| reply to ThaiGuy
Re: Reliable Outgoing Toll Free Calls without Reg Voxbeam passes caller ID correctly on TF calls.
However, if you are not also using their paid service, there may be a gotcha: the free $1 test credit that you get at signup might expire after 365 days, just like a regular payment. And, without a balance, even a free call might be rejected. |
|
garys_2kPremium
join:2004-05-07
Farmington, MI Reviews:
 ·Callcentric
·Future Nine Corp..
| reply to nitzan
said by nitzan:said by garys_2k:Nitzan, I route my TF through you. Can you accept 855 calls or should I still send them elsewhere?
You should be able to. Give it a try and let me know if any problems. It works, thanks! |
|
|
|
| reply to nitzan
said by nitzan:said by ThaiGuy:Can anyone explain why Future-Nine and CallWithUs might not be able to deliver CallerID correctly? If you send us anything other than a 10-digit CID we'll override it. If you send a 10-digit CID it should pass through as-is. I was previously aware that F9 didn't pass cid, Nitzan said it himself. I just tested with one of my other test numbers, 800-Comcast. If you send a phone number of a customer, it reads off the last 4 digits. I couldn't get it to work on F9, via the open route, but worked on others like tf.callwithus. As others said, 800 all 4's doesn't work, but that number is picky about cid/cpn as there is a mis-trust thing going on with the carriers contacting their switch, so I don't take it as 100% failure. |
|
 TrimlinePremium
join:2004-10-24
Windermere, FL | reply to nitzan
Out of curiosity, what is the maximum call length using tf.callwithus? |
|
nitzanPremium,VIP
join:2008-02-27
kudos:2 | reply to ThaiGuy
For the CID issue, please message me privately with:
1. How were you calling? (registering and sending calls, or sending without registering)
2. What CID did you set your equipment to send?
3. What TF number did you call? |
|
| reply to Trimline
We run www.tollfreesipgateway.com, Thanks for the support!
Not only do we take all of the toll free traffic, we also pay you for it!
I am pretty sure every VoIP provider limits the call time at 4 hrs to prevent run away calls. 
You can also visit www.teleinx.com for more info. |
|
 espaethDigital PlumberPremium,MVM
join:2001-04-21
Minneapolis, MN
kudos:2
 ·Clear Wireless
| said by clearchannel:I am pretty sure every VoIP provider limits the call time at 4 hrs to prevent run away calls. Definitely not all. I made another call to our TF conference line via Vitelity than ran just over 10 hours yesterday.
FlowRoute cuts off at 8 hours, I believe. |
|
PX EliezerPremium
join:2008-08-09
Hutt River
kudos:12 Reviews:
·voip.ms
·callwithus
·Callcentric
·Vitelity VOIP
·Optimum Voice
·Gizmo5
| reply to clearchannel
Your website does not display properly on Firefox-type browsers.
Considering that many of your potential users are smart people who don't use Microsoft Internet Explorer, you may want to fix that.  |
|
PX EliezerPremium
join:2008-08-09
Hutt River
kudos:12 Reviews:
·voip.ms
·callwithus
·Callcentric
·Vitelity VOIP
·Optimum Voice
·Gizmo5
| reply to clearchannel
Your VoIP services are quite new, anything else you'd like to highlight about them (any special features etc)? |
|
| reply to ThaiGuy
is it possible that 8004444444 (and others) are not working with passed CID because they are reading ANI instead? |
|
| said by nickdigger:is it possible that 8004444444 (and others) are not working with passed CID because they are reading ANI instead?
800-437-7950 announces both CLI and ANI, and I'd assume that most providers would route that the same as 800-444-4444, since they are both MCI. |
|
TrimlinePremium
join:2004-10-24
Windermere, FL Reviews:
·voip.ms
·Callcentric
 ·RoadRunner Cable
| reply to PX Eliezer
said by PX Eliezer:Your website does not display properly on Firefox-type browsers. Considering that many of your potential users are smart people who don't use Microsoft Internet Explorer, you may want to fix that. I tried with IE too, looks kludge there as well. For a pretty page, use »www.teleinx.com/paid-toll-free/. With this service, you do have to sign-up first as they require your IP address to be entered in to their DB. |
|
TrimlinePremium
join:2004-10-24
Windermere, FL Reviews:
·voip.ms
·Callcentric
·RoadRunner Cable
| reply to espaeth
said by espaeth:said by clearchannel:I am pretty sure every VoIP provider limits the call time at 4 hrs to prevent run away calls. Definitely not all. I made another call to our TF conference line via Vitelity than ran just over 10 hours yesterday. FlowRoute cuts off at 8 hours, I believe. Which brings up a fairly good point. I wonder what the max call time is with these services actually is. I can safely say that tollfreegateway is 4 hours. What provider are you using for 10 hours? |
|
OZOPremium
join:2003-01-17
kudos:2 | reply to Trimline
Too bad, 'cause I need to make calls with my smartphone from any location... Don't care about "pretty page" though.
--
Keep it simple, it'll become complex by itself... |
|
espaethDigital PlumberPremium,MVM
join:2001-04-21
Minneapolis, MN
kudos:2 Reviews:
·Clear Wireless
| said by OZO:Too bad, 'cause I need to make calls with my smartphone from any location... Don't care about "pretty page" though. So route it through an Asterisk/FreeSwitch/whatever PBX with a fixed IP.
Problem solved. |
|
OZOPremium
join:2003-01-17
kudos:2 | You forgot to mention the extra hop to your SIP server, causing bigger latency and more dropped voice packets, associated with using the intermediate server... So, I'd say, problem added.
--
Keep it simple, it'll become complex by itself... |
|
espaethDigital PlumberPremium,MVM
join:2001-04-21
Minneapolis, MN
kudos:2 | I forgot to mention those things because they are almost all untrue or inconsequential. (Hint: Direct RTP)
But hey, use whatever works for you. |
|
OZOPremium
join:2003-01-17
kudos:2 | Untrue and inconsequential? Good for you...
May be you know that FreeSWITCH doesn't support direct RTP with Google Voice. It means, that, when I make calls using GV, I have no choice, but to send RTP to FS (and not to GV directly). If I'm on a remote site and call to a cell phone (even next to me, just to see how it works) - the voice quality becomes close to unbearable. When I make similar call from LAN, where FS resides, the quality of audio is noticeably better (still not perfect though). That shows me how important additional hop could be when I use FreeSWITCH as a proxy for audio channel.
With this experience in mind, I don't have any doubt choosing toll-free SIP provider, that supports direct RTP connection, over one, that require FreeSWITCH (or any other SIP server) as a proxy in the middle. And, BTW, why it's even a question, when there is a plenty of offers, not asking for a static IP?
--
Keep it simple, it'll become complex by itself... |
|
| reply to Trimline
Trimline,
Will you please contact sales@teleinx.com regarding the 4 hr call limit. We will raise that for you. |
|
TrimlinePremium
join:2004-10-24
Windermere, FL | Will do... thanks! |
|
