similar to: Two FXOs getting bridged?

Hey all, been noticing some oddness on a new AAH install... occasionally an incoming zap line with automatically connect with an outgoing extension, even though the incoming line hasn't specified what extension it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's trying to call out from inside the office & are automatically connected with an incoming line.

Hi all, Does anybody have the experience of using optim to estimate variables with integral forms? here the code: trun.mean<- function(x) # t is the threshold { mu=x[1]; sigma=x[2]; t=x[3]; f <- function(x) (1/(sigma*sqrt(2*pi)))*exp(-(x-mu)^2/(2*sigma^2)); pdf.fun <- function(x) x*f(x); integrate(f,thre,upper=Inf)$value/integrate(pdf.fun,thre,upper=Inf)$value ; } when I

Hi, I've been running some simulations for a while and the performance of R has been great. However, I've recently changed the code to perform a sort of chi-square goodness-of-fit test. To get the observed values for each cell I've been using table() - specifically I've been using cut2 from Hmisc to divide up the range into a specified number of cells and then using

hey all, trying to get a zap extension to work & I can dial out normally with it, but if I try to access any of the features (i.e. *97 for voicemail) the zap channel doesn't hear it, and all i get is dialtone. Is there a dialplan setting or something to make the zap channels recognize keys like * or # ? Thanks in advance

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it

http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4 jitter buffer, however it raised a question in my mind. In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP RTP packets renumbered on transmit, or is the original sequence number preserved in the UDP header? A comment is made on the referenced blog that jitter buffering is best implemented at the

Hi - Just downloaded and installed the memcache-client and cached_model gems and am trying to test it out on a development setup. I added this to my environment/development.rb CACHE = MemCache.new :c_threshold => 10_000, :compression => true, :debug => true, :namespace => ''eztrip'',

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

Hi All... Well, I'm still experiencing LOTS of dropped calls since installing the new (non pri) T1 here... I keep noticing a few things in the logs when this happens, namely the "Wink/Flash" statements and the "Didn't get a frame" messages... Anyone got any ideas on if this is a telco issue, a wiring issue, or an asterisk issue? Been trying to track this down via all 3

Dear R helpers, I have following loss data and I need to fit LEFT truncated Log Normal distribution to this data which is Truncated at 1000000. dat = c(1333834,5710254,9987567,7809469,6940935,3473671,1270209,1102523,1124002, 5830159,4302300,3925242,2638409,2324421,7238436,9088709,7439250,4976551,4864319, 8741334,1863770,7098310,4942288,4971829,4986372) library(gamlss.tr) gen.trun(5, LOGNO)

>From the manpage: --csum-length=LENGTH By default the primary checksum used in rsync is a very strong 16 byte MD4 checksum. In most cases you will find that a trun- cated version of this checksum is quite efficient, and this will decrease the size of the checksum data sent over the link, mak- ing things faster. You can choose the

Hi I am trying to turn on register_globals, but I am failing. someone trold me that I should change php.ini and I did it. ? - register_globals = Off - register_globals = On I made a php test page html/test.php <?php phpinfo(); ?> and checked it, but I can not make it. output_buffering no value no value output_handler no value no value post_max_size 8M 8M precision 14 14

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

On Thu, Oct 27, 2016 at 5:39 PM, Maxime Chevalier-Boisvert via llvm-dev <llvm-dev at lists.llvm.org> wrote: > Hello, > > We’d like to keep track of which clang version was used to build our LLVM binaries. We use cmake and ninja with clang to build. What do you people think would be the cleanest way to know which version of clang is used, on a user’s machine, to build those binaries.

Anyone used voicetronix FXOs with * ? I'm interested to know how they compare with eg TDM400P. Specifically I'm interested in how good the echo canceller is. -Dan

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dan Phillips wrote: > Charles Steinkuehler wrote: >> Dan Phillips wrote: >> >> What we are left with is a requirement in abcde to overcome this and >> >> until then we have the manual method. Have you any thoughts on the best >> >> way to overcome this problem apart from the hacked toc3cue (do you have

Hi all, Theoretically, I know it's possible, but is any using multiple tdm400ps (fxo) in single * box? In a production environment? Any gotchas aside form irq sharing? Thanks Ryan

Should asterisk be intercepting DTMF on a bridged ZAP call? If so, how do I disable it recognizing #, as it's hanging up my users when they try to enter #. ________________________________ This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the

Charles Steinkuehler wrote: > Dan Phillips wrote: > >> What we are left with is a requirement in abcde to overcome this and > >> until then we have the manual method. Have you any thoughts on the best > >> way to overcome this problem apart from the hacked toc3cue (do you have > >> a copy of this?) > > I do not have a copy of toc3cue...I made my own

Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup: PRI IAX <-------->* PSTN <------->* Dialplan I've configured a number on the dialplan server to make and outbound call to the pstn. This call then comes back into the dialplan server to SayDigits(). I'm seeing that a few of my digits are being duplicated