similar to: Psst... Top secret information: Codename Pineapple

Friends, I have gotten a few questions lately on the status on the Codename Pineapple project, the project that hopefully will produce a more stable and SIP compliant SIP stack for Asterisk. Due to lack of funding, it's postponed until further notice. I have a few sponsors, but not enough to be able to dedicate time to work on it. And since Digium hasn't made up their minds after

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk "just" as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone

Friends, Beginning next week, I will travel around Europe to teach Asterisk - the one day Meet Asterisk training. MeetAsterisk is organized by Edvina in cooperation with Digium and Voop. In many places, local Asterisk equipment resellers participate and show their equipment. This is the tour plan: * Amsterdam April 26 * Copenhagen April 27 * Oslo April 28 * Paris May 3 * Brussels May 4 *

Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload

Hello, I am altering an asterisk configuration and would like to eliminate the loading of modules I do not want or do not need at the moment. For example I am do not want to use chan_zap (I'm using chan_capi) and don't want to be bothered with music on hold at the moment. Is there a way to configure these things off so asterisk doesn't try to load them? Or do I have to just

Hi, I have a data set look like this: ================================= Fruit Quty apple 20 banana 10 orange 17 apple 30 apple 15 orange 26 banana 15 .........and so on .......... ================================= The level of fruit is 30, that is, there are 30 different fruits. I'd like to compute some simple statistics for each different fruit and get output like this:

The Asterisk Developer Team is proud to announce the Asterisk SPE v1.0 Beta Release for immediate download on tftp.digium.com. The SPE has been developed as a joint project between Digium, the Asterisk Company, Voop, the European Asterisk Dialtone provider and the Asterisk community. The Asterisk Service Provider Edition is focused on the needs for the new breed of Telecom companies - the

WinXP-x32, R-21.13.0 Dear list, I have a problem that (I think) relates to the interaction between Windows and R. I am trying to scrape a table with data on the Hawai'ian Islands, This is my code: library(XML) u <- "http://en.wikipedia.org/wiki/Hawaii" tables <- readHTMLTable(u) Islands <- tables[[5]] The output is (first set of columns):

Hi, is it possible to use asterisk to serve two SIP-domains with different users? It does work to define two domains with 'domain=' in sip.conf, but that allows all sip-users to register with both domains. I want to define users for a one domain only and not allow them to use the second. Lets say I have domains domainA.org and domainB.org and a sip-user user1. User1 should be able to

There was a flurry of "Vonage is going to unlock SIP" activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you

So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt

* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND VIDEO TO MICROBLOGGING! In a surprising move, Digium in partnership with Edvina today released a new channel driver for Asterisk, chan_tweet. The driver connects seamlessly to several microblogging platforms, including Twitter, Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of this new module is to

A friend of mine just sent me this email - he is looking for an IAX inbound service in Holland - any thoughts? Voip info only has Nadiz which seems to be SIP only. Hi Dean, I need a Dutch number with IAX support. Do you have any leads in that direction? It's been difficult for me to figure it out -- especially since most of their sites seem to be in Dutch... Regards,

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

Does anyone know enough about the implementation of AstDB to know whether the data structure is a Hash function, a Balanced-Tree, a b-Tree, or a Linked List? I'm trying to estimate the lookup 'cost' of a AstDB with around 160,000 keys? Obviously I already know that it WILL WORK, but the question is whether the data structure is optimal in the Berkeley DB AS IMPLEMENTED in Asterisk.

Hi All, I read some discussions about the new SIP authentication methods for 1.6.X branches and possible addition of new type of user, type=trunk. I'm wondering about the disposition about this. Will it be added? In 1.2 and 1.4 branch, a SIP invite was first checked for a valid [user] then a valid host=ip, then if not present send call to [general] context=incoming. In 1.6, a SIP invite

On Thu, Sep 13, 2007 at 04:32:27PM -0400, Jay R. Ashworth wrote: > I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6 > stations, mostly IP (I'm looking at the Grandstream 201, to start), and > maybe X-lite on a couple of laptops via VPN. > > We've got a 4xFXO box we bought off eBay, which unfortunately I can't > find to quote a model number off