similar to: nested dial, or jump to another line to continue dialing.

so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com> wrote: > after adding the ww: > root at Pbx: /etc/asterisk $ asterisk -rvvv > Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits > 184 > == Using SIP RTP CoS mark 5 -- Executing > [9211123456 at AllCalls:1] Goto("SIP/500-00000003",

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [s at macro-dialout-trunk:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack [Feb 25

Started a new thread as my problem is somewhat different than the OP. Seems his somewhat different problem doesn't work as advertised either. Eric Wieling wrote: > I don't know what the problem is, but this is what I use and it works on my analog FXO port. > exten => _9NXXNXXXXXX,1,Dial(${PSTN}/w${EXTEN:1}) So, I modified slightly to fit my dialplan: exten =>

Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate

I need to execute account number, device number after dialing main number; what is the best solution? Is it possible to pause during dialing. Dial 1-800-number press 1 for English wait 5sec enter device number wait 5sec enter device ID What are my best options? -- #Joseph

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel (you can also use 'w' to produce .5 second

Hi R users, I am trying to use the optim function to maximize a likelihood funciton, and I got the following warning messages. Could anyone explain to me what messege 31 means exactly? Is it a cause for concern? Since the value of convergence turns out to be zero, it means that the converging is successful, right? So can I assume that the parameter estimates generated thereafter are reliable MLE

I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551212 at sip.com,60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten =>

I have a work around for this, but can someone explain why the first example does not work properly? I believed it worked in the previous version of R, by selecting just the rows=200525 and omitting the na's. I just upgraded to 2.13. I am also concern with the row numbers being different in the selections, should I be worried? FYI, I just selected the first few rows for demonstration, please

Hi everyone, On CentOS 7 I'm running into an issue with the latest nvidia driver from elrepo: kmod-nvidia-384.98-1.el7_4.elrepo.x86_64 This driver version seem to introduce issue in detecting video modes when a monitor is connected using DVI. As soon as the machine attempts to start X, nothing happens and the monitor goes into sleep mode reporting that it has 'no signal'. It is

Sorry for this dumb question. Suppose I have a named array ww defined as ww <- 1:5 names(ww) <- c("a", "b", "c", "d", "e") How can I extract the whole array of numbers without the names? ww[1:5] does not work while ww[[1]] can only extract one number at a time. Thanks, Gang

Hi all, Typically I dont have issues with configuring X. However this time its an issue. I have an NVIDIA graphics card trying to do 1920x1080. If I configure the screen by hand with the nvidia config utility it works at 1920x1080. However when I reboot its 800x600. When I use the nvidia config utility to set 1920x1080 then save the xorg.conf file, and reboot (yes I copied /tmp/xorg.conf to

Is there a way to dial DTMF after bridging the call. The current option D() in Dial will dial DTMF before the call is bridged and this doesn't do the job. I need to dial DTMF after the call is bridged and the message is played with "Background" -- #Joseph

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

-----Original Message----- From: Chad Brown Sent: Tuesday, December 21, 2004 8:02 PM To: 'el_flynn@lanvik-icu.com' Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not thePSTN Flynn, Yes, that makes sense. However, in my case I have incoming calls arriving on an IAX channel from a PSTN gateway. I think the concept is the same. That said, if incoming calls have access

I've only been using R on and off for 9 months and started using the glasso package for sparse covariance estimation. I know the concept is to shrink some of the elements of the covariance matrix to zero. However, say I have a dataset that I know has some underlying "baseline" covariance/correlation (say, a value of 0.3), how can I change or incorporate that into to the

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

I want to pass a predefined string ww ("fa*fb+fc") to function lme so that I can run > lme(y ~ fa*fb+fc, random = ~1|subj, model) I've tried something like > lme(y ~ paste(ww), random = ~1|subj, model) and > lme(y ~ sprintf(ww), random = ~1|subj, model) but both give me the following error: Error in model.frame(formula, rownames, variables, varnames, extras,