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SCADAGeo

join:2012-11-08
N California
kudos:2
reply to PX Eliezer70

Re: SIP forwarding

said by PX Eliezer70:

If it is done by creating a Virtual SIP number and using that as an inbound SIP, there is a nominal fee:

Monthly fee is $0.25 (25 cents) / month per active Virtual SIP number. Incoming calls are billed $0.001 (one tenth of a cent) per minute.

»www.voip.ms/m/virtualnumbers.php

HOWEVER, a method that is quite similar (using a subaccount to make an internal extension, and using that internal extension as an inbound SIP, appears to be [free]. Their website PREVIOUSLY talked about a charge of one-tenth of a cent per minute for THIS method, but now the language is gone.
»www.voip.ms/m/subaccount.php

I used sub-accounts for testing, and I wasn't aware of the first method. Thanks for the clarification.

PX Eliezer70
Premium
join:2008-08-09
Hutt River
kudos:13
Reviews:
·callwithus
·voip.ms

1 recommendation

reply to SCADAGeo
said by SCADAGeo:

When I tested VoIP.MS's inbound SIP, I did it without funding my account, so it appears as if it the inbound SIP calls are free - don't know if this is still true...

If it is done by creating a Virtual SIP number and using that as an inbound SIP, there is a nominal fee:

Monthly fee is $0.25 (25 cents) / month per active Virtual SIP number. Incoming calls are billed $0.001 (one tenth of a cent) per minute.

»www.voip.ms/m/virtualnumbers.php

HOWEVER, a method that is quite similar (using a subaccount to make an internal extension, and using that internal extension as an inbound SIP, appears to be [free]. Their website PREVIOUSLY talked about a charge of one-tenth of a cent per minute for THIS method, but now the language is gone.
»www.voip.ms/m/subaccount.php

SCADAGeo

join:2012-11-08
N California
kudos:2
reply to PX Eliezer70
said by PX Eliezer70:

There CAN be small Voip.MS charges for [inbound] SIP calling.
»What is the voip.ms sip url?

When I tested VoIP.MS's inbound SIP, I did it without funding my account, so it appears as if it the inbound SIP calls are free - don't know if this is still true...

Outbound calls, even to extensions on sub-accounts, requires a funded account, but there is no cost for the call.


Arne Bolen
Happy Anveo customer
Premium
join:2009-06-21
Cyberspace
kudos:4
Reviews:
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·voip.ms
reply to ConstantineM

Re: Alternative to voip.ms ?

said by ConstantineM:

P.S. It's surprising noone ever mentions pbxes.org on this forum. I wish I would have found them earlier.

PBXes has often been mentioned on this forum by many members including myself.
--
My VoIP News

PX Eliezer70
Premium
join:2008-08-09
Hutt River
kudos:13
Reviews:
·callwithus
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reply to OZO

Re: SIP forwarding

said by OZO:

"SIP URI" routing is billed using usual incoming rates and not free.

No, it is billed at a [very small] charge.

cf:
»What is the voip.ms sip url?

grand total

join:2005-10-26
Mississauga
kudos:2
Reviews:
·VMedia
·Anveo
reply to OZO
said by OZO:

I was under the impression, that you have mentioned "Call Forwarding", while you actually meant "SIP URI" routing...

I thought you were confusing the two and went along with your vocabulary.
--
DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server
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OZO
Premium
join:2003-01-17
kudos:2
reply to grand total
I was under the impression, that you have mentioned "Call Forwarding", while you actually meant "SIP URI" routing...
--
Keep it simple, it'll become complex by itself...

grand total

join:2005-10-26
Mississauga
kudos:2
Reviews:
·VMedia
·Anveo
reply to OZO
said by OZO:

It's not "Call Forwarding". They called it "SIP URI" routing.
"SIP URI" routing is billed using usual incoming rates and not free.

Yes, I don't think anything you have written here conflicts with anything I have posted in this thread.
--
DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server
Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI

OZO
Premium
join:2003-01-17
kudos:2
reply to grand total
It's not "Call Forwarding". They called it "SIP URI" routing.
"SIP URI" routing is billed using usual incoming rates and not free.
--
Keep it simple, it'll become complex by itself...

grand total

join:2005-10-26
Mississauga
kudos:2
reply to OZO
DID Numbers|Manage DID|Edit DID shows DID routing. One option is to route the DID to SIP URI.

OZO
Premium
join:2003-01-17
kudos:2
reply to grand total
How to configure "Call Forwarding" to SIP URI in VoIP.MS?

They have "Call Forwarding" page, which allows to add a new forwarding rule. It has "Phone Number" field where it accepts only digits and not any SIP URI...
--
Keep it simple, it'll become complex by itself...

PX Eliezer70
Premium
join:2008-08-09
Hutt River
kudos:13
Reviews:
·callwithus
·voip.ms
reply to zapattack
said by zapattack:

V.MS does not appear to offer SIP CFWD.

As [grand total] said, that is incorrect. They do offer free outbound SIP forwarding.

There CAN be small Voip.MS charges for [inbound] SIP calling.
»What is the voip.ms sip url?

said by zapattack:

The menu further warns that BOTH incoming and outgoing charges apply.

Outbound charges would apply only to calls forwarded out to phone numbers, not to SIP URI.

grand total

join:2005-10-26
Mississauga
kudos:2
Reviews:
·VMedia
·Anveo
reply to zapattack
said by zapattack:

V.MS does not appear to offer SIP CFWD. The menu further warns that BOTH incoming and outgoing charges apply.

Huh? Of course VoIP.MS offers SIP forwarding. If you forward a call to a SIP URI you only pay for the incoming call (as usual) the forwarding is free.
--
DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server
Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI

zapattack

join:2012-07-02
CANADA
reply to ConstantineM
V.MS does not appear to offer SIP CFWD. The menu further warns that BOTH incoming and outgoing charges apply.
For my light usage, it is easier to let PBXes and V.MS deal with DDoS problems. As PBXes does not offer DIDs, it is a simple matter to change PBXes servers if one is subject to an attack.
The last problem was NY area congestion on the route to Germany during Sandy, so I changed to Seattle.

ConstantineM

join:2011-09-02
San Jose, CA
reply to Goliath13
Basically, yes. But, again, you first have to make sure that VoIP.ms has a SIP-forwarding server that is more-or-less immune to DDoS; I think it's relatively easy to establish such a server or two, and I think I would be more surprised if they didn't have one already, but, on the other hand, I'm not certain they specifically have such a server, since they didn't seem to have suggested such a solution yet as one of the alternatives.

You can still try it to see if it might work for you. CallWithUs lets you provide any +1 number as Caller ID (non-+1-numbers don't seem to be sent correctly to at least +1 destinations, see »CallWithUs Caller ID to T-Mobile USA prepends a "1"), and Anveo also lets you provide +1 or +44 numbers as Caller ID (probably all other codes work, too, but you have to verify numbers first).

pbxes.org has a feature where you can integrate it with external providers. You would integrate it with VoIP.ms only from within the VoIP.ms interface itself, for your incoming calls (e.g. just do SIP forwarding from VoIP.ms to pbxes.org), and then integrate CallWithUs and/or Anveo from within pbxes.org interface for outgoing calls. (Integrating voip.ms from within pbxes.org would be kinda pointless; since the whole exercise is to avoid the external SIP registration servers of VoIP.ms that may be prone to DDoS.) Then register your phones with pbxes.org. They, pbxes.org, do seem to have some minute-based usage limits on free accounts, but you can still see if something like that fits your needs.

Also, CallWithUs lets you have multiple sub-accounts for free, too, and they also support TCP, so you might also try going with CallWithUs and do forwarding onto CallWithUs, avoiding the pbxes.org route. The only negative about CWU accounts is that the usernames are random numbers with no aliases, so, SIP calling with a CWU account is not user-friendly. But since they let you specify your own Caller ID, the integration can be quite nifty for PSTN calling.

Goliath13

join:2009-07-10
reply to ConstantineM
So, based on this type of setup, how would you configure this?
Create a new account with another carrier and just have it accept forwarded SIP URIs from voip.ms?

ConstantineM

join:2011-09-02
San Jose, CA
reply to zapattack
The whole purpose of avoiding VoIP.ms DDoS is to _not_ register with any of their servers (neither directly nor through pbxes.org). Don't they let you forward calls to external SIP addresses for free? Provided it is so, such an approach is supposed to be more reliable than manually registering, especially during DDoS (again, with a good architecture, forwarding is supposed to be immune to DDoS and such, but I cannot say whether VoIP.ms forwarding is or is not).

In theory, it's impossible to DDoS a server that's only doing the forwarding, since the only way to initiate connections is through PSTN. But if VoIP.ms only has one server at each location, and with only one interface and one IP-address, and their forwarding is distributed to a location where you would otherwise manually register through SIP, then, of course, it's a different question, and traffic shaping would probably not be there or may not have much effect if the interfaces are completely saturated.

I would assume that if they don't confirm that it's more reliable, then maybe in the end it is not. :-)
Expand your moderator at work

zapattack

join:2012-07-02
CANADA
reply to ConstantineM

Re: Alternative to voip.ms ?

PBXes.org with V.MS is problematic. It refuses to let me register to most of the American servers. Fortunately LosAngeles does work for me and is close enough.
Also the V.MS portal shows me as unregistered (to PBXes) at times, even though DID calls complete.
The free PBXes package doesn't give you any way of submitting a real trouble ticket.
But, for the amount of working (FREE) features, it is still a good deal and, strangely, more reliable than V.MS.

ConstantineM

join:2011-09-02
San Jose, CA

1 recommendation

reply to Goliath13
Depending on your usage, and if Anveo addons are deemed expensive, you might also integrate pbxes.org into the mix (although it also has its own usage limits), and, for example, configure pbxes.org to use Anveo for incoming calls (or, hey, for starters, you can even try it with VoIP.ms!), and CallWithUs for outgoing etc.

And, yes, you might as well try to use pbxes.org with VoIP.ms itself; in theory (and depending on the specific architecture of their setup), they're supposed to be able to forward incoming calls just fine even when under a relatively severe attack (how it is in practice, I don't know, plus it seems that some of their colocation providers (e.g. the one in Atlanta) don't provide them with "reasonable", whatever that would be, DDoS protections/handling).

P.S. It's surprising noone ever mentions pbxes.org on this forum. I wish I would have found them earlier.

PX Eliezer70
Premium
join:2008-08-09
Hutt River
kudos:13

1 recommendation

reply to Goliath13
I've never needed to do that but there are lots of folks here who can advise you.

Note that there are a lot of choices besides Asterisk, such as YATE.

You may want to start a new thread asking for advice on PBX setup.

Mango
What router are you using?
Premium
join:2008-12-25
www.toao.net
kudos:13
Reviews:
·AcroVoice
·Callcentric
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reply to Goliath13
It's easy to set up Asterisk. But it's not easy to build a fully-featured, high-reliability business phone system. It's certainly possible, and we'll be delighted to give you advice if you need it, but I wouldn't call it easy.

There is some value in using a PBX which allows you to use multiple unreliable service providers, but I would rather use a reliable service provider with PBX features. In this case there are typically fewer points of failure.

My suggestions would be for you to use Anveo if you want to DIY, or otherwise hire a company such as AcroVoice to set up and maintain your system.

SCADAGeo

join:2012-11-08
N California
kudos:2

1 recommendation

reply to Goliath13
said by Goliath13:

It seems to me that if we can get an Asterisk set up going, we would have the most flexibility to go with whichever provider we want. I'm just not super confident in my ability to set it up nicely. How hard would this be to setup for a noob?

It's very easy: The 5-Minute PBX: It’s Incredible PBX 11 Virtual Machine for VirtualBox.

Goliath13

join:2009-07-10
reply to PX Eliezer70
I've looked into PBXes as well which is another solution.

It seems to me that if we can get an Asterisk set up going, we would have the most flexibility to go with whichever provider we want. I'm just not super confident in my ability to set it up nicely. How hard would this be to setup for a noob?

PX Eliezer70
Premium
join:2008-08-09
Hutt River
kudos:13
Reviews:
·callwithus
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reply to Goliath13
said by Goliath13:

I need to have 3 different DIDs because they handle different parts of our businesses basically.

OK, that's good to know. So it [would] be easiest for you to have a provider with subaccounts or the equivalent.

Sometimes we see folks who ended up having multiple DID's in the POTS days even though they didn't need them. Your setup clearly does need them....

Goliath13

join:2009-07-10
reply to PX Eliezer70
Sorry, I didn't understand that question PX Eliezer.

We might have a situation where 2 calls could be coming in simultaneously into 2 different DIDs. I need to have 3 different DIDs because they handle different parts of our businesses basically. One DID for business A, one DID for business B & C and toll free for business C.

PX Eliezer70
Premium
join:2008-08-09
Hutt River
kudos:13
Reviews:
·callwithus
·voip.ms
reply to Goliath13
Do you need/use your 3 DID's or do you just need the ability to get 3 incoming calls at a time (which could for example be [one] DID with 3 channels)?

With a modern IP phone plus extra handsets (such as a Panasonic or late-model Gigaset as often discussed here) a [single] DID with multiple channels can do for you what you used to need multiple POTS DID's.

Goliath13

join:2009-07-10
reply to connor79
Thanks for the help connor79.
The lines are for a business environment but our level of usage is very low. We probably have 400 minutes of usage a month between the 3 lines. While we have low usage, it's definitely important to not miss calls from clients or have some other issues with the calls.

I thought about doing some kind of Asterisk setup but I'm not a networking specialist or anything and voip.ms allowed us to achieve PBX functionality without complicating the setup with another piece of equipment.


Trimline
Premium
join:2004-10-24
Windermere, FL
Reviews:
·ObiVoice
·Bright House
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reply to Goliath13
Check out »www.vitelity.com/services_voip/

They also offer "sub-accounts" for multiple sip registrations on a DID.

They can do outbound CNAM for a one time charge as well. There are discounts available, so check around.


Davesnothere
No-BHELL-ity DOES have its Advantages
Premium
join:2009-06-15
START Today!
kudos:7
reply to Mango
said by Mango:

said by MartinM:

I'm currently on a yacht in Mexico for New Year, and that's how much I care about our customers: While everyone is sipping champagne, I'm surfing DSLReports.

You have no idea how envious I am. I've always thought this to be an excellent combination.

 
Yacht-ernet ?

And to paraphrase/bork an old song :

♫ ♬ "Don't it make my Blue eyes Green !" ♫ ♬

--

We have only 2 things about which to worry :
(1) That things may never get back to normal
(2) That they already HAVE !