similar to: Have Voice Mail tell the extension?

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

How do I do this 1) ZAP-> * -> IAX(1) ------> IAX(2) -----> DG104S ------> Handset 2) No Answer on Handset 3) Back to IAX(1) 4) IAX(1) tries a cell phone 5) Still no Answer 6) Local * Voicemail. I have 1 working, and I had 4 working when there was only one box, i.e. when the handset did not answer the DG, asterisk went to the next step. Now that I have step 1 going to another

For those of you that are in the Portland, Oregon area: I am giving a talk today on Asterisk at the PLUG Advanced Topics Meeting. Details below. JT >From: "Zot O'Connor" <zot@whiteknighthackers.com> >To: PLUG LIST <plug@lists.pdxlinux.org>, > PLUG Announcement List <plug-announce@pdxLinux.org> >Organization: White Knight Hacklers >Subject:

I am installing a used DG104S.... I got it to ring from gnophone, but all I got was fast busies. so I upgraded based on Pavel's link: ftp://ftp.dlink.ru/pub/VoIP/DG-104S/Firmware/MGCPDG104.zip So I now have: PROM Version: 3.0B22-D RUNTIME Version: 3.0B44-D But when I pick the phone up I get: ggdbg>000001604 DIM: 0 DSP ERROR: Reason= DIM ERROR: State Timeout 000001604 DIM:

Is there any reference for the dg-104 telnet(shell) I need to log into a remote unit and reboot it over telnet. Its shell is not clear. David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 287 bytes Desc: not available Url :

It seems that the DG "gets lost" and keeps attempting to send RTP packs to asterisk and it get an icmp deny. The phones on that port will not work. Other phones do. So is this asterisk failing to hang up on the DG, or is DG not seeing a "call over" message? It is happening more frequently, but I am not at the location, so it is tricky to catch the streams live. Whenever I

I once (for a brief period) had dialtone, but I do know why :) Otherwise I get a boooop-booop-booop sequence. I cannot tell if this is the D-Link doing this, or asterisk... Who should be giving solid US dialtone? My indication.conf says: [general] country=us ... [us] description = United States / North America ringcadance = 2000,4000

I would like to write data so that I can re-read it and reproduce results, preferably in both R and Splus. In the past when I have done this my data has been relatively simple and I've just scan()ed it. Now I have a fairly complicated structure I would like to preserve and I've been trying to use dput and dget. Is there a better way? If not, the following truncation by dput, which I

Dear R users, I have a data frame with categorical Vars. "Groups" and a couple columns of numeric Vars. I am trying to make two-sample t.test on each variable(s01-s03) by Groups. A data generated as following: zot <- data.frame(Groups=rep(letters[1:2], each=4), s01=rnorm(8), s02=rnorm(8), s03=rnorm(8)) I have written a piece with a for loop. for (i in 1:(length(zot)-1)) {

Hello All, I wrote a PHP function to parse the icecast stats log and return as much information in a structured array as possible (well, as much as I care about, anyway). Because the stats log is not easily machine-parseable, I thought this might be useful to somebody. The log is parsed using Perl-compatible regexps, so it should easily port. I have only my own setup to test this on, so

The postscript driver in R-devel seems to choke on box(col=0). This is not a problem in R-1.3.1. R : Copyright 2001, The R Development Core Team Version 1.4.0 Under development (unstable) (2001-09-22) ... Type `q()' to quit R. > plot(1:10, 1:10) > box(col=0) > postscript(file="zot.ps") > plot(1:10, 1:10) > box(col=0) Error in box(col = 0) : invalid value specified

OK - finished some tests over tcp and ironed out a lot of problems. rdma is next; should be snap now.... [I must admit that this is my 1st foray into the land of IB, so some of the following may be obvious to a non-naive admin..] except that while I can create and start the volume with rdma as transport: ================================== root at pbs3:~ 622 $ gluster volume info glrdma

Hello, I would like to access my data frame without one variable. E.g.: > colnames(x) [1] "Besch" "Ang.m" "Arb.m" "i10" "Umsatz" "arbstd" I can try x[,-1], but this variable must be called by it??s name. x[,-"Besch"] x[,!"Besch"] attach(x) x[-Besch] ... ... does not work. I could not found a solution of

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

We're considering implementing gluster for a genomics cluster, and it seems to have some theoretical advantages that so far seem to have been borne out in some limited testing, mod some odd problems with an inability to delete dir trees. I'm about to test with the latest beta that was promised to clear up these bugs, but as I'm doing that, answers to these Qs would be

The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort

Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong?

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

I've just finished a general purpose configuration utility for the GS phones: 1) Generates files from scratch (using MAC), from HTML config listing, or by directly downloading from the phone. 2) Does multiple simulteneous edits. 3) Can reboot as many or as few phones at a time as you like. I would like to offer it to the list, but there are 2 issues: 1) I want to GPL it first, if

All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way?