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Forums » VOIP etc » Voice Over IP - VOIP » VOIP Tech Chat » [Other] Flat rate US calls
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You can plotz with POTS. »
« Speedtouch 780 voip configuring Dial Plan settings  
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Test99
Premium
join:2003-04-24
San Jose, CA
·DSL EXTREME
·InPhonex

reply to ccowling
Re: [Other] Flat rate US calls

said by ccowling See Profile :

I've had that done now.
Changes don't get made much faster than that. Must be a record.

Test99
Premium
join:2003-04-24
San Jose, CA
·DSL EXTREME
·InPhonex

reply to ccowling
said by ccowling See Profile :

Do most providers allow you to send any CLI or is it only of numbers you own?
Some providers let you set the outgoing caller ID. CallWithUs and Teliax come to mind.

See this post: »Re: [General] Any service with this feature set? The legislation mentioned there has not become law. My guess is that it won't, at least not this year.

ccowling

join:2007-11-14
reply to ccowling
Ah ok.

We're more then happy to set the CID as a users home number, etc. The legislation is interesting though.

Chris.

wierdo

join:2001-02-16
Tulsa, OK
·Future Nine Corpor..
·Teliax VOIP

said by ccowling See Profile :

We're more then happy to set the CID as a users home number, etc. The legislation is interesting though.
If by interesting you mean idiotic. When I first had the capability, I would occasionally send odd caller ID..the novelty wore off in about 10 minutes.

We do make extensive use of that ability, though. Each user of our system gets their caller ID set appropriately when they dial in, and we have some DIDs that forward to cell phones, and we forward the caller ID through.

Only one of our three (at the moment) termination providers has any restriction on it. We have an arrangement with the other that allows us to set any caller ID from a predetermined list. We don't use that one much, as I haven't gotten around to programming the necessary logic to allow its use when someone on that list is placing the call.

Basically, I use it for personal calls and for certain destinations where our other carriers are either too expensive, don't pass DTMF, or are otherwise unsuitable.

It always amazes me that these days Caller ID can show up correctly on phones in nearly all the countries we call to or from.

Even at home I find it handy, so I can set the caller ID to the appropriate person's direct number, although that list is much smaller than the one at work.
--
It's wierdo, not weirdo. Yes, I know that's not the 'proper' spelling of the similar english language word.

BruceN
Hi

join:2006-11-17
Roswell, GA
reply to ccowling
Welcome to this great place!

Bruce Nicklin Voice Stick and MyGlobalTalk

dalrun

join:2008-01-09
Bellingham, WA
reply to ccowling
Has anyone actually made outgoing calls (using sip protocol) with the $5 credit account? Audio is fine when I use the test numbers. Outgoing calls connect but at no point is their any audio on either end.


deheza

join:2004-06-10
Allen, TX
·QuantumVoice


1 edit
  Using IAX with Idefisk works for me, but have tried a lot of combinations with SIP and can't get any audio. Audio doesn't come through with Sjphone or my SPA2102 with or without DMZ enabled on my WRT54G-TM router running Tomato. Calls on my system work great using SIP on CallWithus, Future-Nine, NuFone, QuantumVoice, TutTele and Callcentric. FaktorTel has a SIP audio problem in my opinion. The SIP calls answer and charge the account, but no audio either direction.
--
RoadRunner, QuantumVoice, Future-Nine, Vitelity, CallWithUs, TutTele, NuFone, Stanaphone


Yippz Voip

join:2005-05-04
Atlanta, GA


2 edits
reply to dalrun
Exactly the problem I've been having. The call is running but there is no audio.

I've been in email contact with Chris but I still can't get audio to work. We thought it could be a firewall issue on my end, but it clearly isn't as we can make calls with audio using 2 other SIP providers we're testing.

Anyone else besides Dalrun, Deheza and I that are having the audio issue? BTW, this is using SIP, not IAX.

Also, I did make note that 1-800 calls and 123 echo test were charging the .10c rate -- Chris told me that is being fixed.

dalrun

join:2008-01-09
Bellingham, WA
My sip phone has a fixed public IP address, so I know its not a firewall issue. I have yet to have audio problems with any other provider. My bet is that all sip users have no audio.

atrevete

join:2008-02-24
No audio here either using a Linksys PAP2. FreeDigits, VBuzzer, etc., continue to work flawlessly using the same PAP2.

cbrain

join:2000-05-21
Silver Spring, MD
·VoicePulse for Bus..
·Verizon FIOS
·Comcast
·Future Nine Corpor..
·VoiceStick

reply to ccowling
I signed up yesterday. It was simple, fast, no credit card info and I got the credentials immediately. I entered the basic registration string for an IAX2 trunk in Asterisk and was registered. This afternoon I google settings and finished my trunk setup. The first call worked. I made several more and all were good and the CID from my calling extension is sent by default. I made several test calls to my IPKall line and heard about 30 seconds of DTMF tones from both ends then one way audio with high latency. All other calls so far are good and calls to IPKall from cell work. I looked at the web "my account" and set "Account Codecs / Protocols" to match my Asterisk settings but it made no difference.

I won't be able to play until tomorrow. Any ideas of why this one problem? IPKall is set to directly forward by SIP/URI to my Asterisk box and has always worked well. Could it be an IAX2 to SIP problem?

PS: Just had another thought. I made a call from an Asterisk extension to IPKall. It went VoicePulse Connect. No DTMF's but one way audio. I never had an audio problem with either provider. Must have something to do with originating an termination at the same box.

jay_rm

join:2002-04-12
Netville
·Fox Valley Internet
·ViaTalk

reply to RockyBB
said by RockyBB See Profile :

IMO excessive consumption of Fosters and Vegemite lead to bad business decisions
I don't think anybody in Oz actually drinks Fosters - make it a VB, mate !
--
3500/512 5.7 GHz Motorola Canopy Wireless; FoxValley.net
"Peace through superior firepower"

jester121
Premium
join:2003-08-09
Lake Zurich, IL
·surpasshosting
·ViaTalk

said by jay_rm See Profile :

I don't think anybody in Oz actually drinks Fosters - make it a VB, mate !
Oh, HELL yes. I drank so much VB in a 3 week period that my girlfriend was threatening to drop me off at detox on the way home from the airport.

ccowling

join:2007-11-14

reply to ccowling
quote:
FaktorTel has a SIP audio problem in my opinion

We're checking our firewalls on our side at the moment, as it's possible were blocking the incoming audio stream. :-/ There is always a fine line between useability and security.

I'll get back to you guys asap (today).

Chris.


dot_null
Premium
join:2004-06-28
Kennesaw, GA
·Callcentric
·Comcast
·VoiceStick
·AT&T Southeast

reply to ccowling
Am I reading correctly on the main signup page that we get a free DID upon signing up? If so, how would one go about adding a DID to their account? Thanks!
--
A little off center & loving it! »www.euphoricarythmia.com/

ccowling

join:2007-11-14

reply to ccowling
quote:
Am I reading correctly on the main signup page that we get a free DID upon signing up?

Where is this and which page? As it is included on the non $0 packages, but not the $0 p.mo package. Something to definately fix. (Sorry)

Chris.

ccowling

join:2007-11-14

reply to ccowling
Ok, Wow.. I just checked my PM list and apparently I have 33 messages waiting.

Please bare with me as I check your messages and get back to you all. I really do appreciate all of the testing / calls you guys and gals are putting through and the reports back of things which may or may not be working correctly. Such as the purchase of a DID etc.

- Note we're currently doing some kernel upgrades on the servers so they may be off line for a bit as part of making sure everything is rock solid.

Chris.

venk25

join:2004-05-12
Nashua, NH
·VoicePulse

Hi Chris,

Apart from the SIP voice issue, I also noticed that calls to some of your network/test numbers (like 600) are being charged too (meaning, money deducted from the $5 credit).

For the Outgoing CID/CLI, I'd prefer to have my current home phone number go out rather than 09xxxxxx number. Let me know if you want me to PM you or if it'd be enabled in the web interface.

Cheers !


dot_null
Premium
join:2004-06-28
Kennesaw, GA
·Callcentric
·Comcast
·VoiceStick
·AT&T Southeast

reply to ccowling
said by ccowling See Profile :

quote:
Am I reading correctly on the main signup page that we get a free DID upon signing up?

Where is this and which page? As it is included on the non $0 packages, but not the $0 p.mo package. Something to definately fix. (Sorry)

Chris.
Haha, I thought it sounded a little too good to be true. It's toward the bottom of this page: »www.faktortel.com/products-freeoffer1.shtml

I did notice you have very fair DID prices - I saw some as low as ~$1.60/month! Is this for unlimited usage, or billed per minute?
--
A little off center & loving it! »www.euphoricarythmia.com/

Test99
Premium
join:2003-04-24
San Jose, CA
·DSL EXTREME
·InPhonex

reply to ccowling
said by ccowling See Profile :

we're currently doing some kernel upgrades on the servers so they may be off line for a bit
Servers are back up, but calls have no audio.
-
Forums » VOIP etc » Voice Over IP - VOIP » VOIP Tech ChatYou can plotz with POTS. »
« Speedtouch 780 voip configuring Dial Plan settings  
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