SOLVED Flowroute absolutely awesome

[tiggerpaws]

Sorry did not mean to flame them, I guess I have to be convinced BEFORE I belive anything.
If it means much to those who know doctor stuff, I have autism, that can affect how I think,
I'm just glad you were not around the day it failed on me, I raged and threw things and swore at it,
I have been fighting for 40 years to find a cure for what this condition plagues me, and I still have
not found it.

I am unable of "blind faith", I have to be showed something before I b elive it,
it has to be proven to me, as it has been here.

Sorry in the future if I do it again, I will try my best not to.

I thank you all again for such comprehension of my situation.

 

[b0b z33]

I began using Flowroute recently and have had nothing but a great experience plus near rockstar support

I'm also new to Asterisk/PIAF and have asked a "wobble-headded" question or two.

Flowroute has been very gracious

Found in most instances, it was conf file issues and nothing more

2 cents

 

[rastasean]

My in-laws have been using flowroute for 1 year next month (after my strong encouragement). for some crazy reason they ported their number to flowroute with the move.

Some stats about the calls:
Date range: 1 November, 2013 to 14 January, 2014
Total Calls: 309
Total minutes: 499
Inbound calls: 278
Outbound calls:31

About 98% of the inbound calls are telemarketers! I really would like to see an export from February 2013 to February 2014 but currently flowroute doesn't offer call data for starting periods 91 days or older in the past.
Fortunately the total billed is $5.855 including the USF fees, so while these calls are annoying, its less than a lunch to have them come in.

Those who are using flowroute, tell us about your monthly rates and minutes used. While I'm saving my employer ~$250 by making some adjustments to our current provider, I'm strongly considering testing flowroute out and then making it a recommendation to the bossman. I suspect I'll be able to reduce the bill to double digits per month with flowroute.

 

[AndyInNYC]

If you start a ticket with them they will email you a link to *all* the exported data you want. I was trying to decide if their 'all you can eat' pricing would work better than the pay by the minute plan I have.

They gave me access from 1/1/13 to 12/XX/13, where XX was the current date. All my calls, all the details.

I know I can find cheaper service elsewhere, but I don't know how I'd find better service. And, as was stated in another post - if I count up all my savings going cheaper and use it at Starbucks I would pre-spend my savings for the year by the end of January.


Andrew

 

[rastasean]

If you start a ticket with them they will email you a link to *all* the exported data you want. I was trying to decide if their 'all you can eat' pricing would work better than the pay by the minute plan I have.

Andrew,

That is awesome to hear! I'll create a ticket in about a month and request flowroute to provide an export for the entire year that my inlaws had the service. It will still have far more inbound calls than outbound because of the swarm of telemarketers.

 

[AndyInNYC]

If I were you, I'd phrase the request along the lines of trying to make sure the plan they have is the most cost efficient given Flowroute's pricing options (or something along those lines).

I absolutely love the quality of service and responsiveness I've gotten from the tech guys at Flowroute. I should probably send them cookies.

Andrew

 

[hbonath]

For SIP nerds... FlowRoute is actually pretty unique amongst VoIP aggregate/lcr providers i.e. Vitelity, etc.
What they do differently is reinvite your calls and connect you directly with the upstream providers. In doing that, they have a huge bandwidth savings as they are only proxying SIP Signaling only. RTP Media goes direct from your endpoint to the provider endpoint, competely bypassing their networks.
Pretty neat!
The tradeoff is that your latency is not always the same. For example, with Vitelity if you know that you are consistently 20ms away, you know that your latency in your media will always be endpoint latency + 20ms. With Flowroute, signaling will be consistent, but RTP media will depend on what provider you are being routed through because you connect *directly* to the provider's endpoint IP.
It's stuff like that which allows Flowroute to be very efficient and pass good cost savings to it's customers.

On a side note, not to get too off topic, but I wanted to reply to billsimon on his comment about IT providers getting into VoIP and thinking "It's just another application on the network!"
-HA!
I'm a veteran IT provider and have been in the business for over 15 years - I got into VoIP probably 2 1/2 years ago with those exact same thoughts in my head. Boy was I wrong! Learning all the Telco Lingo and acronyms, and doing number ports, etc. It's a whole new strange world. But this forum, and guys like Bill (Who personally helped me quite a bit with his Google Voice Gateway) when I was getting started are a fantastic resource and community. This is hands-down the best asterisk forum that there is! Love you guys!

 

[Rrrr]

Glad to hear about Flowroute through you guys.
As I am in Europe I immediately asked them about the usability of "Direct Media Delivery" and their Phil Baker responded immediately, well done!

Interestingly in my recent RASPBX for BBB, I already have a predefined trunk for Flowroute.
What confused me are its predefined settings, given here:

username=USERNAME

type=friend
secret=YOURPASSWORD
qualify=yes
insecure=port,invite
host=216.115.69.144
fromdomain=216.115.69.144
dtmfmode=rfc2833
disallow=all
directmedia=no
context=from-pstn-e164-us
canreinvite=no

allow=ulaw

I am not a SIP trunk expert, so even if things may work, I prefer to have a better understanding.
Here my three questions:
1. type=friend Is it not safer to use peer instead?
2. host=216.115.69.144 Is using an IP address better than what is given as example in post #15 of this thread or here
host=sip.flowroute.com ; use this when just benefiting from Flowroute DNS SRV records to to ensure redundancy
host=sip-lv1.flowroute.com ; to build your own primary and secondary trunk
host= sip-la1.flowroute.com
3. directmedia=no That is contrary to what is needed for the Flowroute infrastructure to work properly, no?

Instead of using above settings, I noticed that Flowroute actually provides a system configurator.
That is amazing, its the first provider I see doing that and providing you with complete customised trunk settings for your account!

I went for the extra secure option provided by Flowroute to disable outbound SIP credentials:
removed username and secret, replaced friend by peer, removed the registration string, added tech prefix into trunk Outbound Dial Prefix.
I also added Flowroute to iptables.
My trunk settings are now as follows and working fine:

type=peer
port=5060
insecure=port,invite
host=sip.flowroute.com
fromdomain=sip.flowroute.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw

I will be using them for a while to verify call quality to Europe.

 

[atsak]

There are several other carriers who do not proxy the RTP stream for anyone who is interested. Anveo Direct is one, and so is Voip Innovations, in case someone would like to compare various options. Those are two I use that work. Voip Innovations has minimums.

 

[jeff.h]

I have been using Flowroute for years across many different PIAF boxes and have never had a problem with their actual service. Sorry I am late to this thread, but here is the outbound trunk config that I use. I use SIP registration with them for incoming calls and as long as the DID on the incoming route has a 1 in front of it everything works fine. My context statement even formats the incoming callerID properly on the handsets and removes the +1 that many end users find annoying. I have been using this exact config for about 4 years now with zero issues. Hope it helps.

username=XXXXXXX
type=friend
secret=XXXXXXXXX
qualify=yes
insecure=port,invite
host=sip.flowroute.com
fromdomain=sip.flowroute.com
dtmfmode=rfc2833
context=from-pstn-e164-us
canreinvite=no
allow=ulaw

 

[mustardman]

One thing I absolutely hate about flowroute is IP authentication. It is probably the most needlessly complicated setup I have run into with any SIP trunk provider. Not even documented anywhere that I could find. I had to open a trouble ticket to get the setup info.

IP authentication is the way to go all other things being equal. Why flowroute can't get their act together in that regard is beyond me. Seems most other providers do it and make it painless so I don't see why they can't.

 

[jeff.h]

I am sorry you had trouble. Just log in, go to interconnection and outbound allowed IPs to enter your info. For inbound go to interconnection and inbound routes, enter the IP of your host then go to the DID page and use the functions drop down menu, pick the host you want, select the DIDs you want it to apply to and hit apply. Hope that helps.

 

[mustardman]

I am sorry you had trouble. Just log in, go to interconnection and outbound allowed IPs to enter your info. For inbound go to interconnection and inbound routes, enter the IP of your host then go to the DID page and use the functions drop down menu, pick the host you want, select the DIDs you want it to apply to and hit apply. Hope that helps.

It's a lot more than that unless you guys have changed things recently.

Here are the instructions I got. Again, not documented anywhere as far as I can tell. This is JUST for outbound dialing so maybe there is even more for DID?

Configuring IP Based Authentication for outbound calls requires two configuration changes:

1. Entering your system's Public IP addresses in Flowroute Manager under Allowed IPs in the Outbound section of your Interconnection tab: https://www.flowroute.com/accounts/interconnection/outbound/
2. Configuring your system to prefix the outbound number with your account's tech prefix followed by an asterisk.

Your system needs to send the dialed number in the SIP URI in the following format:
[Tech Prefix]*[Dialed number in E.164 Format]Sip.flowroute.com
The E.164 format for calls to the US and Canada is 1 + Area Code + Telephone Number. For example Flowroute's number is (206) 641-8000, in E.164 format it's 12066418000. For calls to other countries the E.164 format is Country Code + Number. For example to call 1 5871 1234 in France (country code 33) the number in E.164 format is 33158711234.
Using the example tech prefix of 00000000, a call to (206) 641-8000 should be sent using the following SIP URI:
00000000*[email protected]
Similarly a call to 1 5871 1234 in France, country code 33, should be sent using the following SIP URI:
00000000*[email protected]

 

[jeff.h]

If you use the settings in the outbound config for your trunk in your PBX that I shared in post #31 and put the correct info in their webpage as I shared in post 33 then your trunk should be online and allow outbound calls which will be properly formatted to pass traffic through Flowroute. That process has been the same for years.

They even have a link to their system configurator page right on the Interconnection page as well. It is full of info specific for your account.

 

[mustardman]

If you use the settings in the outbound config for your trunk in your PBX that I shared in post #31 and put the correct info in their webpage as I shared in post 33 then your trunk should be online and allow outbound calls which will be properly formatted to pass traffic through Flowroute. That process has been the same for years.

They even have a link to their system configurator page right on the Interconnection page as well. It is full of info specific for your account.

They don't have the instructions on their website as far as I know so I think what you are saying is a bit misleading. The info spec as you call it has the required Tech prefix number and that's about it. You still have to add it in based on their instructions which don't even mention how to do this in Freepbx (Dialed number manipulation rules in trunk settings).

If you are using IP authentication as you claim then why are you using a username/password in your configuration? My working configuration does not use a username/password. Hence why it's called IP authentication as opposed to username/password authentication.

 

[jeff.h]

All the information you need to connect to flowroute is available on their site. It sounds to me like you need to know how that information applies to PIAF/asterisk. You'll need to learn that by researching and asking questions around on this forum. It's not the carrier's responsibility to tell you how to configure your own equipment.

IPs are also very easy to spoof, you really should think about also having a username/password setup as well as only allowing certain IPs to connect.

 

[mustardman]

All the information you need to connect to flowroute is available on their site. It sounds to me like you need to know how that information applies to PIAF/asterisk. You'll need to learn that by researching and asking questions around on this forum. It's not the carrier's responsibility to tell you how to configure your own equipment.

IPs are also very easy to spoof, you really should think about also having a username/password setup as well as only allowing certain IPs to connect.

IP authentication is actually considered more secure in addition to being much simpler. That is why it's usually the recommended method by providers who support both. I don't know any providers that use both at the same time.

I don't think you are using IP authentication so we are not even on the same page. Sign up for service with voip.ms or vitelity and see how simple it is. Check the box and enter your IP address. Done. No usernames, no passwords, no registration strings.

If you are happy with flowroute then great. If you want to defend their overly complicated setup as "not their responsibility" maybe they should use that to market their services. "We are better because our set up is more complicated and we don't even bother providing the documentation for it because it's just that awesome!" Sounds like a winning sales strategy to me. If someone wants to just use username/password then I'm sure they are fine for that. For me it's kind of a deal breaker.

 

[AndyInNYC]

Perhaps I'm confused.

If I log in to my flowroute account and go to "Interconnection" and then (it defaults to it) "Status and Setup", I then have in the bottom left hand corner of the page "System Configurator"

If I click on that and select "trixbox/Freepbx" I'm given a step by step guide to configuring a new flowroute trunk with a whopping 4 steps. This info includes a cut and paste with my personal (password) settings which get pasted into the trunk setting. Once you've followed the total of 4 steps (and one of the steps is telling freepbx you're adding a trunk, a sip trunk, for example) you're up and running without pain or sweat.

I'm not commenting on security or IP authentication, etc., just that to describe flowroute's setup as "overly complicated" is a bit absurd. If it takes more than 20 seconds (including on an RPi) from start to finish, the issue is with the operator, not the system.

Andrew

 

[mustardman]

Perhaps I'm confused.

If I log in to my flowroute account and go to "Interconnection" and then (it defaults to it) "Status and Setup", I then have in the bottom left hand corner of the page "System Configurator"

If I click on that and select "trixbox/Freepbx" I'm given a step by step guide to configuring a new flowroute trunk with a whopping 4 steps. This info includes a cut and paste with my personal (password) settings which get pasted into the trunk setting. Once you've followed the total of 4 steps (and one of the steps is telling freepbx you're adding a trunk, a sip trunk, for example) you're up and running without pain or sweat.

I'm not commenting on security or IP authentication, etc., just that to describe flowroute's setup as "overly complicated" is a bit absurd. If it takes more than 20 seconds (including on an RPi) from start to finish, the issue is with the operator, not the system.

Andrew

Yes I am aware they have user/pass instructions. It's apples and oranges (shouldn't be but for their service it is) so not sure what your point is.