similar to: Total newbie here looking to do a VoIP confe rence call?

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 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

Patrick hi. Asterisk can do that, and you don't need VOIP lines. If you connect Asterisk to the net, and all employees have a VOIP phone (either hardware or software) then you're good to go. What do you need? To begin with, install linux on an old pc (well, not too old). Then go to voip-info.org and take a look at the Asterisk wiki. Everything you need is there. And of course, we're

**** 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

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

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:

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 a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to

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

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

Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, AT&T, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability &

I thought this might interest a few asterisk users. I don't use them so I have no idea about Adtran's but I know a heap of people on this list swear by them. Cheers, Dean ________________________________ From: vclass.coordinator@adtran.com [mailto:vclass.coordinator_adtran.com@mail.vresp.com] Sent: Monday, 19 September 2005 9:59 AM Subject: ADTRAN Virtual Classes: Ensuring

Has anyone put together a list of basic modifications & recommended settings to optimize a TCP/IP (mostly wireless)network for VOIP? Specifically G.711 & IAX protocol, and the Vonage and Packet8 services. My network is mostly StarOS (which uses linux cbq and ipchains) and Mikrotik and I'd like to put in some tweaks to get VOIP working better. It is working just fine right now, with

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

Hello! I would like to get connected with a VoIP provider for home. At some point, I'm sure I will be connecting to it via an Asterisk box, but for now, I will be using whatever hardware they provide. What recomendations do you in the Asterisk community have for a reliable VoIP service that will hopefully interoperate with Asterisk? A company that is actually willing to work with an

Suggestions on a VoIP provider that works with *? I know there is vonage.com but have read numerous bad things about their agreement and that you don't get a username and password from them rather a preconfigured Cisco ATA 186 that you are not allowed to get in to. Top most being you have to use a X100P with them to work with * which in my mind is just too many links in that chain. I have

I realize this is a shot in the dark, but I'm trying to find a VoIP provider that offers 402 or 712 area code DID numbers. I'm almost completely convinced that no one offers these area codes (eastern Nebraska, western Iowa), however considering the wide audience of this mailing list I thought this would be a good place to ask. I would prefer a provider that allows for Asterisk use, but I

I am concidering taking the Jump to Linux. I have SuSE. 6.3 at the moment.. but don't want to loose some of my M.$. Apps.. So I hear about wine.. as being able to run some of the apps.. this is great.. News.. for the Weary anyway.. So does this Wine work with SuSE.. Please forgive me I haven't even played with Linux. yet.. just trying to get all my questions sorta answered before I wage