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Asterisk

 
 
Jonathan Roberts
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      10-27-2004
Can Asterisk utilize multiple VOIP accounts as lines? I am thinking either
Vonage or Broadvoice

Thanks!


 
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Pepperoni
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      10-27-2004
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" <(E-Mail Removed)> 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!
>
>



 
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Glitch
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      10-27-2004
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-Asteri...ngs+Broadvoice
 
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Kyler Laird
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      10-27-2004
"Jonathan Roberts" <(E-Mail Removed)> 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
 
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Wolfgang S. Rupprecht
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      10-27-2004

Kyler Laird <(E-Mail Removed)> writes:
> "Jonathan Roberts" <(E-Mail Removed)> 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/
 
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Kyler Laird
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      10-28-2004
"Wolfgang S. Rupprecht" <(E-Mail Removed) .wsrcc.com> 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 "(E-Mail Removed)".

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
 
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Wolfgang S. Rupprecht
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      10-28-2004

Kyler Laird <(E-Mail Removed)> writes:
> "Wolfgang S. Rupprecht" <(E-Mail Removed) .wsrcc.com> 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 "(E-Mail Removed)".


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/
 
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Kyler Laird
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      10-28-2004
"Wolfgang S. Rupprecht" <(E-Mail Removed) .wsrcc.com> 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 "(E-Mail Removed)".


>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
 
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Karl A. Krueger
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      10-28-2004
Kyler Laird <(E-Mail Removed)> wrote:
> "Wolfgang S. Rupprecht" <(E-Mail Removed) .wsrcc.com> 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:(E-Mail Removed) -- 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 <(E-Mail Removed)> { 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.
 
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Kyler Laird
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      10-29-2004
"Karl A. Krueger" <(E-Mail Removed)> 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:(E-Mail Removed) -- 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
 
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