similar to: Direct connection to Packet8 without DTA

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

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

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:

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

I know someone mentioned doing this once before however I can't find it. Anyone remember if or how it was successful? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031223/003bf8ad/attachment.htm

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

I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware version (Application Code Version: DTA version 1.0 US (8x8 001111)) onto it via TFTP, so I could access the SIP configuration. Under the SIP config, I put the IP of my * system, the 5060 port, and for Domain Name, I put default (is that right?). I checked off the Send Registration Request box. Dial Plan I left at the default,

**** 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'm trying to get asterisk working w/ packet8 (incoming and outgoing) and a dta310 so I can have more control over voicemail. I've looked at the data stream coming from the dta310 and from packet8, but I haven't managed to get the phone to actually place invites through asterisk. On the asterisk end with chan_oss.so, I can make it dial and I hear ringing and the first second of

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

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

Does anyone know of an ATA which has a modifieable (open source) firmware AND that supports faxing. Granted most of them don't do faxing well but that is more of a problem with the network side. Michael Blood -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/8c13db6b/attachment.htm

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

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

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

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

We're on track for a release of Asterisk 0.4.0 soon. I'd like to try to see to it that we have squared away our SIP implementation by then, and after that point, try to keep it in tip top shape. In general, I find that SIP is extremely fragile, and every time I try to fix one bug, I end up creating another somewhere. What I need are strategies for verifying that the SIP implementation

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