search for: singalling

Hi, I'm working with custom slides(Cy5) and working in the normalization of the arrays. I have three arrays (technical replicates). I have sucesfully normalized the data using vsn, however i would like to normalize using spike in controls. My controls are annotated as CTL-1 to x and i would like to do etiher a normalization by block per array or the mean of all the controls per array. The gal

hi: our company want to use samba3+openldap with singal sign on. we have several branches.people would travel arround head quarter and branches with their notebooks. so we don't have roaming users, but we do have roaming computers. we want to use a single domain for every site, and we want every site keep working even when wan link is broken. my plan below: 1. place

How I change my configuration to reduce this issue. I have this setting on my zapata.conf signalling=fxs_ks group=1 callgroup=1 pickupgroup=1 channel=1 signalling=fxs_ks group=2 callgroup=1 pickupgroup=1 channel=2; singalling=fxs_ks group=3 callgroup=1 pickupgroup=1 channel=3; singalling=fxs_ks group=4 callgroup=1 pickupgroup=1 channel=4 and for outbound calls I have this context on my extensions.conf [out-callb] exten => 44,1,Set(LANGUAGE()=es) exten => 44,n,ChanIsAvail(Zap/g1&Zap/g2&Zap/g3&Zap/g...

Dear all I have asterisk 1.2 and now i want to install E1 card with support Q.SIG singaling so which E1 card is best for my setup i need single port E1/PRI card which support Q.SIG Regards Satish patel --------------------------------- Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. -------------- next

...n=3.0 callgroup=1 pickupgroup=1 immediate=no ;busydetect=yes ;busycount=4 callprogress=no ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ;callprogress=yes faxdetect=incoming faxdetect=outgoing signalling=fxs_ks group=1 channel=1 signalling=fxs_ks group=2 channel=2; singalling=fxs_ks group=3 channel=3; ;singalling=fxs_ks ;group=1 ;channel=4

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

Dear R developers, i've got a question concerning an embedded R instance. I called the initialization routines as seen in the "writing extensions" guide (Rf_initialize, setup_Rmainloop and R_ReplDLLinit). I also changed the function callback pointer of readconsole and writeconsoleex to my local functions. when i call mainloop there is no problem and the commands are evaluated

I'm terribly sorry if this isn't the proper place/method by which to get help with troubleshooting Samba errors/issues, but it's the best one I could find. I have an instance of Samba 4.8.3 running on a Centos 7.6 VM server (kernel 3.10.0-957.1.3), along with Centrify 5.5.2-578, that will allow folks to map a shared drive from their Windows 10 machines generally fine, but quite

Hi all, How do I directly send plots to printer, instead of showing them on the screen device? That's to say, with my current code for ploting on screen device, how to modify a few lines to send automatically the plots to the printer? If that's not doable in R, could it be possible to automatically generate a multi-page PDF file for all the plots, so that I can print one singal PDF file

Hello all. I am currently using ext3 0.0.5d with great success. I am a bit conflicted about what to tell the system regarding my root filesystem however. I have my root filesystem configured and working as an ext3 filesystem, but it is/was not without some fraught. Using RedHat 7.0, if you simply create your journal on the root file- system, figure out it's inode number, issue a lilo -R

Hi list! I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5. Very regularly asterisk seems to lose connectivity with the ISDN line. If you try to call in you get the information tone that the number is not in use. Outbound calls do stil work however. Unloading the modules and reloading them and start/stop asterisk will solve the problem. Another problem that occurs

Ok, so I think I have a complete explanation for the difference between the *BSD behaviour and the Linux/Solaris behaviour. Well, almost complete :) Pull out your trusty copies of "The Design and Implementation of the 4.4BSD Operating System" as well as "Unix Internals: The New Frontiers". Specifically, pages 111-112 and 344 of the former and page 108 of the latter. It comes

On Fri, 25 Jan 2019 23:42:57 +0000 "Scott Z. via samba" <samba at lists.samba.org> wrote: > I'm terribly sorry if this isn't the proper place/method by which to > get help with troubleshooting Samba errors/issues, but it's the best > one I could find. I have an instance of Samba 4.8.3 running on a > Centos 7.6 VM server (kernel 3.10.0-957.1.3), along with

Hi, I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104 FXS. I can not enter mailbox number (voicemail) or pin code (meet-me). Asterisk shows 'username not entered' when dialing in voicemail. Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ? Best regards, Dave

Hi, I'm recording wave files but I cant get Asterisk to play them, only if they are in 8000 Hz. What is the maximum sample rate Asterisk can handle? I have been using 16-bit 44.1, 22050 and finally 8000 kHz. Many thanks, Christian

Dear all I have setup with mediant 2000 with avaya now i want to install E1/PRI card with asterisk and trunk with E1 with Avaya E1 port so i want to buy E1 card for asterisk so which card is best and cast effective for my setup i want 1 port E1 card so can you suggest me which card is best for my setup and i want QSIG singaling with avaya Regards satish patel

...h TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme or FXO signalling problme i have configuraed singalling=fxs_ks ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. -------------- next part -------------- An HTML attachment was scrubbed... URL: ht...

Hi. Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to Asterisk List. If somebody knows where to search (dahdi lists or libSS7 lists) will be appreciated. Im getting this error after a certain time, My config is: Hardware: 3 Digium Quad E1 TE4XXP libss7 version: SVN-branch-1.0-r286 DAHDI Version: 2.4.0 Echo Canceller: Asterisk 1.6.2.14 CentOS release 5.5 (Final) Kernel

Hi, I am tryinig to work on the Mongrel examples. I am trying to modify the HttpResponse. def process(request, response) response.start do |head,out| if request.params["REQUEST_METHOD"] == OPTIONS" else end end One can detect that the request is OPTIONS like this, but how does one generate the custom response by using HttpResponse. Please help. Shubham

Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling type. When I dial out, I get this message: Dec 4 11:37:31] WARNING[27983]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I am using Dial(DHDI/g63/5555555555|20) in my dialplan. Anyone who has a tip, I would appreciate. I see...