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Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier. Thanks, Walker -- ******** DataCrest, Inc. -- Technically Superior ****************** Walker Haddock...

Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is asterisk@home 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833.

Hi List, I'm having a problem with detecting incoming dtmf tones with chan_capi, using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0, expecting that the capi module will detect the tones, but it did not. I also set both to 1, expecting that the asterisk dsp functions will detect them but it did not either. Can anyone provide any ideas how to overcome this issue?

Helo List, Sorry I missed the rest of my email in my previous post. Please see below. I'm having an issue using the AND (&) operator evaluation in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in two diferent contexts in the dial plan.

Hi Can someone please help with the following, We are using asterisk@home 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice

I have a problem receving calls via the ISDN line, using the followin components Asterisk 1.0.9 with asterisk@home chan_capi-cm-0.6 AVM Fritz card datalink protocol = point to multimode I can make calls out with no problems so the issue is only incoming calls. When I make the call from an external line to the ISDN line connected to asterisk, I get a busy signal after about 5 seconds. I have

Helo List, I'm having an issue using the AND (&) operator in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in my dial plan. When a caller presses 1, yes is passed as a varialble and same when 2 is pressed a "no" is passed.

Hi List, Can any one please let me know how to pass arguments to the agi script from the dialplan? I read that it is possible to pass arguments to an AGI script here, http://home.cogeco.ca/~camstuff/agi.html, by entering the variable followed by a vertical bar but it doesn't seem to work for me. I'm using a basic AGI script to query a database and then returns to specific contexts

Hi List, I am experiencing an issue in a server that I have installed asterisk; configured an AVM FRITZ card to work with the capi module. Once istalled the card works perfect; however every time I reboot the machine I found that I have to re install the capi4k-utils before I can load asterisk otherwise the capi module will not loadup. Can anyone direct me in the right direction in order to

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone