similar to: short ringing

returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxxxxxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not behind NAT or a firewall -------------- next part -------------- An HTML attachment was scrubbed...

I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 & 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories & buddy lists. Unfortunately some of the

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

Hi guys, What is different and the "context" to play unavail message and busy message? When a SIP connection is unregistered, voicemail will play unavail message, right? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/1a2d1c81/attachment.htm

Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf

Buenas noches a todos las personas del foro, Me dirijo a vosotros con la siguiente cuestión: ¿Cómo es posible obtener, para cada individuo (identificado con la variable ID) de un data frame, la CANTIDAD ACUMULADA hasta una determinada fecha (día, mes y año), que es diferente para cada individuo y que se denota por la variable ENTRADA? Ejemplo: require(data.table) datos2 <-

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk

Hi, My client wants incoming callers who do not press a digit to go straight to the operator. Does anyone have an idea of how this could be done? I've looked for some examples, but I'm still not clear on it. Here's the relevant portion of my extensions.conf: ------- ; Wait 15 seconds for an answer (pick up the local phone) exten => s,1,Wait,2 ; Answer the phone exten =>

First, the Asterisk settings: ----- sip.conf ----- [general] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) context=default ; Default context for incoming calls disallow=all ; First disallow all codecs; Set order allow=ulaw ; also known as g.711 PCMU; allow codecs in

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

En primer lugar, muchas gracias a Carlos J. Gil Bellosta, tanto por la rapidez como por la precisión de su respuesta. El código que planteas es exactamente lo que preguntaba. La única duda que me queda es si el data.table final que queda, sólo puede recoger cantidades totales que no sean nulas. A modo de ejemplo, supongamos que los datos contuvieran, para el sijeto ID=100, una variable

Dear R Helpers, I have noticed that when I use lmer to analyse data, the summary function gives different values for the AIC, BIC and log-likelihood compared with the anova function. Here is a sample program #make some data set.seed(1); datx=data.frame(array(runif(720),c(240,3),dimnames=list(NULL,c('x1','x2','y' )))) id=rep(1:120,2); datx=cbind(id,datx) #give x1 a

Is there any way to have users be able to turn on or off call forwarding at the asterisk server, so they can configure their own forwarding number and enable/disable it? Hopefully, with the added benefit that it will remain on between server reloads and restarts? I have written a hack -- a AGI script to do various checking, and if the destination is "ok" set a database variable

Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary

one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in

Hi, I had ordered a TDM40B and developers kit a few months ago. I have everything installed and working, with one exception - sound quality. When placing a call it sounds like a very bad cordless phone - lots of hiss / static in the background. This even happens with the dialtone, though it is much worse one the call is connected. This does not occur when the phone is directly connected to the

Colleagues, I am in the process of learning R. I've been able to import my dataset (from Stata) and do some simple coding. I have now come to coding situation that requires some assistance. This is some code in SPSS that I would like to be able to execute in R: if (race eq 1 and usborn=0) confused=1 . if (race eq 2 and usborn=0) confused=1 . if (race eq 1 and usborn=1) confused=0 . if (race

............ -- Executing [s at incoming-pstn-line:5] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') .......... But there is no /var/spool/asterisk/voicemail/default/100/unavail.gsm', indeed no unavail.gsm on the machine. [asterisk at

Hi guys, I have a quick question, I am playing around with ZFS and here''s what I did. I created a storage pool with several drives. I unplugged 3 out of 5 drives from the array, currently: NAME STATE READ WRITE CKSUM gpool UNAVAIL 0 0 0 insufficient replicas raidz1 UNAVAIL 0 0 0 insufficient replicas c8t2d0 UNAVAIL 0 0