similar to: Where did my DID's go??

I have been looking around for VoIP providers but have not found a good listing. Is there no "yellow pages" for VoIP providers? Google mostly returns services like Vonage, Packet8, NuFone, ect. None seam to be very reseller friendly and none offer LNP or local DID's for my area. Anyone know of a list (even a partial one) Jeromie Reeves

I found some old messages regarding a possible pkt8 DTA "bypass". Anyone is using Packet8 with Asterisk? ========== http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat Got Softphone Working with Packet8 Friendly name: {Anything you'd like.} SIP domain: packet8.net SIP proxy: packet8.net Leave everything else at the default. When you login, it will ask for a username

I received a message from Packet8 last night telling me they added some new area codes. I took a look at their new area code/rate center finder page and it looks like the added a LOT of new area codes. They now have phone numbers available in almost every state. The URL for finding out which area codes and rate centers are available with Packet8 is at http://www.packet8.net/about/areacodes.asp

Looks like www.packet8.com <http://www.packet8.com/> has been hacked :-( The phone service is offline as well. Interesting to note that the whois is showing an update today but doesn't look like details have changed. Domain Name: PACKET8.NET Registrar: REGISTER.COM, INC. Whois Server: whois.register.com Referral URL: http://www.register.com Name Server:

Hi John, I've been using Packet8 via a physical ATA and XP100 card for some time. As far as I know it is not possible to connect to the Packet8 service without the ATA. If this is not the case I would be very interested to hear this. In addition since moving to the USA I now only have a single packet8 line into my asterisk box (I used to have this and a 2nd regular pstn line) I used to be

Noob here, my apologies in advance. Recently my employer decide to stop paying for my home POTS line, so I ordered packet8 for a home line instead of another POTS line, since I really dislike my local phone company, and the POTS line without any long distance would cost more than the $20 to packet8 with unlimited US calling. Anyway, this whole thing got me started thinking about VOIP a lot

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems

I have packet8 and I have spent many hours on the phone with them. If someone has found away around there DTA configuration I would like to know so I can bring it in house to my * box. But as far as your question is concerned. No. Not that I know of. They wouldn't give me any information about the configs. .o-------------------------------------------------------o. Brian Fertig Network

**** DRAFT **** DRAFT **** DRAFT **** DRAFT **** I've been using the DTA310 from Packet8.net for a couple of weeks. The DTA310 is about $130 without the Packet8.net VoIP service. It only supports SIP. On the back of the DTA310 is a power connector (power supply is provided with the product), a 10/100 Ethernet port, an FXS port, and a reset button. The front of the device has LEDs for

I am aware of a least a few people (including me) who were using the Packet8 service along with Asterisk for outgoing calls. Last night Packet8 did a software upgrade and both last night and this morning I have been unable to make any outgoing calls. Has anyone else noticed this behavior and/or been able to correct it? I get Got SIP response 403 "Forbidden" back from 4.42.235.170 when

Come to think of it since the DTA310 uses DNS to find the SIP server, you could setup a DNS cache and override the DNS entry for what packet8 uses (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your own SIP server? Kind of a hack but it should work as long as it's running on port 15062. I am very new to this so I don't know if there's a port standard for SIP

Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX

I need 100 DID's in a specific rate center (916-854-xxxx). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas

Apologies if this is an FAQ, I wasn't able to find an answer googling: Will any of the public PSTN/VoIP gateway services (Vonage, Packet8 etc) interoperate with * ? I'd like to deploy a box which provides PBX service for analog handsets, and handles inbound/outbound calls via both analog PSTN lines, and, say Packet8 VoIP service. I understand that I can do this by connecting the analog

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from

Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up

Has anyone gotten Packet8 setup as a sip trunk for Asterisk? Would they be so kind as to share their config or point me to a website that shows how it can be done? I would be much obliged. Grady -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060621/311435fd/attachment.htm

Does anyone know of any provider(s) that can provide DID's for the Czech Republic? Regards, -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040802/4885836b/attachment.htm

I'm looking for a company that offers Guatemala DID's. I saw that Lingo does, but Lingo isn't easily compatible w/ Asterisk, so they're a last resort. Thanks in advanced, Phil Astin.