similar to: New Asterisk Implementation

Hello Everyone, Pardon me if im sounding like a total idiot, but im new to this and have to ask. Numerous people have been telling me that I will be able to somehow do long distance calling for free when I roll out Asterisk.. and yet none of them can explain to me how exactly that will be. So.. I'll ask the community at large.. is this total BS or is there actually a way to reduce my long

Hello, I am having an issue with hangups being handled within Asterisk. Right now, when an inbound call hits the Asterisk box, Asterisk picks up the call just fine. When the caller enters an extension to call, the Asterisk dials out on Zap/3 and rings the extension with no problem. If the extension is answered, there is no problem. If the caller hang's up before the phone is answered,

Hi, I'm looking at setting up an Asterisk PBX in our office, which gets its phone lines (digital signaling, analog voice) from the main campus, which uses Centrex. Does anyone know if this falls under analog or digital for hardware buying? I was looking at getting a Digium TDM-series, but apparently our lines aren't pots (due to the digital signaling). Could someone enlighten me a bit?

Hello, I am still researching my dive into Asterisk at my workplace, and I was wondering about how switchboard activities are handled.. Right now, a call comes into our switchboard, and the operator forwards them to the appropriate line, thus freeing up the primary number and allowing more calls in. Everyone on campus has a direct-dial line as it is right now. I want to eliminate most of

Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the

Hello All, I have a request from a prospectieve client to deploy a PBX capacity that can do up to 3000+ lines within a geographic region similar to a campus. The client wants analog lines for extensions and maybe VoIP for some backhaul traffic while the other traffic would be carrid via E1 channels. The client has other proposals to buy a mid, range telco switch from alcatel or simens but i am

I haven't dealt with Centrex for a long time, and one of my customers is being courted heavily by a Sprint salesperson. Am I not correct in assuming that each "line" of Centrex corresponds to an "extension" in the PBX world? This site has 2 POTS lines and 5 extensions, and they told me that for the same thing they're paying right now (~$40/POTS line) they will be

Working under Windows XP, I am compiling a package called 'pgirmess' with the command rcmd build --binary --auto-zip pgirmess I have this message error after having listed: functions text html latex example chm .... zipping help file hcc: not found cp: cannot stat 'c:/TEMP/Rbuild365620874/pgirmess/chm/pgirmess.chm': No such file or directory make[1]: *** [chm-pgirmess] Error 1

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

Full_Name: Zivan Karaman Version: 2.7.2 OS: Windows XP Submission from: (NULL) (195.6.68.214) I'm puzzled by the readRegistry function. Shouldn't the "hive" argument be something like c("HLM", "HCR", "HCU", "HU", "HCC", "HPD") rather than c("HLM", "HCR", "HCU", "HU",

Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards,

There are at least four clang frontends for offloading to accelerators: 1 Cuda clang 2 OpenMP 3 HCC and 4 OpenCL. These frontends will want to embed object code for multiple offload targets into a single application binary to provide portability across different subarchitectures (e.g. sm_35, sm_50) and across different architectures (e.g nvptx64,amdgcn). Problem: Different frontends

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would

anyone know of a solution where mobile handsets out roaming the pstn cellular network can be used and treated as full fleged centrex extentions, i.e. I can transfer a call that comes in on a wired centrex copper pair out to a cell phone and the cell phone can transfer the call back or vice versa where the cell phone recieves the call directly and can transfer to the office all without hairpinning

Does anyone know what would be involved in making Asterisk work as a voicemail system in a Centrex environment? We have a Centrigram voicemail system that belongs in the Smithsonian. There are analog lines coming into the box and a 56KB data feed from the phone company's switch. Peter

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Hi, I have been trying to add labels to the rows of a heatmap produced using image() function. It is simply not working. Here is what I did. A2Rplot.hclust(hcc,k=length(num),col.up="black",col.down=num,lty.up=2,lty.down=1,lwd.up=1,lwd.down=2,show.labels=FALSE) #used the above external program to create a colored dendrogram xsort <- x[1:nrow(x), hcc$labels[hcc$order]]

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

Does anything have to be done at compile time in order for Asterisk to take advantage of 2 CPU's? Thanks