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Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

...05 10:03:49 +0100 (CET) From: Remco Barende <asterisk@barendse.to> Subject: Re: [Asterisk-Users] France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo To: Asterisk Users List <asterisk-users@lists.digium.com> Message-ID: <Pine.LNX.4.61.0501181003290.16971@raveon.vaag.nu> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Mon, 17 Jan 2005, Wilson Pickett wrote: >> Can you offer any clue where I would need to look, I guess its not in >> extensions.conf that is the problem? > > 1) Are you registering with proxy1? no, from the...

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files