similar to: Selective Masquerading

Gracias. Es necesario instalar "forecast"?? Obtener Outlook para Android<https://aka.ms/ghei36> ________________________________ From: Carlos Ortega <cof en qualityexcellence.es> Sent: Friday, September 6, 2019 1:37:08 PM To: Christopher Estrada Ruiz <christopher_estrada_ruiz en hotmail.com> Cc: Juan Abasolo <juan.abasolo en ehu.eus>; R-help-es <r-help-es en

I have a closed network (no outside access) with a small user base (less than 20), with no other network connections) running Samba 1.9.18p10 on Solaris 2.4. Problem: Some users utilize the workstations on the network as stand alone systems. They need access to printers, but I don't want to create an account just for this purpose. I would like to provide access to the printers without

Hello Does anybody know if it's possible to monitor a Powercom EBk 500 (RS232)? the trick is that I am using a raspberry pi, it does not have RS232 so I connected a keyspan RS232 to USB adapter, but so far I can?t make this work. Thanks for your time to read this. -Edgar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Information: gcc -v: gcc version 4.3.3 (Debian 4.3.3-3) os: Debian/Testing Pulled latest release from asterisk site, compiled, installed it. I have a barebones configuration: $ ls -l asterisk extensions.conf modules.conf sip.conf users.conf voicemail.conf You can see them here: http://home.comcast.net/~jpiszcz/20090418/extensions.conf

Good Afternoon I?m really new at this and i don?t know if i?m posting at the right place. My problem is that our Dovecot IMAP is stopping from time to time with no info in the log, i just realize about this when our support person call me saying that the whatsup monitor is showing the IMAP4 process down. I can figure about this too by trying to get in our webmail Horde and i cannot login.

Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1:

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac",

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Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqbala at improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106

I figured I''d post here before submitting a ticket, but I''m seeing some confusing stuff when dealing with YAML now. I was using it to freeze objects in my database, and so i had some data already around to mess with. I upgraded both Ruby (1.8.3 -> 1.8.4) and Rails ( -> 1.1RC1) and this junk started. I''ve outlined the problem in two pastes, which I''ll

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hello, Let's say we have the following function: foo <- function(x) { line1 <- x line2 <- 0 line3 <- line1 + line2 return(line3) } And that we want to change the second line to be: line2 <- 2 How would you do that? The two ways I know of are either to use fix(foo) And change the function. Or to just write the function again. Is there

I've tried several different ways to accomplish this, but as yet to no avail. My y-axis for a plot has a rather long label, and thus I have been using "/n" to break it into two lines. However, to make it technically correct for publication, I also need to use superscript in the label. For example: par(oma=c(0,0,2,0),mar=c(5,6,0.25,2),lheight=1) plot(1:10,

I am trying to implement a rollover of extensions. exten => 3000,1,GotoIf($[${line1} = Congestion]?3:2) exten => 3000,2,Dial(${line1},15,rt) exten => 3000,3,GotoIf($[${line2} = Congestion]?5:4) exten => 3000,4,Dial(${line2},15,rt) exten => 3000,5,GotoIf($[${line3} = Congestion]?7:6) exten => 3000,6,Dial(${line3},15,rt) exten => 3000,7,GotoIf($[${line4} = Congestion]?1:8)

Pregunta, ?Como puedo leer datos de SPSS con R studio? Obtener Outlook para Android<https://aka.ms/ghei36> ________________________________ From: R-help-es <r-help-es-bounces en r-project.org> on behalf of Juan Abasolo <juan.abasolo en ehu.eus> Sent: Thursday, September 5, 2019 11:28:37 AM To: Carlos Ortega <cof en qualityexcellence.es> Cc: R-help-es <r-help-es en

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

http://www.linuxquestions.org/linux/answers/Networking/Spanning_Multiple_DSLs -- Thank you Indunil Jayasooriya _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

I have the Polycom 500 and 600 phones. Rather than put an entry in for each line appearance, I would like to use the feature that shares one extension for the lines, so that I will get the call on the enxt available button. How do I configure that? Chris Mason