similar to: Verizon DTMF Recognition

One would assume they have better things to do as they are quite busy. I think this is just a proactive measure meaning they say you cannot do it upfront so that in the event of a problem, it was predeclared. As to the rest of the TOS, I could be wrong but I got the impression that the owner of VoipJet speaks English as a second language due to some email exchanges. If that is the case, the TOS

For outbound only, I have traditionally recommended VoipJet. They just recently has a spat of issues that seem to have resolved though. I am still using them via their east coast server and it seems to work quite well again. Cost is around 1.3 cents minute I believe. Use IAX and g711 for best quality to VoipJet. Thanks, Wiley -----Original Message----- From:

So far my experience with TOS has been that most of them are pretty odd. No one wants the liability of a stock trade gone foul or a call to the doctor that gets disconnected. Essentially, those things in the TOS are just a CYA. They are un-enforced but should someone decide to attempt to sue based upon a financial loss, the ITSP is covered. So, yep. That is weird but not unexpected. Heaven

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

I have a question for those of you out there using VoicePulseConnect for incoming did I have in my Realtime extensions Database (the x's are replaced with my phone number) context = voicepulse-in-01 exten = xxxxxxxxxx/ Priority=1 app=NoOp appdata = Incoming call with no callerid on xxxxxxxxxx However it never triggers I also tried using one of my other providers (voipjet for outbound) and

If you asked this question on the biz list you would get a lot of people that will tell you that they offer services where you can set the caller ID to what ever you want. To name a few:: Nufone Teliax Voipjet ----- Original Message ----- From: "Doug Crompton" <doug@crompton.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks! G. ----- Original Message ----- From: <asterisk-users-request@lists.digium.com>

I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because DTMF is not understood. If I dial from a land line I can get in with no problems. Any tweaks recommended

We are looking for SIP trunks for our * pbx for our business. Being able to port our numbers is an absolute requirement. teliax can do it, but I am unsure of the others. Anyone have experiences (good, bad) with the above mentioned providers to share? Eg reliability, quality, etc. -Dan

Hello All, Right now I have several providers. Voipjet, Teliax, and more recently Broadvoice. Broadvoice gives me unlimited to europe, but what I want to do is determine the best way to setup a dialplan so for example, certain countries will go through the cheapest route. I am really only interested in Poland, Russia and Turkey. Poland is free on broadvoice, but not for cellular, which I may

I have a macro and when I call it I have something like this: exten => s,1,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)}) exten => s,n,Set(GROUP()=MYGROUP) ;Set Group exten => s,n,NoOp(Group List: ${GROUP_LIST()}) exten => s,n,NoOp(Group Count: ${GROUP_COUNT(MYGROUP)}) The GROUP_COUNT returns zero before the call to GROUP but also returns 0 after the call to GROUP. If I

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

Is it possible to turn off DTMF recognition (and all transfer services etc.) pending on CallerID (or FXS channel)? Some of the FXS channels I will setup soon, is going to work exactly like POTS. It will be used by people not knowing their within Asterisk. They will be pretty confused when "Transfer" is playbacked in the handset. :)

Hello, I am experiencing a strange problem and I am wondering if anyone may have some pointers as to how to overcome it. I have an account with VoipTalk here in the UK which I have connected to my Asterisk server. VoipTalk supports IAX2 and SIP and I have connected to my Asterisk box using both methods. The problem is when I dial into my Asterisk box via my VoipTalk incoming PSTN phone

Hello everyone, I have problem with DTMF recognition when calling from PSTN, my Asterisk box won't read DTMF tone at all. I've tried use cellphone, normal telephone and voip lines, nothing worked. softphone to softphone within extensions are ok. I'm a newbie at this, can anyone point me out where to look? I'd really appreciated. Thanks a lot Nate -------------- next part

I tried this with no luck on the machine, seems the fax did not like even ulaw. Ended up just putting it inline, but of course, in a multi-line setup it could be a problem. Just a word of advice, get another line for it if you can. I fought for about a year with my pingtel ip phone over a wireless connection, now I got serious and got a normal line about 5 months ago. Much better off now...

Hi, is there a possibility to tell zaptel or Asterisk to modify the DTMF sensibility? The problem what i have is that the Asterisk don't get all Numbers which the analog-FAX dial, let say the FAX dial 123456789 the Asterisk get to number 24679. I think that can be to DTMF Tone duration or the Frequenzy. so you got yna idea what it could be? Thx for helping me. Bye Timm

Hi, I am using a channel bank and zaptel hardware. I have a credit card machine on one of the channels that appears to be dialing "too soon" for asterisk, every complete number recognized by asterisk is missing the first 1-4 numbers. This is a serious problem for me, anyone have any ideas on whats going on? The pstn picks up on the dtmf tones just fine. I was able to get it to

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex