similar to: Problem with Hangups

Hello, I've been planning to replace my aging CENTREX switch with a new PBX and am seriously considering Asterisk as my solution. I work at a college, and we currently support just under 300 regular analog lines to the offices and whatnot. I was wondering.. Is asterisk ready for such a job? Would I be making a mistake to deploy this across the campus at this time? It seems that

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

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

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

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",

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

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]]

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

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

Good afternoon, I ran heatmap and hclust on the same matrix x (strictly, I ran heatmap(x), and hclust(dist(t(x))), and realized that the two dendrograms were slightly different, in that the left-right arrangement of one pair of subclusters (columns) was reversed in the two functions (but all individual columns were grouped correctly). Looking through the code for heatmap as a most definite

We have the ability to create random UID's on own system through a custom CGI API. These UID's are written to individual sip configuration files based on the account name, so for instance sip_TEST.conf, sip_TEST2.conf, and sip_TEST3.conf, etc. Many of these UID's are created on the fly and at random times throughout the day. Right now, I have it setup to do a reload every night.

Hi, I want to prevent Asterisk from sending the h extension across to the SIP provider or to prevent it from hitting the script at all. The SIP Provider does not know what to do with the h extensions once it receives it. My SIP Provider takes all digits and forwards them off to a softswitch for processing. Everytime a call hangs up, it complains about running AGI scripts on hungup

Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It

I have asterisk FXO module on a TDM400P hooked to an Intertel Single Line Card. I can place Intertel intercom calls to Asterisk (both SIP and analog phones) and the reverse, but transfering calls doesn't work. Here what the transfer looks like: Intertel FXS > Transfer > Asterisk FXO > Asterisk FXS > any extension When the call is answered Asterisk hangs up. Intertel

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

Hello all, I am having a big issue that i just cant get fixed for a while now and its causing me lots of grief. It seems like since we got FIOS installed (including switching to fios phone lines which are supposed to be the same on our end) i am having massive problems with asterisk not hanging up dead calls for days, even weeks if i dont catch it. It slowly builds up randomly not ending a call

2016-10-11 3:52 GMT+08:00 Mr.X <mrx at hcc.im>: > Network extension apps are just like ordinary apps, only they can utilize > the utun interface and create VPN entries. So I think, yes, tinc need to be > linked from such apps. link to and link from is mark as the main application's view, so, for NE extension it's link to libtinc( if there is a libtinc lib), that's

Hi! We will soon start upgrading our Ubuntu mail server (running 2.0.19) from 12.04.1 LTSto 14.04 LTS. The dovecot config is still pretty much in 1.x style; one flat file called dovecot.conf. I can easily switch to a dovecot.conf with the "!include" lines and rename my flat config file to local.conf. I've tested that and it works fine. I could also try to move all lines in local.conf

Hi, When I do a sip show peers in the cli, the status is lagged. This peer its behind a satellite link with 600/900ms of delay. May I change some parameter in the Asterisk? Some times I cant make a phone call from the remote site to my central site. Thanks Este mensaje ha sido analizado por C4I Mail Server en busca de virus y otros contenidos peligrosos, y se considera que esta limpio.