similar to: Verifone-Asterisk-AGI

Estimado Elisabete Juaristi Arrieta Al leer su correo pensé que usted quería resolver un problema para lo que R no es adecuado, luego pensé que usted deseaba solucionar un problema donde hay datos y una ecuación a resolver. En este segundo caso yo muchas veces me pregunto cuál sería la forma para algunos problemas que tengo, muchos de estos tienen solución pero pienso que esa solución es puesta

Gracias Emiliano, pero no me vale. No conozco los valores de a,b y c. Tengo que plantear tres condiciones y luego a través de la resolución de un sistema lineal que me encuentre los valores de a,b, c. Puede que no lo tenga que hacer a través de function y haya algún otro modo. Gracias de nuevo! El vie., 25 oct. 2019 7:14, Emilio L. Cano <emilopezcano en gmail.com> escribió: > Hola

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Just run ‘core show calls’ as a command from the AMI, and parse the results. I don’t think there is an equivalent pure AMI command. From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent: Sunday, June 14, 2020 5:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re:

Way back in the mists of time, I built my asterisk installation with SNMP support. It's a bit tedious to get the sub-agent for snmpd set up but once you have it you can poll the OID for the asterisk sub-agent and it will tell you how many calls are up at that moment in time. That said, I actually prefer ARA/ARI to flat file configuration of endpoints and dialplans.  Changes are more or less

Thank you... but "just update the database" - hmm, what database? Did you mean ARI? I still can't find the command! The asterisk wiki is somewhat, um... spread around! On Sat, 13 Jun 2020 at 16:56, Steve Edwards <asterisk.org at sedwards.com> wrote: > > On Sat, 13 Jun 2020, Jonathan H wrote: > > > I need to ensure that a MusicOnHold stream is only running when

Trying to redirect to -user... On Tue, 7 Jul 2009, Bryant Zimmerman wrote: > Hey guys I posted this earlier and did not get any responses. You posted what appear[s|ed] to be a user question to the dev list. I did reply (on June 3), but I may have mis-understood. > I am working on some AGI development that requires control of audio file > playback. The control stream file is working

I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance,

On Fri, 24 Jan 2020, Benoit Panizzon wrote: > I have stumbled of this problem. > > I need the P-Asserted-Identity header in an AGI scrip. > > In the Dial-Plan I do: > > same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) > > In the AGI I do: > > my $pai = $AGI->get_variable(PAI); > > This works fine, unless the PAI contains quotes: > >

On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards <asterisk.org at sedwards.com>wrote: > On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards <asterisk.org at sedwards.com> >> wrote: >> >> I strongly suggest using an existing library for the language of your >> choice. >> > > On Mon, 6 Jun 2011, A E [Gmail] wrote: > > Copy that. Not planning to

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

Wow! I've been *-ing for about 6 years and had literally no idea about that! I can see a way I could put it to a different use, but it seems to be a bit of a sledgehammer to crack the walnut of "how many current callers" compared to one line of (albeit hacky) dialplan. That's making me sound ungrateful. I don't mean to be! On Sun, 14 Jun 2020, 22:39 Steve Edwards,

Hello, i have two xen hosts servers configured to support migration of domus, but when i try to migrate, the command never ends, i can see on the remote server the machine with xm list but the machine is not accesible, and on the own server i see migrating-{machine_name} and i have access to it. Every server have one nic, with public ip (that is using with xenbr0), and i have a virbr0 configured

Buenas noches, Tengo una función del tipo f(x)=a*x^2+b*x+c y sé que esta función pasa por el punto (1,3). Quisiera que R me sustituyera la x por 1, y me sacara la expresión que quedase para luego ese resultado igualarlo a 3, es decir, que hiciera lo siguiente, sustituyera x=1 en f(x), y me devolviera lo siguiente: a*1^2+b*1+c He probado a meter f de distintas formas: f<-function(x)

Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look

Hi All, I feel this is the most simple question but I am currently going around and round in circles and searches keep bringing me up Windows tools!! :-( I have a 512MB USB drive that has a 12MB FAT16 partition on it. How can I resize this 12MB partition to grow and fill the whole 512MB drive? Just in case I am being stupid, here is what I am doing... :-) I would like a quick USB drive that

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my

"asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> wrote: Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to

I see the following on one of my new servers: -ts10::sedwards:~$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 2979045 2988620 87780075 87779501 IO-APIC-edge timer 1: 1 3 2 3 IO-APIC-edge i8042 8: 0 0 0 1 IO-APIC-edge rtc 9: 0 0 0