similar to: Detect line is busy with Zap?

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to Dahdi (2.1.0.4). Everything seemed to go smooth with the exception of meetme. Meetme seems to not be able to find a zap channel for conferencing. We use voice introductions in our conference bridge and it seems to break that feature. The error from the console is.... # app_meetme.c:2593 find_conf: No Zap channel available for

I'm new in hire so Hello to everyone! I'm beginner user of Asterisk CVS-HEAD-10/01/04-14:31:34 . I just installed it on my Mandrake Linux 10.0 kernel 2.6.3-4mdk with sample configuration (used make samples). I would like to make phone connections between X-Lite (SIP) installed on computers in LAN. How to make this? I was reading manual, and tried to make changes in sip.conf but this all

This may be obvious but I have not found the answer in the archives or web searching. I am in the process of transitioning to Asterisk. While I have two systems connected to the same PSTN line, I want to configure Asterisk to not answer an incoming call. Is this a setting that you would have in the zapata.conf file? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I've got an MD3200 modem which is working as a Zaptel FXO interface for Asterisk (X100P clone I believe). It seems to work, but on incoming or outgoing calls I can hear the other party ok but when I speak, I hear my voice echo back at me (quite quietly but it's distracting!) on everything I say. The other party doesn't seem to hear the echo - they just say I sound a bit quiet.

Hi, Can anyone help me to understand the ScheduleDAGInstrs::buildSchedGraph() method? I find the handling of AliasChain is disturbing since: 1. A new alias chain add deps to all possibly aliasing SUs, and then clears those lists. 2. When AliasChain is present, the addChainDependency() method is called, but the target hook areMemAccessesTriviallyDisjoint() called inside

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either.

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the

Hello again, Sorry -- I think I found the problem somewhere else. I was a bit confused and missed the fact that adjustChainDeps() is called a few lines down and does just what I wanted :-) I would like to instead ask another question: Why is I->isCtrl() used in code like // Iterate over chain dependencies only. for (SUnit::const_succ_iterator I = SUb->Succs.begin(), E =

Hey guys, I'm looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, I'm not sure the extent of the coding as far as technical terms go at all. Currently we have a "call center" with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use

Dear R developers, sessionInfo() below Please have a look at the following two versions of the same function: 1. Intended behavior: > Su1 = function(u=100, l=u, mu=0.53, sigma2=4.3^2) + { + print(c(u, l, mu)) # here, l is set to u?s value + u = u/sqrt(sigma2) + l = l/sqrt(sigma2) + mu = mu/sqrt(sigma2) + print(c(u, l, mu)) + } > > Su1() [1] 100.00 100.00 0.53 [1]

Hi, Thank you for the reply. >It looks to me like we can choose any subset of edges here and be correct. We're basically trying to prune/pinch the DAG edges here. They can easily blow up with AA sched. I would guess that isCtrl() edges are good ones to bypass because they could be a low-latecy edges, whereas true data dependencies from a load are expected to be >higher latency, so they

I'm using * at home and I planned on having * let the answering machine in my kitchen to the "general" voicemail getting. However, about 6s into the call * will hang up the line. I found a post about OHT somethingorother, so I can probably work around it, but I'd like to know what's happening and if there's a better way around. Thanks! -- -M There are 10 kinds of

Hello, One way of preventing that is to use ?force. Just put force(l) right after the commented out print and before you change 'u'. Hope this helps, Rui Barradas Citando Matthias Gondan <matthias-gondan at gmx.de>: > Dear R developers, > > sessionInfo() below > > Please have a look at the following two versions of the same function: > > 1. Intended

Mathias, If it's any comfort, I appreciated the example; 'expected' behaviour maybe, but a very nice example for staff/student training! S Ellison > -----Original Message----- > From: R-help [mailto:r-help-bounces at r-project.org] On Behalf Of Matthias > Gondan > Sent: 02 September 2017 18:22 > To: r-help at r-project.org > Subject: [R] Strange lazy evaluation of

Another way to avoid the problem is to not redefine variables that are arguments. E.g., > Su3 <- function(u=100, l=u, mu=0.53, sigma2=4.3^2, verbose) { if (verbose) { print(c(u, l, mu)) } uNormalized <- u/sqrt(sigma2) lNormalized <- l/sqrt(sigma2) muNormalized <- mu/sqrt(sigma2) c(uNormalized, lNormalized, muNormalized) } > Su3(verbose=TRUE)

Dear Bill, All makes perfect sense (including the late evaluation). I actually discovered the problem by looking at old code which used your proposed solution. Still I find it strange (and, hnestly, I don?t like R?s behavior in this respect), and I am wondering why u is not being copied to L just before u is assigned a new value. Of course, this would require the R interpreter to track all these

Hi All, I was made aware today that some of my calls coming in are not going to voicemail... Below are some logs, and the macro that should run on the incoming_pstn context for that extension. I can see that theres a non-zero exit before it gets to voicemail, but I've no idea why. In this case theres 2 SIP clients to sim-call. On other occasions it works fine. In the CDR logs, I can see

Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing