My own email address as a SIP address?

Discussion in 'UK VOIP' started by Guest, Jan 5, 2006.

  1. Guest

    Jono Guest

    alexd wrote:
    || Jono wrote:
    ||
    ||| I'd got as far as that, however I can't get my head round dealing
    ||| with the incoming call on asterisk - defining the inbound route &
    ||| how to make whichever extension or ring group actually ring.
    ||
    || Isn't it just as simple as creating an extension called 'jono'? You
    || can then tell Asterisk what to do with all calls to 'jono', much as
    || you would if it was a regular SIP trunk eg from my extensions.conf
    ||
    || [daytime_sipgate]
    || exten => 6402044,1,NoOp(--- ${CALLERID} calling on sipgate
    || (${EXTEN}) ---) exten => 6402044,2,LookupCIDName()
    || ; looks up CLID in AstDB/cidname/...
    || exten => 6402044,3,Dial(SIP/6010&SIP/6012&SIP/6011,45)
    || ; extensions 6010 6011 and 6012 ring on incoming calls from sipgate.
    || ; Goes to voicemail after 45 seconds
    || etc etc
    ||
    ||
    || ie extension 'jono' doesn't have to correspond to a physical phone,
    || it can just be pointed to the extensions of your choice.
    ||


    I'll give it a go, thanks.
     
    Jono, Jan 7, 2006
    #21
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  2. Guest

    Jodokus Guest

    Yes it can! I know, because I just got it working. :) This topic gave
    me some inspiration.
    I happen to have a fixed IP on my ADSL connection and a Asterisk PBX
    for home/small office use.

    You can use sip:<any number>@<your ip or (sub-)domain>. You can route
    <any number> in your dialplan just as you route the number you have
    got from your sip provider. I use [email protected] and incoming calls use the
    context [from-sip-external]. <any number> can also be your first name
    or whatever you like.

    One warning: make sure that there is no way someone can abuse your
    Asterisk by calling out via a paid subscription. Keep your incoming
    and outgoing traffic separated!

    The result is that anyone can download a softphone and reach you for
    free! (That makes a Dutchman happy!)

    Kind regards,
    Jodokus
     
    Jodokus, Jan 7, 2006
    #22
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  3. Guest

    Tim Bray Guest

    I don't think that will work.

    If the SPA-3000 is working as it should, then it should reject any SIP
    packets that arrive from places other than where you registered. Its a
    security thing.

    Making the Asterisk register to Sipgate, and then the SPA-3000
    registered to the Asterisk would make more sense to me.


    Tim
     
    Tim Bray, Jan 10, 2006
    #23
  4. That would render options it has like answering calls when not
    registered as pretty pointless.

    Anyway, I just checked on one of mine, and it happily accepts a
    connection made directly to it, rather than via the SIP registry.
    It might be that if you disable "Ans Call Without Reg", it would
    also refuse to answer a call which didn't come via the registry,
    but I didn't try that.

    I think it would in any case be valid for the SIP server to return
    a redirection response to a client, telling them to contact the
    peer directly (although this doesn't appear to be the norm, and
    it would not work for NAT'ed peers).
    Yes, if you have an Asterisk system.
     
    Andrew Gabriel, Jan 10, 2006
    #24
  5. Indeed this is one of the main uses of my SPA3K, which acts as a bridge
    between a PABX of a charity I am a trustee for and my SOHO phone system.

    I _could_ make all the calls via a SIP provider, but that would simply add
    another potential source of faults into the system - much simpler to connect
    direct.
    I think not, because this is set to "NO" no my box [which reduces the impact
    of my "potential faults" comment, though I am only reliant on the
    registration working not the full call routing and underlying bandwidth
    issues to registry - anyway the 3K registers to my Asterisk Box
    upstairs...].
     
    Thomas Sandford, Jan 10, 2006
    #25
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