similar to: LARGE BREASTS Handoff back to * from * via IAX?

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.

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.

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:

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

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 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

Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to

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)) {

I've searched the archives a bit and have not really come up with a good answer to my queries. I have * running on a RH9 box behind a LinkSys NAT box. I can talk with iConnectHere outbound just fine. I am trying to configure an inbound Xten softphone from outside. I have that user set as NAT in sip.conf (seems to help), but I still cannot establish a full session. I think the problem comes

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

Has anyone had any luck getting a ReplayTV DVR box to connect through an Asterisk box? Mine seems to dial just fine, but can't negotiate a connection. I am using: exten => _95380024,1,Dial(Zap/1/${EXTEN:1},120,d) exten => _95380024,2,Congestion I don't have any problems doing a fax though my system. For this setup, I am running a simple Digium developer's kit on a

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

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

I'm using ezstream-0.1.2 KJ -----Oorspronkelijk bericht----- Van: Joel Ebel [mailto:jbebel@ncsu.edu] Verzonden: donderdag 3 februari 2005 21:05 Aan: Klaas Jan Wierenga Onderwerp: Re: [Icecast] Stream drops during handoff. Suggestions? Thanks. I'll have to try that. I wonder why ezstream would ever stop sending data for that long though. What version of ezstream are you running? Joel

I have a hardware FXO/FXS which handle my voip calls, and they support G723 internally. Asterisk hands off these calls just fine, and everything works, as long as I don't want PBX menues available... The problem is, once I want it to return messages, it will only return them as GSM... which is fine, since my FXO/FXS support multiple coders. However, even though Asterisk lets me specify a

So, when I get no comments on this at all, either here or on any of the forums, does that mean nobody knows what I'm talking about?? Or does nobody know the answer?? Or is it just a stupid question and nobody wants to bother telling me where to look?? It *is* a question that I have to answer somehow; I've read all through TFOT and see nothing relevant to this issue. It's silly to

Sorry if this has been asked before, but I've searched high and low for the last couple days for an answer. An Internet radio station I DJ for is using Shoutcast and MP3, but we are considering moving to an Icecast/Ogg Vorbis combination instead. We work in 3-hour shifts. When we hand off, the DJ on-stream stops teh encoder, shouts "go" on IRC, and the DJ in line starts his