PSTN to VOIP

Discussion in 'UK VOIP' started by R Johnson, Sep 25, 2009.

  1. R Johnson

    R Johnson Guest

    A long shot, but other than going for a fully blown solution like
    Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    to IP solution (other than diverting at my cost)?

    I think I've answered my own question, but it's worth asking.
    R Johnson, Sep 25, 2009
    #1
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  2. R Johnson

    Chris Davies Guest

    R Johnson <> wrote:
    > A long shot, but other than going for a fully blown solution like
    > Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    > to IP solution (other than diverting at my cost)?


    It depends entirely on what you expect from this "solution". I use
    Sipgate, which provides PSTN to VoIP (and v.v.). I also use SMSListo
    (betamax), which provides VoIP to PSTN. Depending on your definition of
    VoIP you might want to consider Skype. There's also pbxes.org/com to
    take a look at, if you need somewhat more complex routing.

    Chris
    Chris Davies, Sep 25, 2009
    #2
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  3. R Johnson

    Graham. Guest

    "R Johnson" <> wrote in message
    news:4abd1fb9$0$2479$...
    >A long shot, but other than going for a fully blown solution like
    > Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    > to IP solution (other than diverting at my cost)?
    >
    > I think I've answered my own question, but it's worth asking.


    If it's just for one PSTN line I think some of the AVI Fritz!boxen
    have both FXO and FXS ports.
    No personal experience though

    --
    Graham.

    %Profound_observation%
    Graham., Sep 25, 2009
    #3
  4. R Johnson

    Graham. Guest

    "Chris Davies" <> wrote in message
    news:...
    >R Johnson <> wrote:
    >> A long shot, but other than going for a fully blown solution like
    >> Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    >> to IP solution (other than diverting at my cost)?

    >
    > It depends entirely on what you expect from this "solution". I use
    > Sipgate, which provides PSTN to VoIP (and v.v.). I also use SMSListo
    > (betamax), which provides VoIP to PSTN. Depending on your definition of
    > VoIP you might want to consider Skype. There's also pbxes.org/com to
    > take a look at, if you need somewhat more complex routing.
    >
    > Chris

    The way I read the OP, he has a physical POTS line that he wants to
    integrate into an IP setup.

    --
    Graham.

    %Profound_observation%
    Graham., Sep 25, 2009
    #4
  5. R Johnson

    Ivor Jones Guest

    On 25/09/09 21:36, Graham. wrote:
    > "R Johnson"<> wrote in message
    > news:4abd1fb9$0$2479$...
    >> A long shot, but other than going for a fully blown solution like
    >> Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    >> to IP solution (other than diverting at my cost)?
    >>
    >> I think I've answered my own question, but it's worth asking.

    >
    > If it's just for one PSTN line I think some of the AVI Fritz!boxen
    > have both FXO and FXS ports.
    > No personal experience though
    >


    My Fritz 7170 will divert PSTN to VoIP and vice versa. Never needed to
    use it though so no idea how well it works.

    Ivor
    Ivor Jones, Sep 25, 2009
    #5
  6. In article <4abd1fb9$0$2479$>,
    R Johnson <> wrote:
    >A long shot, but other than going for a fully blown solution like
    >Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    >to IP solution (other than diverting at my cost)?


    You may be able to port the number into a VoIP platform, or failling
    that, there are many ITSPs who can register new telephonoe numbers and
    present them via VoIP.

    The down-side of porting is that all services associated with that line
    cease - e.g. ADSL.

    There are also "appliances" which do run Linux and Asterisk which I'd not
    call "another bleeding server", but do what you might need - i.e. take BT
    lines in and present VoIP out. The ones I make are diskless and fanless
    for the smaller end of the market. (Up to 60 extensions)

    Gordon
    Gordon Henderson, Sep 25, 2009
    #6
  7. R Johnson

    R Johnson Guest

    On Fri, 25 Sep 2009 22:49:28 +0000, Gordon Henderson wrote:

    > In article <4abd1fb9$0$2479$>, R Johnson
    > <> wrote:
    >>A long shot, but other than going for a fully blown solution like
    >>Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    >>to IP solution (other than diverting at my cost)?

    >
    > You may be able to port the number into a VoIP platform, or failling
    > that, there are many ITSPs who can register new telephonoe numbers and
    > present them via VoIP.
    >
    > The down-side of porting is that all services associated with that line
    > cease - e.g. ADSL.
    >
    > There are also "appliances" which do run Linux and Asterisk which I'd
    > not call "another bleeding server", but do what you might need - i.e.
    > take BT lines in and present VoIP out. The ones I make are diskless and
    > fanless for the smaller end of the market. (Up to 60 extensions)
    >
    > Gordon


    I think I'm going to go down the road of Asterisk and put a FXO(?) card
    in one of the dual core mail servers I have that is doing very little.
    I'm guessing with a prod and a play I can get this to handle my SIP
    accounts and POTS and let me just have *one* phone on the desk. LOL
    R Johnson, Sep 26, 2009
    #7
  8. In article <4abdc501$0$2528$>,
    R Johnson <> wrote:
    >On Fri, 25 Sep 2009 22:49:28 +0000, Gordon Henderson wrote:
    >
    >> In article <4abd1fb9$0$2479$>, R Johnson
    >> <> wrote:
    >>>A long shot, but other than going for a fully blown solution like
    >>>Asterisk with cards in *another* bleedin' server, is there a simple PSTN
    >>>to IP solution (other than diverting at my cost)?

    >>
    >> You may be able to port the number into a VoIP platform, or failling
    >> that, there are many ITSPs who can register new telephonoe numbers and
    >> present them via VoIP.
    >>
    >> The down-side of porting is that all services associated with that line
    >> cease - e.g. ADSL.
    >>
    >> There are also "appliances" which do run Linux and Asterisk which I'd
    >> not call "another bleeding server", but do what you might need - i.e.
    >> take BT lines in and present VoIP out. The ones I make are diskless and
    >> fanless for the smaller end of the market. (Up to 60 extensions)
    >>
    >> Gordon

    >
    >I think I'm going to go down the road of Asterisk and put a FXO(?) card
    >in one of the dual core mail servers I have that is doing very little.
    >I'm guessing with a prod and a play I can get this to handle my SIP
    >accounts and POTS and let me just have *one* phone on the desk. LOL


    No real reason why it won't work and although I know Linux is more than
    capable of doing many things on one server these days, I've sort of gone
    back to not having all my eggs in the same basket!

    Asterisk itself has very little overhead on a system.

    Digium analogue card:

    http://www.voipon.co.uk/digium-tdm401b-1-fxo-pci-card-p-759.html

    Or an alternative make at less than half the price:

    http://www.voipon.co.uk/openvox-a400p01-1-fxo-p-669.html

    Look for something called OSLEC to use for echo cancellation.

    One phone on your desk is a reality - apart from your mobile that is :)

    Gordon
    Gordon Henderson, Sep 26, 2009
    #8
  9. R Johnson

    R Johnson Guest

    On Sat, 26 Sep 2009 14:43:00 +0000, Gordon Henderson wrote:

    > In article <4abdc501$0$2528$>, R Johnson
    > <> wrote:
    >>On Fri, 25 Sep 2009 22:49:28 +0000, Gordon Henderson wrote:
    >>
    >>> In article <4abd1fb9$0$2479$>, R Johnson
    >>> <> wrote:
    >>>>A long shot, but other than going for a fully blown solution like
    >>>>Asterisk with cards in *another* bleedin' server, is there a simple
    >>>>PSTN to IP solution (other than diverting at my cost)?
    >>>
    >>> You may be able to port the number into a VoIP platform, or failling
    >>> that, there are many ITSPs who can register new telephonoe numbers and
    >>> present them via VoIP.
    >>>
    >>> The down-side of porting is that all services associated with that
    >>> line cease - e.g. ADSL.
    >>>
    >>> There are also "appliances" which do run Linux and Asterisk which I'd
    >>> not call "another bleeding server", but do what you might need - i.e.
    >>> take BT lines in and present VoIP out. The ones I make are diskless
    >>> and fanless for the smaller end of the market. (Up to 60 extensions)
    >>>
    >>> Gordon

    >>
    >>I think I'm going to go down the road of Asterisk and put a FXO(?) card
    >>in one of the dual core mail servers I have that is doing very little.
    >>I'm guessing with a prod and a play I can get this to handle my SIP
    >>accounts and POTS and let me just have *one* phone on the desk. LOL

    >
    > No real reason why it won't work and although I know Linux is more than
    > capable of doing many things on one server these days, I've sort of gone
    > back to not having all my eggs in the same basket!
    >
    > Asterisk itself has very little overhead on a system.
    >
    > Digium analogue card:
    >
    > http://www.voipon.co.uk/digium-tdm401b-1-fxo-pci-card-p-759.html
    >
    > Or an alternative make at less than half the price:
    >
    > http://www.voipon.co.uk/openvox-a400p01-1-fxo-p-669.html
    >
    > Look for something called OSLEC to use for echo cancellation.
    >
    > One phone on your desk is a reality - apart from your mobile that is :)
    >
    > Gordon


    This looks like 'in budget' - any words on it?

    http://cgi.ebay.co.uk/Authentic-X100P-SE-FXO-PCI-for-Digium-Asterisk-VoIP-
    PBX_W0QQitemZ260480831125QQcmdZViewItemQQptZUK_Computing_Networking_SM?
    hash=item3ca5de0e95&_trksid=p3286.c0.m14
    R Johnson, Sep 26, 2009
    #9
  10. In article <4abe39a8$0$2484$>,
    R Johnson <> wrote:
    >On Sat, 26 Sep 2009 14:43:00 +0000, Gordon Henderson wrote:
    >
    >> In article <4abdc501$0$2528$>, R Johnson
    >> <> wrote:
    >>>On Fri, 25 Sep 2009 22:49:28 +0000, Gordon Henderson wrote:
    >>>
    >>>> In article <4abd1fb9$0$2479$>, R Johnson
    >>>> <> wrote:
    >>>>>A long shot, but other than going for a fully blown solution like
    >>>>>Asterisk with cards in *another* bleedin' server, is there a simple
    >>>>>PSTN to IP solution (other than diverting at my cost)?
    >>>>
    >>>> You may be able to port the number into a VoIP platform, or failling
    >>>> that, there are many ITSPs who can register new telephonoe numbers and
    >>>> present them via VoIP.
    >>>>
    >>>> The down-side of porting is that all services associated with that
    >>>> line cease - e.g. ADSL.
    >>>>
    >>>> There are also "appliances" which do run Linux and Asterisk which I'd
    >>>> not call "another bleeding server", but do what you might need - i.e.
    >>>> take BT lines in and present VoIP out. The ones I make are diskless
    >>>> and fanless for the smaller end of the market. (Up to 60 extensions)
    >>>>
    >>>> Gordon
    >>>
    >>>I think I'm going to go down the road of Asterisk and put a FXO(?) card
    >>>in one of the dual core mail servers I have that is doing very little.
    >>>I'm guessing with a prod and a play I can get this to handle my SIP
    >>>accounts and POTS and let me just have *one* phone on the desk. LOL

    >>
    >> No real reason why it won't work and although I know Linux is more than
    >> capable of doing many things on one server these days, I've sort of gone
    >> back to not having all my eggs in the same basket!
    >>
    >> Asterisk itself has very little overhead on a system.
    >>
    >> Digium analogue card:
    >>
    >> http://www.voipon.co.uk/digium-tdm401b-1-fxo-pci-card-p-759.html
    >>
    >> Or an alternative make at less than half the price:
    >>
    >> http://www.voipon.co.uk/openvox-a400p01-1-fxo-p-669.html
    >>
    >> Look for something called OSLEC to use for echo cancellation.
    >>
    >> One phone on your desk is a reality - apart from your mobile that is :)
    >>
    >> Gordon

    >
    >This looks like 'in budget' - any words on it?
    >
    >http://cgi.ebay.co.uk/Authentic-X100P-SE-FXO-PCI-for-Digium-Asterisk-VoIP-
    >PBX_W0QQitemZ260480831125QQcmdZViewItemQQptZUK_Computing_Networking_SM?
    >hash=item3ca5de0e95&_trksid=p3286.c0.m14


    The x100p cards are indeed budget and I've heard that when used with
    OSLEC can be quite effective, but I've no first-hand expeirence of them.
    There are in all probability people using these with asterisk in their
    thousands though, so I imagine they're OK.

    Gordon
    Gordon Henderson, Sep 26, 2009
    #10
  11. R Johnson

    Ivor Jones Guest

    On 27/09/09 13:25, Dave wrote:
    >
    >
    > "Ivor Jones" <> wrote in message
    > news:...


    [snip]

    >> My Fritz 7170 will divert PSTN to VoIP and vice versa. Never needed to
    >> use it though so no idea how well it works.
    >>
    >> Ivor
    >>

    > Is that box 'better' than a Draytek 2600VG Ivor ?


    As I don't know the Draytek I can't really say, but it works well. It
    has 3 FXS ports, 1 FXO port, 4 LAN ports and an ISDN port, as well as a
    USB host port (only 1.1 though unfortunately) plus wireless, so it's
    quite well equipped.

    I wrote a review of it here, if you're interested:


    http://www.voipon.co.uk/avm-fritzbox-fon-wlan-7170-pri-560.html?reviews_id=10


    Ivor
    Ivor Jones, Sep 27, 2009
    #11
  12. R Johnson

    R Johnson Guest

    On Sun, 27 Sep 2009 17:33:15 +0100, Ivor Jones wrote:

    > On 27/09/09 13:25, Dave wrote:
    >>
    >>
    >> "Ivor Jones" <> wrote in message
    >> news:...

    >
    > [snip]
    >
    >>> My Fritz 7170 will divert PSTN to VoIP and vice versa. Never needed to
    >>> use it though so no idea how well it works.
    >>>
    >>> Ivor
    >>>

    >> Is that box 'better' than a Draytek 2600VG Ivor ?

    >
    > As I don't know the Draytek I can't really say, but it works well. It
    > has 3 FXS ports, 1 FXO port, 4 LAN ports and an ISDN port, as well as a
    > USB host port (only 1.1 though unfortunately) plus wireless, so it's
    > quite well equipped.
    >
    > I wrote a review of it here, if you're interested:
    >
    >
    > http://www.voipon.co.uk/avm-fritzbox-fon-wlan-7170-pri-560.html?

    reviews_id=10
    >
    >
    > Ivor


    I have found the firmware on the Draytek 2600v to be flaky as far as VoIP
    is concerned (and in other places). It's at the point where it works, but
    I don't trust it. Specifically, despite entering the details for SIP
    accounts over and over, they refused to register. I checked and checked
    settings, in the end I rebooted the router and they registered. I'm sure
    the thing has a will of it's own. Randomly I'll get an email to tell me I
    have voicemail, which is pretty good considering the phone has never even
    rung. Never had this kind of agro with my PAP2, but alas I sold that like
    a fool :-(
    R Johnson, Sep 28, 2009
    #12
  13. R Johnson

    Steve Hayes Guest

    Jono wrote:

    > R Johnson explained :
    >> A long shot, but other than going for a fully blown solution like
    >> Asterisk with cards in *another* bleedin' server, is there a simple

    PSTN
    >> to IP solution (other than diverting at my cost)?
    >>
    >> I think I've answered my own question, but it's worth asking.

    >
    > Fritz!Box
    > Linksys/Sipura SPA3102



    I use an SPA3102 for this. It does work but the echoes on forwarded
    calls are truly diabolical - even worse than the totally unnecessary
    echoes on FXO-FXS calls. I tried all the setting tweaks I could find
    (including after googling) and none of them made much difference.

    I'd be interested to know if the Fritzen suffer the same way and, even
    more, if the Draytek 2820 can do this.
    --
    Steve Hayes, South Wales, UK
    ----Remove colours from reply address----
    Steve Hayes, Sep 28, 2009
    #13
  14. R Johnson

    David Knell Guest

    Sign up at http://www.ukddi.com, and port you number over
    for a tenner with no ongoing per-month, per-minute or per-
    channel costs.

    Disclaimer: I set this up.

    --Dave
    David Knell, Sep 29, 2009
    #14
  15. R Johnson

    R Johnson Guest

    On Tue, 29 Sep 2009 09:56:24 -0700, David Knell wrote:

    > Sign up at http://www.ukddi.com, and port you number over for a tenner
    > with no ongoing per-month, per-minute or per- channel costs.
    >
    > Disclaimer: I set this up.
    >
    > --Dave


    Err no. Not until hell freezes over. I'll retain control of it thanks.
    R Johnson, Sep 29, 2009
    #15
  16. In article <4ac04285$0$2493$>,
    R Johnson <> wrote:

    >I have found the firmware on the Draytek 2600v to be flaky as far as VoIP
    >is concerned (and in other places). It's at the point where it works, but
    >I don't trust it. Specifically, despite entering the details for SIP
    >accounts over and over, they refused to register. I checked and checked
    >settings, in the end I rebooted the router and they registered. I'm sure
    >the thing has a will of it's own. Randomly I'll get an email to tell me I
    >have voicemail, which is pretty good considering the phone has never even
    >rung. Never had this kind of agro with my PAP2, but alas I sold that like
    >a fool :-(


    Never seen that issue with the 2600v's I've used - have seen other issues
    - NAT in particular though. Do you have the latest firmware in it?

    Gordon
    Gordon Henderson, Oct 3, 2009
    #16
  17. In article <>, Dave <@me.com> wrote:

    >My problem is that I have a Asterisk box and can't use the built-in ATA's as
    >they 'take over' port 5060 inbound so nothing can login from outside, set
    >the port to '0' and everything is ok apart from the ATA's obviously don't
    >work !. Ivor can you use the ATA's on the Fritz if you have a Asterisk box
    >that is ?


    I've seen the same issue with a Fritz box, but I didn't have a chance
    to fully experiment (on a customers site at the time - I just used IAX
    to get their trunk working rather than fiddle with their Fritz box)

    I don't thnink Drayteks will SIP register on the internal LAN either,
    so even trying to use their ATAs on a different port might not work -
    might have a change to try that this weekend though - my own setup sits
    in the DMZ and the Draytek 2600v seems to pass the wife test...

    Gordon
    Gordon Henderson, Oct 3, 2009
    #17
  18. R Johnson

    R Johnson Guest

    On Sat, 03 Oct 2009 18:54:13 +0000, Gordon Henderson wrote:

    > In article <4ac04285$0$2493$>, R Johnson
    > <> wrote:
    >
    >>I have found the firmware on the Draytek 2600v to be flaky as far as
    >>VoIP is concerned (and in other places). It's at the point where it
    >>works, but I don't trust it. Specifically, despite entering the details
    >>for SIP accounts over and over, they refused to register. I checked and
    >>checked settings, in the end I rebooted the router and they registered.
    >>I'm sure the thing has a will of it's own. Randomly I'll get an email to
    >>tell me I have voicemail, which is pretty good considering the phone has
    >>never even rung. Never had this kind of agro with my PAP2, but alas I
    >>sold that like a fool :-(

    >
    > Never seen that issue with the 2600v's I've used - have seen other
    > issues - NAT in particular though. Do you have the latest firmware in
    > it?
    >
    > Gordon


    The thing is just generally slow, the web interface looks like something
    a small child did with some crayons - and it's all a bit flaky.

    I suspect the firmware is 'old' but updating it is a bit of nightmare.

    Model Name : Vigor2800 series
    Firmware Version : 2.8.2
    Build Date/Time : Tue Jun 3 15:26:55.35 2008
    ADSL Firmware Version : E.38.2.23 Annex A
    R Johnson, Oct 3, 2009
    #18
  19. R Johnson

    Bodincus Guest

    Dave:
    >
    >
    > "R Johnson" <> wrote in message
    > news:4ac04285$0$2493$...
    >> On Sun, 27 Sep 2009 17:33:15 +0100, Ivor Jones wrote:
    >>
    >>> On 27/09/09 13:25, Dave wrote:
    >>>>
    >>>>
    >>>> "Ivor Jones" <> wrote in message
    >>>> news:...
    >>>
    >>> [snip]
    >>>
    >>>>> My Fritz 7170 will divert PSTN to VoIP and vice versa. Never needed to
    >>>>> use it though so no idea how well it works.
    >>>>>
    >>>>> Ivor
    >>>>>
    >>>> Is that box 'better' than a Draytek 2600VG Ivor ?
    >>>
    >>> As I don't know the Draytek I can't really say, but it works well. It
    >>> has 3 FXS ports, 1 FXO port, 4 LAN ports and an ISDN port, as well as a
    >>> USB host port (only 1.1 though unfortunately) plus wireless, so it's
    >>> quite well equipped.
    >>>
    >>> I wrote a review of it here, if you're interested:
    >>>
    >>>
    >>> http://www.voipon.co.uk/avm-fritzbox-fon-wlan-7170-pri-560.html?

    >> reviews_id=10
    >>>
    >>>
    >>> Ivor

    >>
    >> I have found the firmware on the Draytek 2600v to be flaky as far as VoIP
    >> is concerned (and in other places). It's at the point where it works, but
    >> I don't trust it. Specifically, despite entering the details for SIP
    >> accounts over and over, they refused to register. I checked and checked
    >> settings, in the end I rebooted the router and they registered. I'm sure
    >> the thing has a will of it's own. Randomly I'll get an email to tell me I
    >> have voicemail, which is pretty good considering the phone has never even
    >> rung. Never had this kind of agro with my PAP2, but alas I sold that like
    >> a fool :-(

    >
    > My problem is that I have a Asterisk box and can't use the built-in
    > ATA's as they 'take over' port 5060 inbound so nothing can login from
    > outside, set the port to '0' and everything is ok apart from the ATA's
    > obviously don't work !. Ivor can you use the ATA's on the Fritz if you
    > have a Asterisk box that is ?
    > Dave.

    You can change the SIP port the Fritz works with.

    Settings - Telephony - Internet telephony - Advanced Settings.

    Tick "Configuring an alternative "source port" for Internet telephony",
    set it to 5062 and you're OK.

    The RTP ports (the audio stream) on Fritzes are from 7078 to 7110 UDP,
    set your * box to use other UDP ports for RTP and Bob's your uncle.

    A standard * install uses UDP 10000 to 20000 for RTP
    (/etc/asterisk/rtp.conf) so you should be OK with that.

    If the Fritz is your router you don't have anything else to do, if you
    have a separate router remember to forward port 5060 and 10000-20000 UDP
    to your * box, and 5062 and 7078-7110 UDP to your Fritz.

    Please report success / failure.

    --
    Bodincus - The Y2K Druid
    ----------------------------
    Law 42 on computing: Anything that could go wron%½ $
    $: Access Violation - Core dumped
    Bodincus, Oct 3, 2009
    #19
  20. In article <4ac7a230$0$2528$>,
    R Johnson <> wrote:
    >On Sat, 03 Oct 2009 18:54:13 +0000, Gordon Henderson wrote:
    >
    >> In article <4ac04285$0$2493$>, R Johnson
    >> <> wrote:
    >>
    >>>I have found the firmware on the Draytek 2600v to be flaky as far as
    >>>VoIP is concerned (and in other places). It's at the point where it
    >>>works, but I don't trust it. Specifically, despite entering the details
    >>>for SIP accounts over and over, they refused to register. I checked and
    >>>checked settings, in the end I rebooted the router and they registered.
    >>>I'm sure the thing has a will of it's own. Randomly I'll get an email to
    >>>tell me I have voicemail, which is pretty good considering the phone has
    >>>never even rung. Never had this kind of agro with my PAP2, but alas I
    >>>sold that like a fool :-(

    >>
    >> Never seen that issue with the 2600v's I've used - have seen other
    >> issues - NAT in particular though. Do you have the latest firmware in
    >> it?
    >>
    >> Gordon

    >
    >The thing is just generally slow, the web interface looks like something
    >a small child did with some crayons - and it's all a bit flaky.
    >
    >I suspect the firmware is 'old' but updating it is a bit of nightmare.
    >
    >Model Name : Vigor2800 series
    >Firmware Version : 2.8.2
    >Build Date/Time : Tue Jun 3 15:26:55.35 2008
    >ADSL Firmware Version : E.38.2.23 Annex A


    Ah, you said a 2600v yet the model name is 2800 series...

    I've never found their web interface to be slow at all - yes, I'm sure
    it could look better, but I don't care about that - it's functional...

    Almost sounds like there is actually a fault in it...

    Gordon
    Gordon Henderson, Oct 4, 2009
    #20
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