Fritz experience so far ..

Discussion in 'UK VOIP' started by T i m, Jul 2, 2007.

  1. T i m

    T i m Guest

    Hi all,

    After lots of deliberation re a replacement router and wanting to try
    a hard VoIP solution I ended up with a Fritz!Box Fon WLAN 1740.

    Luckily had emailed AVM TS and I was advised how you enable the cable
    (rather than ADSL) use but in spite of this info it still took a
    couple of goes to fully understand how it all worked. (The email
    suggested plugging the WAN cable into 'LanA' but my 1740 doesn't have
    a LanA but that translated to Lan1. You also can't plug the WAN cable
    into the box until you have told it to expect a cable modem on 'Lan1'
    or it simply acts as a hub and the PC 'see's the DHCP server of the
    ISP and gives the wrong IP settings etc. I can see an issue in the
    future when 'Factory resetting' the router and not being able to get
    into it without extracting the WAN cable first?).

    Once that was worked out the rest seems to have gone swimmingly. I
    reset the i/p address of the router, set the DHCP scope to something
    more suitable (to duplicate the settings of previous routers) and
    enabled port forwarding for my NAS / ftp server.

    I have enabled the night service so the WiFi is turned off at 1am and
    back on at 7am and it's pushes a stats email to me (and into a
    FritzStats mail folder) every day at midnight (a bit of which looks
    like this).


    Period Online Data Volume S/R Connections

    Yesterday 24:01 1491MB 80MB / 1411MB 0
    This week 00:04 30MB 2MB / 28MB 0
    This month 24:04 1520MB 81MB / 1439MB 0
    Last month 33:53 421MB 127MB / 294MB 3

    The WiFi seems to work ok (my Palm T|X and a laptop connect to it ok
    anyway) and it accepted and shared (via ftp) my 2G pen drive first
    time (neat that, moreso that you can also share that to the internet)
    ;-)

    I have configured the VoiP to the Sipgate account I setup the other
    day and although the Fitz box says the 'Internet Number is registered
    I haven't actually tested it with a phone yet.

    The firmware update went very well (it found the upgrade file first
    time) and added a few new features like child online time (duration)
    restrictions, energy monitor (power saving) and (I think) the Night
    Service.

    I've run 'Shields Up' ( https://www.grc.com/x/ne.dll?bh0bkyd2 ) and it
    only finds the open ftp port and that it's responding to a ping (and
    can't see a way of turning that off with the tools provided).

    So, bottom line, I like it (but early days etc) It's a bit different
    re the WAN connection but if I do move away from cable to ADSL I have
    already got a suitable router! (I don't think there are many that do
    both ADSL and Cable are there?) The interface isn't quite the same as
    other makes I know (all the main ones) but once you get the idea where
    things are it's all pretty easy. ;-)

    Knowing it's guaranteed for 5 years is also a plus and it was £135
    delivered.

    All the best ..

    T i m
     
    T i m, Jul 2, 2007
    #1
    1. Advertising

  2. T i m

    T i m Guest

    On Mon, 02 Jul 2007 11:27:48 GMT, T i m <> wrote:

    >Hi all,
    >
    >After lots of deliberation re a replacement router and wanting to try
    >a hard VoIP solution I ended up with a Fritz!Box Fon WLAN 1740.


    Not good etiquette to reply to yourself I know but I just spotted an
    important typo ..

    It's a 7140! (doh).

    All the best ..

    T i m
     
    T i m, Jul 2, 2007
    #2
    1. Advertising

  3. T i m

    John Miller Guest

    > After lots of deliberation re a replacement router and wanting to try
    > a hard VoIP solution I ended up with a Fritz!Box Fon WLAN 1740.


    Congratulations :)

    > Luckily had emailed AVM TS and I was advised how you enable the cable
    > (rather than ADSL) use but in spite of this info it still took a
    > couple of goes to fully understand how it all worked.


    The labeling of the LAN ports is indeed a bit different:

    - the FB with 2 LAN ports: LAN A + LAN B
    - the FB with 4 LAN ports: LAN 1 + LAN 2 + LAN 3 + LAN 4

    The first time you switch the ADSL modem off, there is a warning message
    that from now on you need to do the configuration from LAN B instead of LAN
    A, so I think that is quite clear.

    > I have configured the VoiP to the Sipgate account I setup the other
    > day and although the Fitz box says the 'Internet Number is registered
    > I haven't actually tested it with a phone yet.


    You can also use a SIP compatible software phone (SJphone, Xlite, ...) to
    test calls to your new ATA.

    Also, turn ENUM on - it's a great feature, but unfortunately not many people
    use it yet.

    Another remark: make sure you use a good codec for voice calls. I suggest
    G.711 if you have enough bandwith. In the webinterface there is a setting
    "always use fixed-phone quality for VOIP calls". For advanced users: the
    codec priority list can be edited in the voip.cfg file in the Fritz!box.

    I also prefer to turn VAD off (when on, it doesn't send any data during
    silences in a conversation - but I find it strange to hear such absolute
    silences inbetween words.. that's why some devices even introduce "comfort
    noise" :)

    > but if I do move away from cable to ADSL I have
    > already got a suitable router! (I don't think there are many that do
    > both ADSL and Cable are there?)


    I didn't find any other device that could do this yet! Also, almost all
    "subparts" can be disabled: for instance, you can use it solely as a
    wireless access point, or as a telephone exchange, or as wifi repeater, ...
     
    John Miller, Jul 2, 2007
    #3
  4. T i m

    T i m Guest

    On Mon, 2 Jul 2007 14:14:37 +0200, "John Miller"
    <> wrote:

    >> After lots of deliberation re a replacement router and wanting to try
    >> a hard VoIP solution I ended up with a Fritz!Box Fon WLAN 1740.

    >
    >Congratulations :)


    Thanks, like I said, still early days though (but at least it doesn't
    drop me out my online FPS like the new Belkin did).
    >
    >> Luckily had emailed AVM TS and I was advised how you enable the cable
    >> (rather than ADSL) use but in spite of this info it still took a
    >> couple of goes to fully understand how it all worked.

    >
    >The labeling of the LAN ports is indeed a bit different:
    >
    >- the FB with 2 LAN ports: LAN A + LAN B
    >- the FB with 4 LAN ports: LAN 1 + LAN 2 + LAN 3 + LAN 4


    But as I don't think there is anything that refers to any of it
    anywhere in the supplied docs / CD (and the 'CD wizard didn't work for
    me, again possibly because 'Cable' didn't seem to be considered an
    option) so it was a bit of pot luck. AVM have acknowledged this as a
    big oversight.
    >
    >The first time you switch the ADSL modem off, there is a warning message
    >that from now on you need to do the configuration from LAN B instead of LAN
    >A, so I think that is quite clear.


    I didn't see an option to 'Switch the ADSL off' John?, just a radio
    button offering "Internet connection via LAN 1" (Select this kind of
    connection if FRITZ!Box is to be connected to an already existing
    network (LAN), a cable modem or a DSL router).?
    >
    >> I have configured the VoiP to the Sipgate account I setup the other
    >> day and although the Fitz box says the 'Internet Number is registered
    >> I haven't actually tested it with a phone yet.

    >
    >You can also use a SIP compatible software phone (SJphone, Xlite, ...) to
    >test calls to your new ATA.


    Ah, I have X-Lite on there, good idea (still need to plug a phone in
    there).
    >
    >Also, turn ENUM on - it's a great feature, but unfortunately not many people
    >use it yet.


    Because they don't know what it is (like me). The description near the
    option doesn't help much .. "An ENUM request will be performed before
    the number is dialed. The Internet number listed in the domain will
    then be used to dial the number." .. ?

    >Another remark: make sure you use a good codec for voice calls. I suggest
    >G.711 if you have enough bandwith. In the webinterface there is a setting
    >"always use fixed-phone quality for VOIP calls". For advanced users: the
    >codec priority list can be edited in the voip.cfg file in the Fritz!box.


    Ok, thanks.
    >
    >I also prefer to turn VAD off (when on, it doesn't send any data during
    >silences in a conversation - but I find it strange to hear such absolute
    >silences inbetween words.. that's why some devices even introduce "comfort
    >noise" :)


    Ok, understood. "Hello, are you still there ..?"

    >
    >> but if I do move away from cable to ADSL I have
    >> already got a suitable router! (I don't think there are many that do
    >> both ADSL and Cable are there?)

    >
    >I didn't find any other device that could do this yet! Also, almost all
    >"subparts" can be disabled: for instance, you can use it solely as a
    >wireless access point, or as a telephone exchange, or as wifi repeater, ...


    I guess my current focus is on it being a good reliable WiFi router.
    The VoIP was the next important facility and I'm not quite sure how I
    will implement it in here.

    We have a BT line going through a PABX to 5 telephones in different
    rooms of the house.

    We have an NTL / Virgin line which is used by our daughter.

    I would *like* to extend (just) my extension (No2 and with a DECT
    phone) via the Fritz!Box so that I can take and make calls either via
    BT or the SIP services.

    The second VoIP port I would like to dedicate to our daughters (DECT)
    phone. If we loose her NTL line I wouldn't mind if she uses our BT
    line to call out if the BB fails.

    I don't particularly want her phone ringing for anything other than
    incoming VoIP calls to her number.

    But that is all in stage 2!

    All the best and thanks for your help re the Fritz solution John.

    All the best ..

    T i m

    >
     
    T i m, Jul 2, 2007
    #4
  5. T i m

    John Miller Guest

    Re: Fritz experience so far .. / ENUM

    >>Also, turn ENUM on - it's a great feature, but unfortunately not
    >>many people use it yet.

    > Because they don't know what it is (like me). The description near the
    > option doesn't help much .. "An ENUM request will be performed before
    > the number is dialed. The Internet number listed in the domain will
    > then be used to dial the number." .. ?


    If you enable ENUM, each time you dial a number, the Fritz!box will do a DNS
    lookup to see if there is way to dial this number directly over the Internet
    (for free), instead of going over the PSTN network.

    How to enable this on your side?

    - First, make sure you have a SIP provider that allows their customers to be
    reachable via a SIP address. If your current SIP provider doesn't allow
    this, go to voipuser.org and create a free account. You will then have a
    SIP address name @ voipuser.org . Register this new VOIP provider in your
    Fritz!box. You can use X-lite now to call name @ voipuser.org and see if
    your phones are indeed ringing. A SIP address looks like an email address,
    but it's meant to receive (free) calls.

    - Go to www.e164.org and create a new account. Here you make the connection
    that your fixed phonenumer 012345678 can be reached in a free and direct way
    as name @ voipuser.org

    Now, if someone else who has an ENUM compatible ATA is dialing your number
    012345678, that's persons ATA will first check e164.org to see if that
    number has a SIP address equivalent. If so, and yes in this case, it will
    not dial this actual number, but chooses the free and direct way name @
    voipuser.org If there is no equivalent SIP address, it will use the normal
    way to dial this number.

    Hope it is a bit clear this way.

    So if you enable ENUM and you dial a number, it will check if it can call
    that number for free and directly (without any conversion from one PSTN
    provider to another).

    > I don't particularly want her phone ringing for anything other than
    > incoming VoIP calls to her number.


    You can setup for each phone for which incoming numbers it should ring!
     
    John Miller, Jul 2, 2007
    #5
  6. T i m

    Herman Guest

    "John Miller" <> wrote in message
    news:0087a75e$0$21250$...
    > Another remark: make sure you use a good codec for voice calls. I suggest
    > G.711 if you have enough bandwith. In the webinterface there is a setting
    > "always use fixed-phone quality for VOIP calls". For advanced users: the
    > codec priority list can be edited in the voip.cfg file in the Fritz!box.


    Would also be interested to have a look at the configuration files - I guess
    you have to telnet into these? Have you got a few tips for doing this -
    have played around a bit with Telnet on a different box but not sure of the
    exact process again. I have had a few minor issues with the box, and I
    would like to see if this is in the configuration (e.g. three way calling
    not working, inbound dtmf issue, ....)

    I had a quick look at the configuration export file and couldn't find
    anything useful other than the codec selection.
     
    Herman, Jul 2, 2007
    #6
  7. T i m

    John Miller Guest

    > Would also be interested to have a look at the configuration files - I
    > guess you have to telnet into these? Have you got a few tips for doing
    > this - have played around a bit with Telnet on a different box but not
    > sure of the exact process again. I have had a few minor issues with the
    > box, and I would like to see if this is in the configuration (e.g. three
    > way calling not working, inbound dtmf issue, ....)


    First activate telnet from a connected telephone: #96*7* (deactivate with
    #96*8*). You then get a confirmation tone. On an ISDN phone you should see
    the text "telnetd on" or "telnetd off".

    Now, open a telnet session to your Fritz!box and use standard linux
    commands. For instance:

    cd var
    cd flash
    ls (to get the folder contents)
    nvi voip.cfg (to edit the voip.cfg file)

    and so on... but be careful what you change! :)
     
    John Miller, Jul 2, 2007
    #7
  8. T i m

    Herman Guest

    "John Miller" <> wrote in message
    news:00de17fb$0$15393$...
    >> Would also be interested to have a look at the configuration files - I
    >> guess you have to telnet into these? Have you got a few tips for doing
    >> this - have played around a bit with Telnet on a different box but not
    >> sure of the exact process again. I have had a few minor issues with the
    >> box, and I would like to see if this is in the configuration (e.g. three
    >> way calling not working, inbound dtmf issue, ....)

    >
    > First activate telnet from a connected telephone: #96*7* (deactivate with
    > #96*8*). You then get a confirmation tone. On an ISDN phone you should
    > see the text "telnetd on" or "telnetd off".
    >
    > Now, open a telnet session to your Fritz!box and use standard linux
    > commands. For instance:
    >
    > cd var
    > cd flash
    > ls (to get the folder contents)
    > nvi voip.cfg (to edit the voip.cfg file)
    >
    > and so on... but be careful what you change! :)


    Cheers!
     
    Herman, Jul 2, 2007
    #8
  9. T i m

    T i m Guest

    Re: Fritz experience so far .. / ENUM

    On Mon, 2 Jul 2007 16:00:04 +0200, "John Miller"
    <> wrote:

    >>>Also, turn ENUM on - it's a great feature, but unfortunately not
    >>>many people use it yet.

    >> Because they don't know what it is (like me). The description near the
    >> option doesn't help much .. "An ENUM request will be performed before
    >> the number is dialed. The Internet number listed in the domain will
    >> then be used to dial the number." .. ?

    >
    >If you enable ENUM, each time you dial a number, the Fritz!box will do a DNS
    >lookup to see if there is way to dial this number directly over the Internet
    >(for free), instead of going over the PSTN network.
    >
    >How to enable this on your side?
    >
    >- First, make sure you have a SIP provider that allows their customers to be
    >reachable via a SIP address.


    Like Sipgate?

    > If your current SIP provider doesn't allow
    >this, go to voipuser.org and create a free account. You will then have a
    >SIP address name @ voipuser.org . Register this new VOIP provider in your
    >Fritz!box. You can use X-lite now to call name @ voipuser.org and see if
    >your phones are indeed ringing. A SIP address looks like an email address,
    >but it's meant to receive (free) calls.


    Ok ..(I wouldn't have to do that with a Sipgate account would I John?)
    >
    >- Go to www.e164.org and create a new account. Here you make the connection
    >that your fixed phonenumer 012345678 can be reached in a free and direct way
    >as name @ voipuser.org


    So would I put <fixedphonenum> then <sipgatephonenum> in my case?
    >
    >Now, if someone else who has an ENUM compatible ATA is dialing your number
    >012345678, that's persons ATA will first check e164.org to see if that
    >number has a SIP address equivalent. If so, and yes in this case, it will
    >not dial this actual number, but chooses the free and direct way name @
    >voipuser.org If there is no equivalent SIP address, it will use the normal
    >way to dial this number.
    >
    >Hope it is a bit clear this way.


    I think so (unless you *have* to do the voipuser.org bit)?

    I've had a look at that (and registered an account) but will admit
    it's still not very clear, mainly because (as usual with things IT)
    the use of phrases like "... In this example "Primary Termination" is
    set to terminate the phone number being administered to the
    Destination 1 location". ;-(

    >
    >So if you enable ENUM and you dial a number, it will check if it can call
    >that number for free and directly (without any conversion from one PSTN
    >provider to another).


    Assuming all called parties also have enabled ENUM I assume though
    John?
    >
    >> I don't particularly want her phone ringing for anything other than
    >> incoming VoIP calls to her number.

    >
    >You can setup for each phone for which incoming numbers it should ring!


    Good, I hoped that was the case but wasn't sure (again, many of the
    (telephony especially) options in the Fritz!Box aren't 'intuitive' to
    me).

    Because the chances are my Fritz!Box will be connected to an extension
    of our PABX I don't think I'll bother with the analogue / PABX side of
    the box for the time being (in case the PBX makes the picture more
    complicated).

    I guess I need to put a phone on it and start making some test calls
    and see if I can work my way through what seems like 1000 options!

    All the best ..

    T i m
     
    T i m, Jul 3, 2007
    #9
  10. T i m

    Brian Guest

    Re: Fritz experience so far .. / ENUM

    On 03-07-2007, T i m <> wrote:

    > On Mon, 2 Jul 2007 16:00:04 +0200, "John Miller"
    ><> wrote:
    >
    >> - First, make sure you have a SIP provider that allows their customers to be
    >> reachable via a SIP address.

    >
    > Like Sipgate?


    No. Not like Sipgate.

    >> If your current SIP provider doesn't allow
    >> this, go to voipuser.org and create a free account. You will then have a
    >> SIP address name @ voipuser.org . Register this new VOIP provider in your
    >> Fritz!box. You can use X-lite now to call name @ voipuser.org and see if
    >> your phones are indeed ringing. A SIP address looks like an email address,
    >> but it's meant to receive (free) calls.

    >
    > Ok ..(I wouldn't have to do that with a Sipgate account would I John?)


    Indeed you would.

    >> - Go to www.e164.org and create a new account. Here you make the connection
    >> that your fixed phonenumer 012345678 can be reached in a free and direct way
    >> as name @ voipuser.org

    >
    > So would I put <fixedphonenum> then <sipgatephonenum> in my case?


    You could. But it won't work.

    >> Now, if someone else who has an ENUM compatible ATA is dialing your number
    >> 012345678, that's persons ATA will first check e164.org to see if that
    >> number has a SIP address equivalent. If so, and yes in this case, it will
    >> not dial this actual number, but chooses the free and direct way name @
    >> voipuser.org If there is no equivalent SIP address, it will use the normal
    >> way to dial this number.
    >>
    >> Hope it is a bit clear this way.

    >
    > I think so (unless you *have* to do the voipuser.org bit)?


    You have to have a provider who will accept a SIP address. That is, one
    who will route user@domain. Apparently voipuser does. Sipgate is one
    provider who doesn't.

    --
    Brian
     
    Brian, Jul 3, 2007
    #10
  11. T i m

    T i m Guest

    Re: Fritz experience so far .. / ENUM

    On Tue, 3 Jul 2007 11:05:30 +0000 (UTC), Brian <> wrote:


    >You have to have a provider who will accept a SIP address. That is, one
    >who will route user@domain. Apparently voipuser does. Sipgate is one
    >provider who doesn't.


    So it seems there is some subtly in the terminology here .. like the
    difference between a SIP 'number' (which I understand, and a
    telephony term) and SIP 'address' which has no relevance to me under
    my very limited exposure to SIP 'std' VoIP.

    ie, I've been using Skype for ages but was therefore shielded from all
    the complicated stuff by the fact that it's a closed network etc.

    By the looks of it there isn't a 'simple' way to explain all the
    various functions in less than a full FAQ (and that would have to be
    aimed at a non regular VoIP user to be of any use to me).

    But, I've got incoming VoIP calls ringing the DECT phone that is
    normally *just* plugged into Ext2 of our PABX and have also tested
    that I can still dial out (although that seems quite messy now ...
    (*111#9 etc) is there a way where outgoing calls default to analogue?)
    so I am getting somewhere.

    I suppose it's like many things for many of us, as long we can get
    something to work even if only in it's very basic form then that's
    often good enough ... (video's, mobile phones, computers ... ) but we
    may pick up some extra bits as / when we are shown or have a specific
    need etc?

    For the moment and certainly for the purpose of replacing our second
    telephone line I'm happy to have a basic Sipgate number and pay for
    the odd outgoing PSTN call.

    All the best ..

    T i m
     
    T i m, Jul 3, 2007
    #11
  12. T i m

    John Miller Guest

    Re: Fritz experience so far .. / ENUM

    > So it seems there is some subtly in the terminology here .. like the
    > difference between a SIP 'number' (which I understand, and a
    > telephony term) and SIP 'address' which has no relevance to me under
    > my very limited exposure to SIP 'std' VoIP.


    Your ATA is registered with a VOIP provider by using the SIP protocol.

    Now, there are several ways how people can reach you:

    - via a regular phonenumber (012/3456789), if your VOIP provider has linked
    one to your account
    - via a SIP address (), if your VOIP provider has linked
    one to your account

    If you use that VOIP provider only for outgoing calls (and your fixed line
    for incoming calls), you don't need any of the above!

    > ie, I've been using Skype for ages but was therefore shielded from all
    > the complicated stuff by the fact that it's a closed network etc.
    > By the looks of it there isn't a 'simple' way to explain all the
    > various functions in less than a full FAQ (and that would have to be
    > aimed at a non regular VoIP user to be of any use to me).


    The advantage of SIP is that it is open, and very flexible and configurable.
    So I guess the "disadvantage" can be that it is a bit more complex maybe.

    I was also struggeling in the beginning with this new technology; but once
    you understand the details it is really exciting how everything works and
    can be configured. The funny thing is that I've spend more money on my VOIP
    equipment lately, than I could ever save because of using SIP compared to
    normal telephony :)

    I am certainly open to help writing a FAQ concerning this matter!

    > But, I've got incoming VoIP calls ringing the DECT phone that is
    > normally *just* plugged into Ext2 of our PABX and have also tested
    > that I can still dial out (although that seems quite messy now ...
    > (*111#9 etc) is there a way where outgoing calls default to analogue?)
    > so I am getting somewhere.


    In the properties of Settings/Telephony/Extensions/FON1 (or whatever
    connector you've connected your DECT phone to) you should select "fixed
    line" as the default for all calls. It will then always dial out over the
    fixed line, if you don't have any dial rules setup. There is no need to
    enter any prefixes at all; it should just work like a "normale" phone
    system.
     
    John Miller, Jul 3, 2007
    #12
  13. T i m

    John Miller Guest

    Re: Fritz experience so far .. / ENUM

    >>So if you enable ENUM and you dial a number, it will check if it can call
    >>that number for free and directly (without any conversion from one PSTN
    >>provider to another).


    > Assuming all called parties also have enabled ENUM I assume though
    > John?


    Yes. That is why we should encourage all SIP callers to setup ENUM for
    their system! Voipuser.org is free and works very well.
     
    John Miller, Jul 3, 2007
    #13
  14. T i m

    John Miller Guest

    Re: Fritz experience so far .. / ENUM

    > Yes. That is why we should encourage all SIP callers to setup ENUM for
    > their system! Voipuser.org is free and works very well.


    I meant e164.org instead of voipuser.org.

    But voipuser.org also works very well :)
     
    John Miller, Jul 3, 2007
    #14
  15. T i m

    T i m Guest

    Re: Fritz experience so far .. / ENUM

    On Tue, 3 Jul 2007 15:13:37 +0200, "John Miller"
    <> wrote:

    >> So it seems there is some subtly in the terminology here .. like the
    >> difference between a SIP 'number' (which I understand, and a
    >> telephony term) and SIP 'address' which has no relevance to me under
    >> my very limited exposure to SIP 'std' VoIP.

    >
    >Your ATA is registered with a VOIP provider by using the SIP protocol.
    >
    >Now, there are several ways how people can reach you:
    >
    >- via a regular phonenumber (012/3456789), if your VOIP provider has linked
    >one to your account


    Ok, that one I understand and make sense John ..

    >- via a SIP address (), if your VOIP provider has linked
    >one to your account


    Ok, that one doesn't. I mean. I hear what you say but I'm not sure I
    see the point of doing so?
    >
    >If you use that VOIP provider only for outgoing calls (and your fixed line
    >for incoming calls), you don't need any of the above!


    Understood. ;-)
    >
    >> ie, I've been using Skype for ages but was therefore shielded from all
    >> the complicated stuff by the fact that it's a closed network etc.
    >> By the looks of it there isn't a 'simple' way to explain all the
    >> various functions in less than a full FAQ (and that would have to be
    >> aimed at a non regular VoIP user to be of any use to me).

    >
    >The advantage of SIP is that it is open, and very flexible and configurable.
    >So I guess the "disadvantage" can be that it is a bit more complex maybe.


    Maybe!?! I never had any issues with Skype but then didn't take it any
    further than just PC to PC comms.
    >
    >I was also struggeling in the beginning with this new technology; but once
    >you understand the details it is really exciting how everything works and
    >can be configured.


    Oh I can see that John and why I was (am) interested in setting
    something up, even if only as a test and to have in the background (to
    save money on line rental) or in case I can ever help someone else.

    > The funny thing is that I've spend more money on my VOIP
    >equipment lately, than I could ever save because of using SIP compared to
    >normal telephony :)


    Isn't that often the way! In fact though, 1) I did need a more
    reliable router, 2) If I *can* loose the rental on the second line
    that will save me ~£10/month and 3) if we just end up with just the
    BB on Virgin (no phones) I may well pay that via DD (rather than
    paying on demand over the net) saving a further £5/month.
    >
    >I am certainly open to help writing a FAQ concerning this matter!


    That could be very handy (for me anyway).
    >
    >> But, I've got incoming VoIP calls ringing the DECT phone that is
    >> normally *just* plugged into Ext2 of our PABX and have also tested
    >> that I can still dial out (although that seems quite messy now ...
    >> (*111#9 etc) is there a way where outgoing calls default to analogue?)
    >> so I am getting somewhere.

    >
    >In the properties of Settings/Telephony/Extensions/FON1 (or whatever
    >connector you've connected your DECT phone to) you should select "fixed
    >line" as the default for all calls. It will then always dial out over the
    >fixed line, if you don't have any dial rules setup. There is no need to
    >enter any prefixes at all; it should just work like a "normale" phone
    >system.


    <Tim tries new settings> Nice, thanks (again) John! ;-)

    Now, say I wanted to dial out via the SIP(gate) service ...would that
    be the "*121#" in the Internet Telephony ? Internet Numbers field,
    Sipgate ?

    All the best ..

    T i m.

    p.s. Is there a particular DECT phone that works well with these boxes
    / SIP in general please (I'm thinking something that has a good
    display / flexible phonebook etc)?

    p.p.s. Can you have more than one number per Sipgate account or would
    I have to create a new complete account for my daughters line?
     
    T i m, Jul 3, 2007
    #15
  16. T i m

    T i m Guest

    Re: Fritz experience so far .. / ENUM

    On Tue, 3 Jul 2007 15:16:51 +0200, "John Miller"
    <> wrote:

    >>>So if you enable ENUM and you dial a number, it will check if it can call
    >>>that number for free and directly (without any conversion from one PSTN
    >>>provider to another).

    >
    >> Assuming all called parties also have enabled ENUM I assume though
    >> John?

    >
    >Yes. That is why we should encourage all SIP callers to setup ENUM for
    >their system! Voipuser.org is free and works very well.
    >

    Makes sense.

    Like I said I have registered with them but gave up on the number
    allocation stage as I really didn't know what the implications of all
    the extra boxes were? ;-(

    All the best ..

    T i m
     
    T i m, Jul 3, 2007
    #16
  17. T i m

    T i m Guest

    Re: Fritz experience so far .. / ENUM

    On Tue, 3 Jul 2007 15:19:44 +0200, "John Miller"
    <> wrote:

    >> Yes. That is why we should encourage all SIP callers to setup ENUM for
    >> their system! Voipuser.org is free and works very well.

    >
    >I meant e164.org instead of voipuser.org.
    >
    >But voipuser.org also works very well :)


    Ah, yes, I've actually registered with both but still trying to put
    the right numbers in the right boxes. ;-(

    All the best ..

    T i m
     
    T i m, Jul 3, 2007
    #17
  18. T i m

    Jono Guest

    Re: Fritz experience so far .. / ENUM

    T i m wrote on 03/07/2007 :
    >> - via a SIP address (), if your VOIP provider has linked
    >> one to your account

    >
    > Ok, that one doesn't. I mean. I hear what you say but I'm not sure I
    > see the point of doing so?


    Think of it like an email address.

    If you were with the same ISP as me, in theory, I could email you
    simply with the address "tim" and the mail would reach you.

    However, if we were on different ISP, I would have to email you using
    "tim@yourISP"

    The SIP number is specific to your VoIP provider, your SIP address
    allows calls to reach you from (any) other network. (SIPGATE have
    chosen to block most calls originating off their network) Your PSTN
    number allows calls to reach you from the normal phone network.
     
    Jono, Jul 3, 2007
    #18
  19. T i m

    T i m Guest

    Re: Fritz experience so far .. / ENUM

    On Tue, 03 Jul 2007 19:19:48 +0100, Jono <>
    wrote:

    >T i m wrote on 03/07/2007 :
    >>> - via a SIP address (), if your VOIP provider has linked
    >>> one to your account

    >>
    >> Ok, that one doesn't. I mean. I hear what you say but I'm not sure I
    >> see the point of doing so?

    >
    >Think of it like an email address.


    Ok ..
    >
    >If you were with the same ISP as me, in theory, I could email you
    >simply with the address "tim" and the mail would reach you.


    Ok, understood, like not using the STD code on the land line etc.
    >
    >However, if we were on different ISP, I would have to email you using
    >"tim@yourISP"


    Ok again .. routing outside your own network etc ..
    >
    >The SIP number is specific to your VoIP provider,


    But transparent to anyone calling it.

    > your SIP address
    >allows calls to reach you from (any) other network.


    And I don't have that from Sipgate I assume.

    > (SIPGATE have
    >chosen to block most calls originating off their network)


    And is that not typical of other similar spec (free local number /
    free incoming PSTN / SIUP calls) SIP providers Jono?


    >Your PSTN
    >number allows calls to reach you from the normal phone network.


    Understood.

    Ok, I would like to use what is the most flexible and I'd rather do
    that from the beginning. However I have registered with Sipgate and
    also have 333 free mins with them to use up and at the moment the only
    system simple enough for me to fully comprehend (because they only use
    telephone numbers in a 'telephone environment' (as I see it)).

    So, if I went with 'another' SIP provider I guess I would get two
    'things' from my registration, a STD number for ordinary PSTN users to
    call me on and a number linked 'address' for some (all) SIP clients to
    use?

    So ...

    I've used X-Lite to take a PSTN call via Sipgate (using my sipgate SIP
    'number').

    I've used a DECT phone connected to my Fritz!Box to take a PSTN call
    via Sipgate (using my sipgate SIP 'number').

    I've used Skype to make and receive calls (using the Skype username,
    similar to MSN or X-Talk for in game chat).

    How would someone with a stand alone SIP phone call me over VoIP, I
    mean, what would they enter on their analogue phone if I only had an
    address?

    (sorry to be slow) ;-(

    All the best ..

    T i m
     
    T i m, Jul 3, 2007
    #19
  20. T i m

    Brian Guest

    Re: Fritz experience so far .. / ENUM

    On 03-07-2007, T i m <> wrote:

    > On Tue, 03 Jul 2007 19:19:48 +0100, Jono <>
    > wrote:
    >
    >> (SIPGATE have
    >> chosen to block most calls originating off their network)

    >
    > And is that not typical of other similar spec (free local number /
    > free incoming PSTN / SIUP calls) SIP providers Jono?


    Not paticularily. Voiptalk, Gradwell and Voipfone don't block SIP-to-SIP
    calls. Vonage and Sipgate do. It does not correlate with freeness.

    > So, if I went with 'another' SIP provider I guess I would get two
    > 'things' from my registration, a STD number for ordinary PSTN users to
    > call me on and a number linked 'address' for some (all) SIP clients to
    > use?


    The important item is the SIP address, user@domain. Without it you are
    not contactable. At some point a PSTN number has to point to it. The
    phone call would not be delivered otherwise.

    > How would someone with a stand alone SIP phone call me over VoIP, I
    > mean, what would they enter on their analogue phone if I only had an
    > address?


    This depends on the ATA used and is not straightforward. If your Sipgate
    ID was 1112223 I would need to dial , and the
    letters present a problem with an analogue phone. It can be done if the
    ATA has a quick dial or address book facility. Or, rather tediously,
    dial 111222333*217*10*79*23 from the handset. * replaces @ and . on my
    ATA.

    --
    Brian
     
    Brian, Jul 3, 2007
    #20
    1. Advertising

Want to reply to this thread or ask your own question?

It takes just 2 minutes to sign up (and it's free!). Just click the sign up button to choose a username and then you can ask your own questions on the forum.
Similar Threads
  1. Joel Rubin
    Replies:
    5
    Views:
    510
    PuppyKatt
    Aug 18, 2004
  2. Ivor Jones
    Replies:
    4
    Views:
    560
    Ivor Jones
    Aug 18, 2008
  3. Gordon Henderson
    Replies:
    1
    Views:
    622
    Brian A
    Aug 17, 2008
  4. Tim
    Replies:
    0
    Views:
    467
  5. alexd
    Replies:
    0
    Views:
    456
    alexd
    Sep 16, 2008
Loading...

Share This Page