and configuration of the PAP2 is excellent, however it is a base model in the range. If you want more than one provider on one line then you can do this via MySIPSwitch, I have several accounts on line one via MySIPSwitch and Line Two is now set-up for the kids.
    so says another ordinary mug fighting the 1% who own the political machine grinding them down from on high...
    :A
  • pin
    pin Posts: 4,265 Forumite
    Part of the Furniture Combo Breaker
    tonyhamm wrote: »
    Thanks Bextech!

    Was quite involved to setup, but now using mySIPswitch to hold my justvoip account for outgoing calls and using another provider for incoming calls.

    Had a LOT of trouble getting justvoip working on mySIPswitch, it didnt work at all - until I registered for a basic account with voxalot.com, and then setup it with my justvoip account, and used mySIPswitch, to use voxalot for outgoing calls. No caller ID showing, but at least can call!

    A great side benefit of mySIPswitch is the ability to write script to replicate a normal phone, so that 7 digit local numbers can be called just as you would with a normal phone, instead of dialling country code+local code+7 digit number etc..


    Hmm, interesting. I was looking at mySIPswitch as well, and was struggling to do exactly as you have done.

    I want to use SIP Gate for incoming calls and one of the Betamax providers for outgoing calls.

    So from what I understand you need to do is:

    - Set Sip Gate up with mySIPswitch
    - Set Betamax up with Voxalot
    - Set Voxalot up with mySIPwitch

    Once I have done all of that, I should then have only one SIP account details to plug into Fring / Nimbuzz to work of my iphone!

    Am I missing anything?
    "An eye for an eye leaves the whole world blind" - Mahatma Gandhi
  • tonyhamm
    tonyhamm Posts: 221 Forumite
    edited 22 May 2009 at 8:29AM
    Setup a basic account with voxalot, make sure your preferred cluster is eu.voxalot.com, goto voip service providers - add. Setup your betamax details - no need to enter register details, if using for outgoing calls only.

    With mySIPswtich, setup your voxalot details as a provider. As 'voxalot'. Look on the status page to see you have connected ok.

    Setup your Sipgate details.

    Enter the following as a dial plan...Change the 0044151 bit to whatever your local area code is!
    #Ruby
    # Welcome to mysipswitch, this is your default Ruby dialplan.
    # The default rules for incoming calls to yourusername@sip.mysipswitch.com and for your
    # outgoing calls to different providers have been configured for you. To use the outgoing rules you need
    # add SIP Providers with names of providerx where x is 1 to 3. You can then use a prefix of *x for
    # calls to providerx. For example to place a call to 0012345678 at provider1 you would dial *10012345678.
    
    sys.Log("call from #{req.Header.From.FromURI.ToString()} to #{req.URI.User}.")
    
    if sys.In then
      # Do your incoming call processing customisations here.
      sys.Dial("#{sys.Username}@local")                  # Forwards all incoming calls to your registered contacts.
    else
      # Do your outgoing call processing customisations here.
    #********************************************************dial 0044151 if dialled outgoing number is 7 numbers
     [B]case req.URI.User.Length
     when 7 then
          sys.Dial("7 char : local")
          sys.Dial("0044151${dst}@voxalot")
     else
           sys.Log("number length : #{req.URI.User.Length}")
     end [/B]
      case req.URI.User
        when /^303$/ then sys.Dial("303@sip.blueface.ie")  # Blueface speaking clock.
        when /^612$/ then sys.Dial("612@fwd.pulver.com")   # FWD speaking clock.
        when /^\*1/ then sys.Dial("${dst:2}@provider1")   # Forward outgoing calls with a *1 prefix to provider1.
        when /^\*2/ then sys.Dial("${dst:2}@provider2")   # Forward outgoing calls with a *2 prefix to provider2.
        when /^\*3/ then sys.Dial("${dst:2}@provider3")   # Forward outgoing calls with a *3 prefix to provider3.
       [B] else sys.Dial("voxalot")[/B]
      end
    
    end
    
    This sets you up with normal phone dialling rules for 7 digit local numbers, so my old mum can use it like a normal phone, without doing acrobatics with country codes etc..:beer:

    For some reason Justvoip wont work via mySIPswitch directly. Its perhaps something to do with the MMS(multimedia message) interface mySIPswitch uses. Who knows! (Still havent got caler ID working via this method yet, but great to get calls working at last.)

    Glad to save people hours of 'fun' getting through the 'glitches'...!!!
    so says another ordinary mug fighting the 1% who own the political machine grinding them down from on high...
    :A
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