similar to: documentation of DAHDI dial options

Hello, would like to have distintive ringtone for internal calls, google gave me blurr answer. My extensions are 46**, any calls made within 46** I want to ring differently than external calls. Thanks in advance.

Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie

Hello, LLVMers. I just wonder How I can pronounce Clang. [see-laeng], [see-lang], [k-laeng], [k-lang]?? Thanks, Seung

On Jul 3, 2008, at 8:06 AM, Seung Jae Lee wrote: > Hello, LLVMers. > > I just wonder How I can pronounce Clang. > > [see-laeng], [see-lang], [k-laeng], [k-lang]?? We've been pronouncing it kl -------------- next part -------------- A non-text attachment was scrubbed... Name: abreve.gif Type: image/gif Size: 62 bytes Desc: not available URL:

On 08/26/2010 11:53, Eric Christopher wrote: > It basically means that a pseudo wasn't lowered to something that the jit can output before the jit was run. Is this on ToT? > I don't know what is ToT. 42 meanings listed here http://acronyms.thefreedictionary.com/TOT don't seem to apply. Yuri

I read this list often, and although I'm not smart enough to contribute, I learn a lot every day. But I read a post from this morning that contained this: >a fair amount of jiggery-pokery Just wondering...is this a technical term? Where is it documented? I won't offer any insight into what vision it gave me. ;-) Thanks for all the work on the distribution and all the invaluable

On 08/26/2010 11:11, Bob Wilson wrote: > A pseudo instruction is used internally without llvm, but it does not correspond directly to a real instruction in the target architecture. Before emitting the final compiled code, all the pseudo instructions must be expanded to real instructions. If you're seeing that error, it means that something was not expanded. > So what is likely to

On Aug 26, 2010, at 11:47 AM, Yuri wrote: > On 08/26/2010 11:11, Bob Wilson wrote: >> A pseudo instruction is used internally without llvm, but it does not correspond directly to a real instruction in the target architecture. Before emitting the final compiled code, all the pseudo instructions must be expanded to real instructions. If you're seeing that error, it means that

Hi guys, Have been passed some weeks since the last wiki theme update. So it's update time :) Don't know if you really want theses updates, but here we have some significant changes that I would like to put in your consideration. Sincerely, I get tiered of writing changes into a readme file ... so I took some screenshots for you. If someone needs a diff file tell me, please. The general

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

Dear volunteer: I am a graduate student of medcine in china.And now,I am devoting myself to constructing a nomograms of bladder cancer.I want to do it with R-project.However, I do not know how to construct a nomograms with R-project.I want to get yours help,thank you! I wish you can tell me the operating procedure of the R-project. And I apologize for my english,it is poor,sorry!

Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf

Hi the list, I am writing a package in S4 and I do not manage to understand a bug. The "R CMD check" and the "R CMD build" both work. Here is links to the package (not on CRAN yet for the raison that I explain bellow): http://christophe.genolini.free.fr/aTelecharger/kml_0.5.zip http://christophe.genolini.free.fr/aTelecharger/kml_0.5.tar.gz Then I install the package and I

Hi all, thanks in advance for those who''ll take the time to help out... i have a list of users who are connected with a habtam relashinship to a set of groups (joint table of groups_users) and now am trying to implement a search operation that will go through all of the records of the users and output user results according to what group was chosen at the searchbar, and the text

I have a system with two 4 port T1 cards, with 5 PRI's configured. Each PRI is configured as an individual PRI and belongs to it's own group (groups 1-5) This system is handling roll-over from another system, where any error in processing the call on that system results in it being sent here. This mainly results in all calls resulting in a busy being sent for retry here. I then

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

I have setup as per the monitor example configuration on the wiki site and all works well for an extension dialing 8 then the number. However, if I dial from an AGI script the recording stops after a few seconds. I see an extra answer in the console and suspect that is the reason. Could any kind soul help me to get around this? Extensions.conf.. exten =>

Does anyone know if realtime extensions allows extensions in the format callerid/extension yet? ie the extensions.conf file allows you to do: 5551212/1000 => exten ... and it matches against extension 1000 when the caller id is 5551212. Last time I checked, realtime didn't support this yet. That's a show stopper for us. -------------- next part -------------- An HTML

Here is the setup: SIP/3044 -> SetCallerID(5551212) -> Call out PRI The CDR shows a src of 5551212. That is a lie! The src of that call was not 5551212, the source was 3044! The "translated source" of that call was 5551212. How can I get "real" source of this call and not some faky nonsense? The "reason" behind using the SetCallerID is because if I

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --