similar to: BGM Music

Is there a way to get the OnHold music to restart without restarting asterisk? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I'm trying to use fine axis controls in lattice for each panel. Specifically, within each panel, I want to set the limits for x and y equal to each other since it is paired data (using the max value of the two). Of course, I have no problems setting the limits for the entire plot but I am having trouble setting them for each specific panel. Could someone please provide me some guidance?

I'm wondering if I am dealing with a limitation in lattice. It's probably due to my own limitations though. I'm working with a lattice dotplot. The x-axis is set at "free". In one panel, there are only two data points that have equal x-axis values. Since no range exists, lattice isn't plotting those points. It reports: "Error Packet using 6 'x' and

I have installed Digium TDM04B with the latest CVS. However I have encountered following problems: 1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement "please check the number and dial again" although I see on the screen that the dialed number is correct. 2. When the call is forwarded outside, with something like exten =>

Hi, Does anyone know who did the prompts for French and Russian for Asterisk? I need some custom prompts. Regards, Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/ae5eea65/attachment.html>

Hello everyone, I am trying to fit the following model All X. variables are continuous, while the conditions are categoricals. model <- lm(X2

I have been using Xen on a daily basis on a production (but not critical) machine for a number of months now. It''s looking really good. One thing that I have not yet seen anyone mention as a feature that I would really like to see is the ability to overcommit memory. I have 2G of RAM in my machine. I would like to give a developer his own virtual domain to sandbox his application

I'm trying to compile the last * CVS version and I got this error: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-09/10/03-18:47:18\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Hallo, All, I have a question about changing the height or scale of the y axis. When I use following two R codes, I can get two plots. Please look at the y axes, the number of indices (x1, x2, ?) on the y axis in the first plot is smaller than that in the second plot, and hence the space between any two indices in the first plot is wider than that in the second plot. As the number of indices

Attached is a .ll with a reproducible test case for the bug addressed by r100044 -- "Fix a nasty dangling-pointer heisenbug that could generate wrong code pretty much anywhere AFAICT." Doing llvm-as < sunkaddr.ll | llc which a llc after r100044 will generate .LBB1_2: # %if-false-block movl $1, 16(%rdi) movl 120(%rdi), %eax

I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about setting break key off in the advanced page of the web config but the Snom 220 has no such option. At the moment I am having to use the # transfer hack which makes this phone look really stupid to have buttons on it that cannot be used. Anyone know how to

Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten => 5150/0,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= wd@infodn.rmi.de =*= FAX: +49 2421 962001

Good day all I'm still looking for a SIP client that will work on fedora core 1? Thanks

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" mode. I mean setup like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same

There were a number of messages about using a digium card as a data t1 (router) kind of thing. Is there a howto out there or a status that I can see whether this is feasible or not? Or is it feasible but very difficult? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I have a client who is looking at replacing their PBX, and I'm interested in putting together an Asterisk solution for them. One feature that would really, really get their attention is if I could do the "Vonage" thing, where if a PSTN caller dials a direct extension (coming in over PRI) both the user's deskset _and_ an external number (their cell phone) would ring, with

Is the Vorbis ACM codec still being maintained? Then I encode an avi with Vorbis audio, the sound gets out of sync after a while no matter what I try. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in the

Hello, I'm having difficulties running WonderKing (http://wonderking.ndoorsgames.com/center/default.asp) on an Intel Mac version 10.5.8 using Wine 1.1.35. I was able to download and install both programs successfully, but when I run WonderKing, I'm able to get to the first screen where you can check to see if you have all the files needed to run the game (I have all of them), change from

Hello All, I am battling a problem I don't know how to fix... Here is the scenario: Fractionated T1 with 1-6 channels voice, 21-24 channels data. Comes into a box with Digium T100P, splits off data channes with HDLC, to devicec pvc0, This works fine, but I have a routing problem getting to my machines! Here is the current set up... sethdlc hdlc0 fr lmi ansi sethdlc hdlc0 create 44