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Vonage 2.0.10 Unlock question! »
« [Femtocell] The Beginning of the End of Residential VOIP  
page: 1 · 2
AuthorAll Replies


reho33

@ptd.net

Sipbroker Down?

My phones would not dial out tonight and I traced it to sipbroker not coming up as a website. I have sipbroker.com in my dialplan but the outage has rendered it useless so I have to go back to the old fashioned dial plan that came with the pap2t-na. Phones work great now but I won't get the savings on outgoing 1-800 style numbers. Has anyone experienced an "outage" with sip broker before? Post info and any history as I have had 3 outages in 6 months. I understand that it is a "free" service and not guaranteed. Just curious.


sream
Premium
join:2002-08-17
Portage, MI
Yeah, it's down.

PX Eliezer
Premium
join:2008-08-09
New Jersey
reply to reho33
Website is down now---1st time I've ever seen that---but the service itself works fine.

There are various other free toll-free termination services for your 800 number calls---or you can use Future/Nine.


reho33

@ptd.net
reply to reho33
OK I use pap2t-na and want to weave future - nine into my dial plan.....Ideas? Nitzan owns it maybe he knows? Please excuse my lack of dp knowledge...thanks in advance.

nitzan
Premium,VIP
join:2008-02-27
·ViaTalk
·Comcast

Please do this:
1. Create an account at: »https://www.future-nine.com/A2BCustomer_UI/signup/
2. Set your ATA dial plan to route TF through FN:
3. Set:
Enable IP Dialing: Yes
Make Call Without Reg: Yes

This should hopefully work in theory, although I haven't actually tested it without registering.

kieranmullen
Premium
join:2005-12-12
Portland, OR
clubs:
·Gizmo5
·Skype
·Vitelity VOIP
·magicjack.com
·Verizon FIOS
·Vonage
·ViaTalk
·VoicePulse

reply to reho33
What savings do you get from it just curious. Most people boo sipbroker.com and e164.org here. I use both sipbroker.com when dialing in on their shared numbers and e164.org for lookups and toll free.

I checked and my toll free numbers are still being terminated through e164.org (Then again the owner did say they have a redendant setup)

»www.rexswain.com/cgi-bin/httpview.cgi is able to pull up a 301 redirect which is more than I came up with on my end.


sbd

@rr.com
reply to reho33
There website is definitely down; but I can still use at least 1 of their dial-in access numbers, and complete calls via the *1800xxxxxxx (for example) free toll-free access.

PX Eliezer
Premium
join:2008-08-09
New Jersey
reply to reho33
Their site is Up again this morning.

Guess they found their ED treatment.


reho33

@verizon.net

reply to reho33
Thanks to all for all your help But I just wanted to make some observations here:
1) My Linksys PAP2T-NA has the following dial plan as Sip broker recommended :

L:20,S:7,( x | *xx | 911S0 | 1[2-9]xx[2-9]xxxxxxS0 | [x*][x*]. | [x*][x*]. )

2) According to their website, this dial plan will attempt to send calls via sipbroker.com, if it fails that, calls will be routed via default route aka your pay provider.

3) What I found out last night is that NO calls go through at all when sipbroker.com domain is "un-resolvable". In theory the call should be routed via the default route but it just "hangs" and fast busys after a long pause. Not even my carriers test number (811) will work. I mean 911 should go straight out hence 911SO but I fear that if Sipbroker went down again, maybe that would nit work either!

4) This morning, it is back up as all of you have told me. But maybe Nitzan can answer this..... How does 800 terminations work? I guess you as a carrier get some money to terminate these calls so you like having the traffic redirected to you? VOIP MS is my provider but they charge you for making 800 calls. Now, in the overall scheme of things, that is not too bad compared to the stability and quality they provide. But it can add up, if a person is on hold with the bank for example it can cost 60 cents for 1 hour on hold. What I don't understand is if they profit from the termination, why do they have to charge the orginator for the call also? Please over look my ignorance on these matters as I am a noob but as a whole, what are the "rules" for 800 terminations on VOIP and who makes money?

hwittenb

join:2003-12-20
·Future Nine Corpor..
·callwithus
·Callcentric

said by reho33 :

(1)My Linksys PAP2T-NA has the following dial plan as Sip broker recommended :

L:20,S:7,( x | *xx | 911S0 | 1[2-9]xx[2-9]xxxxxxS0 | [x*][x*]. | [x*][x*]. )

2) According to their website, this dial plan will attempt to send calls via sipbroker.com, if it fails that, calls will be routed via default route aka your pay provider.
What you posted isn't exactly what the SipBroker web site says, the web site says send the call to the number @sipbroker.com. You did, however, hit on the reliability issue that Nitzan posted about in another thread. The logic relies on receiving a response from SipBroker.
»faq.sipbroker.com/tiki-index.php

You are mistaken about not being able to call 911 with the dial plan. The dial plan doesn't send the 911 call to sipbroker, it sends it to the voip provider that you have registered on Line 1.

hwittenb

join:2003-12-20
·Future Nine Corpor..
·callwithus
·Callcentric

reply to nitzan
said by nitzan See Profile :

This should hopefully work in theory, although I haven't actually tested it without registering.
Nitzan,

On my SPA3102, with Line 1 registered to someone else, the dial plan you posted doesn't work. The outgoing request goes to 65.111.186.2 but Future-Nine immediately returns a Sip 404 Not Found. In my opinion this is because the usr and psw information in the dial plan is not in the Sip Header. The dial plan matches the example given in the Linksys/Cisco ATA Administration guide on pg 102 in the current edition. I think it is a Cisco/Linksys bug. I never was able to get something like that working with the dialplan.

I can send you a WireShark trace if you like.

nitzan
Premium,VIP
join:2008-02-27
·ViaTalk
·Comcast

Thanks. I only did very little testing of this and had problems too so you might be right and it's a Sipura/Linksys bug. (i.e. what they say in the manual is wrong)

I guess it makes more sense to test it while looking at the server. I'll do some testing and let you guys know if we resolve it. The other alternative is just to open everyone to terminate TF calls without a user/password.

nitzan
Premium,VIP
join:2008-02-27
·ViaTalk
·Comcast

reply to reho33
said by reho33 :

what are the "rules" for 800 terminations on VOIP and who makes money?
There are three options:
1. You (VoIP carrier) pay for those calls.
2. You get those calls free but with no compensation.
3. You get compensated for these calls.

1 & 2 are common. 3 is rare at this point. There are a few CLECs that pass a "kickback" to the VoIP provider for terminating TF calls through them. It's not much, but it's better than nothing, and overall quality through them is much better than some other routes that let you terminate for free but quality sucks.

I think voip.ms either doesn't want people callinng TF through it (which is stupid), or they're paying for those calls (which is likely).


sream
Premium
join:2002-08-17
Portage, MI
reply to reho33
Both Link2voip and voip.ms charge $.0025/min to toll free. Thats a quarter penny.


reho33

@verizon.net

reply to reho33
OK, then things must have changed since I last looked. Thanks to all for your answers. But I am like Mazilo......why even pay that when free is better. VOIP.MS charges 6 cents an hour for talking to internal extensions. I go adapter- to -adapter and pay 0.00. 6 cents is not a lot but I choose to max. savings where I can.


sream
Premium
join:2002-08-17
Portage, MI
·link2voip
·voip.ms

reply to reho33
Nothing wrong with free. I've got a bunch of free European did's from bluesip, eutelia, messagenet. But the time I spent signing up for them and entering the registrations (no forwarding) in mysipswitch saves me significant money. Even if you value your time at something low like $20/hour you would have to make a whole lot of tf calls to make it worthwhile.

hwittenb

join:2003-12-20
·Future Nine Corpor..
·callwithus
·Callcentric

reply to nitzan
said by nitzan See Profile :

The other alternative is just to open everyone to terminate TF calls without a user/password.
That's what FWD does, they don't authenticate the call, but you have to precede the number with an asterisk (*).

Looking some more at the packet trace with your original dialplan code vs. doing it where the account is setup on one of the gateways on the SPA3102 the difference is the "Contact" information in the initial header. With the gateway way my future-nine account number is shown, with the dial way the future-nine account number is not shown. I think it is a bug. The password authentication comes later. There may be a bug there too.

mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA

reply to sream
said by sream See Profile :

Nothing wrong with free.
And, if I may add to this: FREE IS BETTER!.

I've got a bunch of free European did's from bluesip, ...
This company assigns a non-geo EU DID number to new member, doesn't it?

Even if you value your time at something low like $20/hour you would have to make a whole lot of tf calls to make it worthwhile.
Honestly, when I encounter a person who values his/her time in term of $$$, I sure think this person is living a pathetic lifestyle. The reason is simply this person will be wasting a lot of his $$$ earning through his night/day sleeps. I always have opportunity to make more $$$$ to accept new clients; however, I mostly reject any new clients mainly because I don't want to spoil my leisure to chasing $$$$, YMMV.
--
Mazilo always prays for FREEBIES!
US Phone: +1-678-601-0907
UK Phone: +44-703-194-2574

nitzan
Premium,VIP
join:2008-02-27
·ViaTalk
·Comcast

What good is a lot of $$$ if you can't actually enjoy it?

Funny thing is, if you REALLY want to measure your time in money, then the calculation is WRONG. Let's say you value your time at $20/hour, which puts you in the 25% tax bracket.

So working an extra hour would net you:
$20 base - $5 (income tax) - $1.53 (social sec & medicrap) - $1.6 (state income tax) = about $11.87/hour
The employer pays another $1.6 on top of that (soc sec, medi, and various other crap taxes)
(if you're self-employed, $20/hr would actually net you $10/hr!)

So in essence, it is a FAR better alternative to do something yourself than pay someone else to do it - unless it is deductible to a business in which case the taxes above don't apply. Anything you pay out of post-taxes money should be really measured in post-taxes money and not how much you're being told you're making.

Did I mention I hate taxes?

kieranmullen
Premium
join:2005-12-12
Portland, OR
clubs:
·Gizmo5
·Skype
·Vitelity VOIP
·magicjack.com
·Verizon FIOS
·Vonage
·ViaTalk
·VoicePulse

Why bother making these type of calculations for hobbyists who enjoy playing around with this stuff? Extensive dial plans aren't really for ATA's now are they? Let's be real. Not saying one can't or shouldn't but if it were a business, one would most likely have several fallback providers SIP/IAX2 and even some hardware lines out as I do on ours.

said by nitzan See Profile :

What good is a lot of $$$ if you can't actually enjoy it?

Funny thing is, if you REALLY want to measure your time in money, then the calculation is WRONG. Let's say you value your time at $20/hour, which puts you in the 25% tax bracket.

So working an extra hour would net you:
$20 base - $5 (income tax) - $1.53 (social sec & medicrap) - $1.6 (state income tax) = about $11.87/hour
The employer pays another $1.6 on top of that (soc sec, medi, and various other crap taxes)
(if you're self-employed, $20/hr would actually net you $10/hr!)

So in essence, it is a FAR better alternative to do something yourself than pay someone else to do it - unless it is deductible to a business in which case the taxes above don't apply. Anything you pay out of post-taxes money should be really measured in post-taxes money and not how much you're being told you're making.

Did I mention I hate taxes?
-
Forums » VOIP etc » Voice Over IP - VOIP » VOIP Tech ChatVonage 2.0.10 Unlock question! »
« [Femtocell] The Beginning of the End of Residential VOIP  
page: 1 · 2


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