Asterisk

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

Thanks!

Reply to
Jonathan Roberts
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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)

Reply to
Pepperoni

I guess it's only possible with Broadvoice (unfortunately not Vonage) . Hopefully this link will help you to set it up:

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Reply to
Glitch

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

Forget Vonage. It's a closed system.

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'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

Reply to
Kyler Laird

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

Reply to
Wolfgang S. Rupprecht

It doesn't seem odd to me.

Looks like this is what you're thinking.

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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 " snipped-for-privacy@livevoip.com".

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

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

--kyler

Reply to
Kyler Laird

writes:

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.

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.

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.

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

Reply to
Wolfgang S. Rupprecht

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

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

Reply to
Kyler Laird

writes:

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: snipped-for-privacy@yoursite.dom -- 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.)

Reply to
Karl A. Krueger

So do I.

They could. It shouldn't matter.

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

--kyler

Reply to
Kyler Laird

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