Asterisk

Discussion in 'VOIP' started by Jonathan Roberts, Oct 27, 2004.

  1. Can Asterisk utilize multiple VOIP accounts as lines? I am thinking either
    Vonage or Broadvoice

    Thanks!
    Jonathan Roberts, Oct 27, 2004
    #1
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  2. Jonathan Roberts

    Pepperoni Guest

    Vonage only lets you use the adapter which they supply.
    They will allow you to use a softphone, but there is a monthly (additional)
    charge involved. Their current softphone is from Xten, I believe.

    It *may* be possible to use a software dialer as a substitute for your
    hardware phone handset through your Vonage hardware. (I haven't tried it)


    "Jonathan Roberts" <> wrote in message
    news:JNFfd.35081$_g6.11457@okepread03...
    > Can Asterisk utilize multiple VOIP accounts as lines? I am thinking

    either
    > Vonage or Broadvoice
    >
    > Thanks!
    >
    >
    Pepperoni, Oct 27, 2004
    #2
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  3. Jonathan Roberts

    Glitch Guest

    On Tue, 26 Oct 2004 23:54:39 -0500, Jonathan Roberts wrote:

    > Can Asterisk utilize multiple VOIP accounts as lines? I am thinking either
    > Vonage or Broadvoice
    >
    > Thanks!


    I guess it's only possible with Broadvoice (unfortunately not Vonage) .
    Hopefully this link will help you to set it up:
    http://www.voip-info.org/wiki-Asterisk settings Broadvoice
    Glitch, Oct 27, 2004
    #3
  4. Jonathan Roberts

    Kyler Laird Guest

    "Jonathan Roberts" <> writes:

    >Can Asterisk utilize multiple VOIP accounts as lines?


    Yes, I am now using BroadVoice, VoicePulse Connect!, Gafachi and
    LiveVoIP. (I'll probably go to just LiveVoIP next year.)

    >I am thinking either
    >Vonage or Broadvoice


    Forget Vonage. It's a closed system.
    http://www.voip-info.org/wiki-Vonage
    There's nothing special about the service they provide anyway.

    BroadVoice is generally o.k. for home use and they do have "unlimited"
    (meaning "we don't tell you what the limit is") plans. VoicePulse is
    better for incoming calls if they happen to serve your area. Gafachi
    and LiveVoIP are better for outgoing calls. LiveVoIP is *the* choice
    for toll-free service.

    --kyler
    Kyler Laird, Oct 27, 2004
    #4
  5. Kyler Laird <> writes:
    > "Jonathan Roberts" <> writes:
    >
    > >Can Asterisk utilize multiple VOIP accounts as lines?

    >
    > Yes, I am now using BroadVoice, VoicePulse Connect!, Gafachi and
    > LiveVoIP. (I'll probably go to just LiveVoIP next year.)


    Thanks for the good info. I'm saving quite a few of your postings for
    reference.

    I'm not quite sure what to make of this though.

    $ dig _sip._udp.livevoip.com any
    _sip._udp.livevoip.com. 86382 IN A 217.160.251.55

    Wasn't there supposed to be an SRV entry with the priority, weight,
    port number and hostname at that location?

    Is this some new standard or are they just another confused telco
    wannabe that hasn't a clue what they are doing?

    -wolfgang
    --
    Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
    Wolfgang S. Rupprecht, Oct 27, 2004
    #5
  6. Jonathan Roberts

    Kyler Laird Guest

    "Wolfgang S. Rupprecht" <> writes:

    >I'm not quite sure what to make of this though.


    > $ dig _sip._udp.livevoip.com any
    > _sip._udp.livevoip.com. 86382 IN A 217.160.251.55


    It doesn't seem odd to me.

    >Wasn't there supposed to be an SRV entry with the priority, weight,
    >port number and hostname at that location?


    Looks like this is what you're thinking.
    http://web.mit.edu/sip/sip.edu/dns.shtml
    Do you know of any provider who *does* do that? I don't see much of
    an advantage of having it. It's certainly not something I'd expect
    of a provider like LiveVoIP. They're not likely to have customers
    who want to advertise availability at "".

    I'm happy to use ENUM and DUNDi. (I have a LiveVoIP DUNDi agreement
    waiting on my signature.)

    >Is this some new standard or are they just another confused telco
    >wannabe that hasn't a clue what they are doing?


    They certainly seem to be clueful. I hope to write more on the
    subject soon.

    --kyler
    Kyler Laird, Oct 28, 2004
    #6
  7. Kyler Laird <> writes:
    > "Wolfgang S. Rupprecht" <> writes:
    > Looks like this is what you're thinking.
    > http://web.mit.edu/sip/sip.edu/dns.shtml


    That is exactly what I'm thinking. For one my sipura-3000 will look
    for those SRV records and will in theory hit the sip servers in the
    correct order. In theory asterisk also uses it, but its
    implementation is somewhat flawed and it doesn't do the fallback
    correctly.

    > Do you know of any provider who *does* do that? I don't see much of
    > an advantage of having it.


    Broadvoice does it, but they use the subdomain sip.broadvoice.com.
    Back during my brief trial with them they had 2 servers listed, which
    gave the ATA's an automatic fallback.

    0 0 5060 proxy.dca.broadvoice.com.
    1 0 5060 proxy.lax.broadvoice.com.

    > It's certainly not something I'd expect of a provider like LiveVoIP.
    > They're not likely to have customers who want to advertise
    > availability at "".


    The entries are useful even if you don't want to give out email-like
    telephone numbers. The other day I wanted to call someone at MIT. I
    knew MIT had a SIP gateway so I added the MIT SRV entry into my
    asterisk extensions file and now call there without a PSTN hop. If
    they ever add more SIP gateways or change its name it will be
    transparent to me.

    Similarly if I wanted to call someone that used livevoip, I might try
    to look up livevoip's SRV entry and have asterisk try to route the
    call directly. It was just by chance that I noticed that they had an
    A-record with a numeric IP address where the SRV record was supposed
    to go. It would be nice if companies didn't make standards up on the
    fly and conformed to the rest of the industry. Standards are there
    for a reason and do make things easier for everyone.

    > I'm happy to use ENUM and DUNDi. (I have a LiveVoIP DUNDi agreement
    > waiting on my signature.)


    It'll be interesting to see how both ENUM and DUNDI fair. So far, I'm
    only listed in my DNS.

    _sip._udp.wsrcc.com. 6147 IN SRV 0 0 5060 sonic.wsrcc.com.

    -wolfgang
    --
    Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
    Wolfgang S. Rupprecht, Oct 28, 2004
    #7
  8. Jonathan Roberts

    Kyler Laird Guest

    "Wolfgang S. Rupprecht" <> writes:

    >> It's certainly not something I'd expect of a provider like LiveVoIP.
    >> They're not likely to have customers who want to advertise
    >> availability at "".


    >The entries are useful even if you don't want to give out email-like
    >telephone numbers. The other day I wanted to call someone at MIT. I
    >knew MIT had a SIP gateway so I added the MIT SRV entry into my
    >asterisk extensions file and now call there without a PSTN hop. If
    >they ever add more SIP gateways or change its name it will be
    >transparent to me.


    Again, this has little to do with the PSTN provider(s) MIT uses. How
    do you know they don't use LiveVoIP?

    >Similarly if I wanted to call someone that used livevoip, I might try
    >to look up livevoip's SRV entry and have asterisk try to route the
    >call directly.


    That's like expecting to contact my mobile phone by calling my electric
    company because I use its service to charge my phone.

    I would not expect typical LiveVoIP customers to even tell you that
    they use LiveVoIP. Why would they? We're not talking about a bunch
    of naive users locked into some system like Vonage or Skype. This is
    a provider of commodity service.

    --kyler
    Kyler Laird, Oct 28, 2004
    #8
  9. Kyler Laird <> wrote:
    > "Wolfgang S. Rupprecht" <> writes:
    >>The entries are useful even if you don't want to give out email-like
    >>telephone numbers. The other day I wanted to call someone at MIT. I
    >>knew MIT had a SIP gateway so I added the MIT SRV entry into my
    >>asterisk extensions file and now call there without a PSTN hop. If
    >>they ever add more SIP gateways or change its name it will be
    >>transparent to me.

    >
    > Again, this has little to do with the PSTN provider(s) MIT uses. How
    > do you know they don't use LiveVoIP?


    MIT runs their own SIP gateway; they do not use an outside PSTN provider
    any more than they would use Hotmail for email service.

    I know. They're in my SER dialplan. I work for an institution that
    offers joint degrees with MIT. :)

    (If you want other sites to be able to ring your SIP extensions by
    username and domain -- like sip: -- then you need
    to have an SRV record for _sip._udp.yoursite.dom in your DNS, to point
    to your domain's SIP proxy or endpoint. That's how a remote SIP user
    will discover the address of your SIP Proxy.)

    --
    Karl A. Krueger <> { s/example/whoi/ }

    Every program has at least one bug and can be shortened by at least one line.
    By induction, every program can be reduced to one line which does not work.
    Karl A. Krueger, Oct 29, 2004
    #9
  10. Jonathan Roberts

    Kyler Laird Guest

    "Karl A. Krueger" <> writes:

    >> Again, this has little to do with the PSTN provider(s) MIT uses. How
    >> do you know they don't use LiveVoIP?


    >MIT runs their own SIP gateway;


    So do I.

    >they do not use an outside PSTN provider
    >any more than they would use Hotmail for email service.


    They could. It shouldn't matter.

    >(If you want other sites to be able to ring your SIP extensions by
    >username and domain -- like sip: -- then you need
    >to have an SRV record for _sip._udp.yoursite.dom in your DNS, to point
    >to your domain's SIP proxy or endpoint. That's how a remote SIP user
    >will discover the address of your SIP Proxy.)


    Right. It makes *no* sense to goof around with some service
    provider's name.

    --kyler
    Kyler Laird, Oct 29, 2004
    #10
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