similar to: Where is my normal dialtone? With DLINK DG-104S (MGCP)

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 :

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 an easy way to make the voicemail system say the extension number after the directory find (via name)? People want to know the extension once they have found the person to speed up the process. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

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:

Good morning everybody, Can someone explain to me the interconnection between these four things: indications.conf, SetLanguage(), zaptel.conf and ring-back ? If there is any !! :- ) I am having this case where some users cannot hear ring back from a DeadAGI script and it seems to be interconnected to these items. These users are from the iaxfriends table, they _can_ hear ring-back from a

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 have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: dlinkgw@[192.168.1.1]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) --

Does anyone know if this unit works with Asterisk? Thx. B.

Many thanks in advance for anyone that can offer help on the following questions: Asterisk Box Using Asterisk@Home build and updated Asterisk to v2.1 P4, 400 Mhz, 384Mb RAM, 40Gb HD 4 OEM X100P Cards Phones Grandstream GXP-2000 2 * Grandstream BT-100 HandyTone 486 Sipura SPA-3000 Questions 1) When someone calls in to one of the FXO lines, there is a 3-4 second delay before the configured

Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

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

extensions.conf has ignorepat => 9 exten => _9X.,1,Dial(Zap/G2/${EXTEN:1}) The first user to try it asked if instead of keeping the same dialtone after pressing 9, if I could play a different dialtone. Can this be done? I'm running asterisk 1.0.0 in case that matters.

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

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

Hi there i have some problems with some clients of my asterisk box, i have some cases when a client tried to make a call and there is no ring back only a silence and then the call hung up. I dont know why this is happening. I have the following stable asterisk version: CVS-v1-0-01/18/05-19:49:31 I did the an update a few days ago, the version that i had installed before was: