Re: Is Nufone going out of business???

I tried broadvoice for two weeks with their BYOD offering. Shortly after signing up with them they tightened their security and asterisk failed to register reliably.

All I want is a way to tie into the PSTN digitally without paying Pacbell an arm and a leg to get their ISDN.

-wolfgang

Broadvoice had no security in their early days, but now their security is not what I would call "tight" in the sense of making it hard to use. Standard SIP AUTH with digest authentication, Asterisk should be able to handle it easily.

I give Broadvoice excellent marks for being one of the few services to let you use Generic SIP equipment to make and receive calls, and I have found their call quality to be quite good with low latency. I had some initial problems calling Canada but it has been fine in recent tests.

If you get a Sipura from Broadvoice, it will come locked but you can buy your own Sipura or other phone from a store and use it with them, and control it all yourself, or even use custom software as I have been doing, so Asterisk should adapt well.

You're not supposed to judge a book by its cover, that's true. However a business generally will make an effort to put forth a good face if it intends to attracts and keep customers, whether that business is good OR bad. If they can't do that, then you have to question their effort in other departments.

[snip]

Well, I wouldn't judge based on just two companies. Vonage has been good to me, though Packet8 looks like they'll be better, so I'm in the process of switching to them now. we'll see how that goes. :)

[snip]

I agree. It's good business to talk up your service... it's bad business to talk trash about competitors, no matter how justified.

Not only that, but this "Jeremy" decides to school you on phone etiquette? Nice.

Unfortuantely, it seems like this is par for the course when people adopt bleeding edge technology. Lots of companies, a few of which do things right, while a vast majority of startups wanting in doing things very, very wrongly. Eventually, there will be a shakeout. Let's just hope we're not all on the losing end of that.

kyler

Update on the Jeremy McNamara situation... After I sent him a full copy of the message (with headers) his system bounced to me (which he claimed was tagged with a "status change" from their ticket system), he claimed that I sent the message that I received. Uh, yeah. I like to pretend to be "Apache " and give my messages IDs like "".

There's something very wrong at Nufone.

Then again, I'm still timing out trying to use my new VoicePulse account. And my IAXy requires a few power cycles almost every time I use it. Maybe this is just a growth stage for VoIP where we should expect everything to be broken.

--kyler

That's good to know. They were next on my list.

Heck, I'd settle for an analog interface right now but Verizon claims that disconnect supervision doesn't exist.

It's been quite an education to finally dip my foot in the VoIP pool.

--kyler

If you look at my messages it will say From: " snipped-for-privacy@nufone.net via RT"

You have a very strange way of offering your services, that's for sure.

Jeremy McNamara

writes:

Well, it did seem like their heart was in the right place. I wish them well, and I suspect they will do well in the long run.

I have a Sipura-3000 on my pots line. Supervision seems to work just fine in my part of the world (former Pacbell, San Francisco Bay Area in the (510) 659-XXXX exchange). The Sipura can detect battery reversal as well as CPC battery removal. I'm impressed with the info the Sipura does provide. (Notice the "Last PSTN Disconnect Reason".)

PSTN Line Status Hook State: On Loop Current: 0.0 (mA) Last Registration At: 8/19/2004 08:55:51 Last Called VoIP Number: snipped-for-privacy@wsrcc.com Last VoIP Caller: Last PSTN Disconnect Reason: CPC Signal

Line Voltage: 46 (V) Registration State: Registered Next Registration In: 2 s Last Called PSTN Number: 12005551212 Last PSTN Caller: John Doe, 2005551212 PSTN Activity Timer: 30000 (ms)

The part that doesn't work so fine is the incoming volume and delay. It is very hard to hear the other party when the call is routed to a Grandstream Bugtome 100. Bumping the incoming gain on the PSTN side by +12db from the config menu mostly fixes the volume problem, but leads to horrendous echo.

Same here! And it all started out because I needed to replace a failing answering machine and thought "Gee, shouldn't I try to have my computer do this?" I never thought it would be this involved.

-wolfgang

In theory asterisk should have worked just fine. In practice it failed miserably. Asterisk is still very rickety and is missing large sections of code. One thing it is missing is any logic to loop over multiple DNS A-records or multiple SRV _sip._upd.example.com. It simply chooses the first one and sticks with it until restarted or reloaded. Since bind9 will round-robbin I saw asterisk more or less toggle between proxy.lax.broadvoice.com and proxy.dca.broadvoice.com (names from memory). The LAX proxy got along just fine with asterisk. The DCA one just refused to get it on. In addition, I didn't really want asterisk to register with or forward packets to the DCA one anyway. It was a continent and 124ms away, while the LAX one was only

40ms away.)

And no, I saw nothing very unusual about the sip packets asterisk sent to the DCA proxy. They looked fine to my untrained eye. Asterisk did seem to do the MD5 auth when challenged. I'm not sure what the DCA proxy was objecting to.

Can you tell if the Sipura tested both proxies and then latched onto the closer one?

-wolfgang

....

Mail from unknown[66.225.202.81]??? Ack. Jeremy needs to fire his DNS admin.

$ host 66.225.202.81 81.202.225.66.in-addr.arpa domain name pointer unknown.ord.scnet.net. (unknown.ord.scnet.net is, as implied, bogus with no such name found in DNS.)

-wolfgang

Oh! Oh! I want more info. I've been really disappointed by my Digium IAXy. I'd like to find something more capable which doesn't require frequent rebooting.

I want to find an FXS or FXS/FXO solution that I can configure and then send to family members and business associates. I don't want to worry about the devices being put on public networks and then hacked to pieces.

It looks like the SPA-3000 provides secure provisioning (with authentication and encryption). That's a great start. I'm also interested in encrypted voice communications. The brochure says something about "Secure RTP". Does that work with Asterisk? I have seen requests for it but no responses.

Would you recommend an SPA-3000 as something you could configure and send to someone as a "plug and play" solution?

Thank you.

--kyler

After the glowing reviews of Nufone from my friend, I was really frustrated by their attitude and ineptitude. I'm feeling *much* better now. Why? I found what appears to be a *wonderful* IAX provider.

After writing my friend about my disappointment with Nufone and VoicePulse, the only IAX providers I knew, he mentioned Gafachi.

My first impression of Gafachi was that they really concentrate on providing information. They don't have the polish of VoicePulse (not a bad thing to me); instead they provide what's needed.

I started filling in their "Sign Up!" form...and I just kept going. So often (like with VoicePulse) I get stopped because they want to do something stupid with JavaScript or cookies and I have to switch to another browser. Nope, not Gafachi. Within moments I was looking at the SIP, IAX, and IAX2 settings to use for my account. (Yes, I used the same Visa card that Nufone had taken such pains to refuse.)

Getting to that point was amazing enough but I'd gotten there with VoicePulse. Would this one work? Try some settings...I'm still not good at this, it took me a minute. Make a call. No go. Stick the password in the dial string (as they show on their Asterisk example page). Call again. My phone is ringing! Wow! That's an exciting moment.

Now back to the Web. The registration procedure left me at a page that showed my minutes used (0). Reload. Bam! There it is, a full log (time, called number, duration, cost, rate classifier, area) of the call just like I expected. I called my wife but got her voicemail. It sounded good but I have yet to hear from anyone how I sound.

Anyway, I am *thrilled* to find Gafachi. I don't know if they will provide consistently competent technical service (like I'm told Nufone does) but this is a fantastic start.

Hmmm...I see that they offer DID lines. Now to learn more...

--kyler

I haven't been forced to reboot my Sipura-3000 yet. Thats already infinitely better than my two Budgetone's.

If you want to peek at the user manual you can grab the .pdf from here:

I'm hoping that they were careful enough with the buffer overflow checks. The fact that it doesn't seem to crash is a good sign. I was planning to eventually move my sipura outside my firewall so that I could do a sip reinvite and eliminate a bit of extra delay.

I haven't been able to find enough information on how to generate the x509 "mini-certificate" that is needed. In fact, google comes up pretty dry for "mini-certificate".

My only reservation would be with NAT. If the person you send the sipura to hides behind a NAT and/or firewall, things could get messy.

I have heard of something call an STUN server you can run to help the NAT-ed box figure out its outgoing address and port number, but I have no experience with setting one up or how well it works in practice.

-wolfgang

My earlier tests were with my Grandstream phone, and with my own custom SIP software which does not try different proxies. I have not put a sniffer on the sipura as yet.

On the other hand I have made a cordless, battery powered phone with a Sipura and a wifi bridge!

Just out of curiosity, do you have any feelings on how to choose a sip proxy?

I'm leaning towards ignoring the priority value that the given in the SRV record and instead choosing the proxy by delay (ping or some sip NOP command) response time.

That sounds like a great hack.

I'm still waiting for the zyxel wifi voip phone to drop in price a bit before I jump in.

-wolfgang

Well, if the Proxies are going to want to load balance this is not going to scale. It might make more sense for proxies to test their connectivity to you (or even trust your statement about it) and then choose, but still have the power to load balance, since they know a lot more than you about that.

It's going to be a SIP based phone booth at Burning Man.

Actually, I've been trying to justify to myself the need for a wifi SIP phone to carry around. It's cool but for most users we are not quite there. In your house a cordless plugged into a Sipura is probably better and cheaper, and gives you a corded base. Out in the field cell phones are probably better as you need one anyway. We need a protocol to be adopted to let keyboardless devices talk to 802.11 hot-spots that require registration. Though for saving money, an

802.11 phone would be handy to take to a different continet where your roaming charges are obscene, but you could really only take and make calls when you specifically go to a hotspot. Over time that will change.

A corporate phone system could make sense with 802.11 phones, especially assuming you have put in the network for data already.

And at Burning Man it will be fun.