similar to: outgoing calls with packet8 and dta310 problems

**** 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 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,

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

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

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

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

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

Sorry for the misspelling... Thanks for the replies. I will set it up and start playing. This is all very exciting. I've been using VoIP as my primary phone but this is going a bit further. At the office we have a T1 that is probably fairly dead after hours. Supporting 5-10 users should be fine I'd imagine. I've read 1 VoIP connection uses about 64kbps or 8KB/s? So...

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

Thanks for that. I just got rid of packet 8 and went with 100% asterisk in my house. But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But would like to have an extra FXS laying around just in case.. .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From:

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

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of

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

Hello I change this from chan_sip.conf (see ipsvr): static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey) { char port[10]; char ipaddr[20]; char regseconds[20]; time_t nowtime; -> char ipsvr[20]; time(&nowtime); nowtime += expirey;

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

The past week or so I have been experimenting with Asterisk and overall find it to be a nice software suite, although I have encountered some problems, and have found almost no documentation (For example in sip.conf I needed the commands fromuser= and fromdomain= and only figured out this was possible after spending a few hours browsing on the internet and reviewing some person's configuration

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.

Hello Users I have one VoIP service from Packet8 ( SIP protocal ) Packet8-----> Astetisk server ---------> My SIP agents My Sip Agents are in Asterisk Server , I configured.. If any one user in My Asterisk has to Call the Packet8 service providers , How can I configure it. Till now I'm Doing on OpenSER and ASterisk (Voicemail and Confereing ..) But My Asterisk has to connect