 | Reliable Outgoing Toll Free Calls without Reg I have searched high and low for reliable toll free voip providers but I have found only Future-Nine to be reliable. I need a backup to Future-Nine and would appreciate if anyone could point me in the right direction. |
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 | »www.voip-info.org/wiki/view/Toll···roviders
IMO, sip.tollfreegateway.com and proxy.ideasip.com are reliable. |
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 Reviews:
·Anveo
| reply to ThaiGuy I using the following as backup
»www.alcazarnetworks.com/wholesal···freeterm
They also seem to send CallerID reliably for me whereas those sent from Future9 or CallWithUs would be incorrect. |
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 garys_2kPremium join:2004-05-07 Farmington, MI | reply to ThaiGuy Does F9 support calls to 855 now? |
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 | reply to ThaiGuy I did some quick tests with each of the providers mentioned on the wiki that Stewart linked to above. I started a few months back and tested at different times of the day, but not extensively.
Alcazar seemed the most reliable but I tried making calls through them the other day and they couldn't connect any of them. I notice there are a few new players on the list, so I'll give them a go. |
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 | Done some more testing and tf.callwithus is the fastest to connect. Having problems with one-way audio and Alcazar, so I have dropped them from my tests. Also dropped Denetron and ideasip as they are troublesome.
callwithus, tollfreetollfree, tollfreegateway, tollfree4free, future-nine and arctele have yet to fail to connect. |
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 | reply to scanchain Can anyone explain why Future-Nine and CallWithUs might not be able to deliver CallerID correctly? |
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 nitzanPremium,VIP join:2008-02-27 kudos:2 | 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. |
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 | Thanks Nitzan. I also tested with RTP Packet size set at 0.010 and it worked well. You previously advised I should use 0.020, so it OK to change or not? |
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 garys_2kPremium join:2004-05-07 Farmington, MI Reviews:
·Callcentric
·Future Nine Corp..
| 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. Nitzan, I route my TF through you. Can you accept 855 calls or should I still send them elsewhere? |
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 TrimlinePremium join:2004-10-24 Windermere, FL Reviews:
·voip.ms
·Callcentric
·RoadRunner Cable
| reply to ThaiGuy said by ThaiGuy:I have searched high and low for reliable toll free voip providers but I have found only Future-Nine to be reliable. I need a backup to Future-Nine and would appreciate if anyone could point me in the right direction. I've found to be quite reliable : »www.tollfreesipgateway.com/Home.html
They will disconnect you after 4 hours, but other than that, they do pass your caller ID and support G722 (even though they don't say so). |
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 gweidenh join:2002-05-18 Houston, TX kudos:1 | said by Trimline:...and support G722 (even though they don't say so). You've sparked my attention. How exactly are they supporting G.722 via the PSTN? |
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 TrimlinePremium join:2004-10-24 Windermere, FL Reviews:
·voip.ms
·Callcentric
·RoadRunner Cable
| said by gweidenh:said by Trimline:...and support G722 (even though they don't say so). You've sparked my attention. How exactly are they supporting G.722 via the PSTN? Good question.... most of the dial in 800 conference bridges now support G722. Since my comrades are in EMEA, it's an instant recognition of that codec. It sounds like they are in the room with me, and visa-versa. The sound quality is amazing. |
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 gweidenh join:2002-05-18 Houston, TX kudos:1 | Well now you really have me interested. Can you provide the name of the conference service you are using? |
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 TrimlinePremium join:2004-10-24 Windermere, FL | at&t - an international bridge. 877-888-4443 |
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 | reply to ThaiGuy said by ThaiGuy:Thanks Nitzan. I also tested with RTP Packet size set at 0.010 and it worked well. You previously advised I should use 0.020, so it OK to change or not? I don't recommend deviating from the standard .02 . First it shouldn't affect the call quality you hear since this setting is only for the transmit audio. Second, it creates an asymmetric call since the proxy sends back the return audio at the standard .02 where you don't have any control over it. |
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 | reply to ThaiGuy
Re: other's comments about caller id delivery tf.callwithus.com delivers caller id if 10 digits are sent from your VoIP setup. There was a concern but that seems to be with just the pickier numbers like 800-444-4444. Most numbers are getting proper caller id. |
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 nitzanPremium,VIP join:2008-02-27 kudos:2 | reply to garys_2k
Re: Reliable Outgoing Toll Free Calls without Reg 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. |
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 | 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. Don't know about anyone else, but I have found that is NOT true, at least in my testing this morning. Asterisk formats outgoing Caller ID as Caller Name (where Caller Name and 1234567890 are replaced by the actual name and number), at least in the CLI, but when that call hits your switch that number gets replaced by something else. However, I just tested Alcazar and it does pass the Caller ID properly. Don't know why they're able to pass it and you're not, but that seems to be the case, and I am definitely passing only ten digits for the number.
Does anyone other than Alcazar pass the actual Caller ID? Would be nice to have a backup. |
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 Reviews:
·Anveo
| said by WhyADuck:Don't know about anyone else, but I have found that is NOT true, at least in my testing this morning. Asterisk formats outgoing Caller ID as Caller Name (where Caller Name and 1234567890 are replaced by the actual name and number), at least in the CLI, but when that call hits your switch that number gets replaced by something else. However, I just tested Alcazar and it does pass the Caller ID properly. Don't know why they're able to pass it and you're not, but that seems to be the case, and I am definitely passing only ten digits for the number.
Does anyone other than Alcazar pass the actual Caller ID? Would be nice to have a backup. That's my experience as well. Tested with 18004444444 and Alcazar consistently passes the CallerID correctly, but not F9. In both cases, I have set the AuthID to 10 digits. |
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