similar to: Syntax (2)

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

Hello, >From: klky3@fibertel.com.ar >To: asterisk-users@lists.digium.com >Subject: Asunto: Re: [Asterisk-Users] Syntax >Date: Wed, 5 May 2004 17:06:56 -0300 > >Somebody knows a Howto that have the examples, but with comments ( the >cookbok >in the digium's page is quite diffcultly !!! ) > >Best Regards > >Ivan Checkout this url:

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

Muchas gracias, Jos Antonio y Carlos. Estaba por escribir sandeces varias. Comento, por si hay algn lector, que como yo, se guarda los correos con respuestas a problemas posibles. No estaba usando|header.tex|, que es necesario para una como para otra solucin. El encabezado queda as: |---|| ||title: "Tablas que no entran"|| ||author: "Sandia Urlia"|| ||date: ""||

Buenas a todos. No me queda del todo claro si éste es el foro en el que preguntar o debería ser otro, ya que incluye cosas que van más allá de R. Sepan disculpar si me estoy excediendo. Estoy trabajando con RStudio y quiero generar un documento de salida en PDF; en ese documento tengo bastantes tablas anchas que no entran usando knitr::kable(la.tabla.ancha) Sí salen perfectamente en HTML

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

muy buenas! acabo de empezar con linux y me he embarcau en un proyecto que = me queda un poco grande y que tengo que entregar dentro de poco... el caso es que quiero hacer funcionar el cbq.init y no puedo (el cbq.init es un script para controlar el ancho de banda de tu red). el caso es que tengo un debian woody 3.0 con el kernel 2.4, puestos todos los modulos de QoS e instalado el iproute.

Hey all, Has anyone ever had problems with flaky behaviour when registering softphones dynamically with asterisk? I'm working with Kphone and having problems getting consistent results with registration. When Kphone is able to register, asterisk reports -- registered SIP '2001' at XXX.XXX.XXX.XXX port 5060 where asterisk sees [2001] in sip.conf as [2001]

Hello, I know that my Kphone question may be a bit off topic, but I have been busy with this again and again for about one month now, sent three mails to kphone@wirlab.net (the contact address mentioned on http://www.wirlab.net/kphone/index.html), asked for a solution in a german ip phone forum and tryed many things by myself. I try to compile KPhone 4.0.3 (tryed CVS Version as well) but

I'm having trouble with kphone on our system. It's using ulaw on an internal network. No NAT. I had it working fine with very similar hardware (an old Dell Optiplex GX1) running as an LTSP terminal. But then I put the same sound card in an Optiplex G1. Kphone will answer the line fine when I call it (call coming from the * machine), but when we try to get kphone to dial, each GUI

Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco ----- Original Message ----- From:

Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime). Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable

Hola a todos, Sigo la discusin sobre la web de usuarios R y me pregunto porque nadie ha mencionado "Ning". A m me parece una herramienta perfecta para crear las redes sociales, y estoy seguro que satisfacera mayora, si no todos los requisitos propuestos (foros, blogs, fotos, videos, miembros ...) Para crear uno : http://www.ning.com/main/createyourown Aqu podis ver un ejemplo:

> > ---------- Mensaje reenviado ---------- > From: Roberto C. Sánchez <roberto@connexer.com> > To: shorewall-users@lists.sourceforge.net > Date: Fri, 4 Jul 2008 18:52:36 -0400 > Subject: Re: [Shorewall-users] Shaper > On Fri, Jul 04, 2008 at 04:47:31PM -0500, RokeFeler wrote: > > Saludos, Tengo 3 inferfaces eth0, eth1, eth2 > > eth0 - Net > > eth1 -

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

tc qdisc del dev eth1 root tc qdisc add dev eth1 root handle 10: prio bands 6 tc qdisc add dev eth1 parent 10:1 handle 20: bfifo tc qdisc add dev eth1 parent 10:2 handle 30: bfifo tc qdisc add dev eth1 parent 10:3 handle 40: bfifo tc qdisc add dev eth1 parent 10:4 handle 50: bfifo tc qdisc add dev eth1 parent 10:5 handle 60: bfifo tc qdisc add dev eth1 parent 10:6 handle 70: bfifo # www traffic

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

Hello all, I am trying to configure a linux box to make some QoS into my netowork and, at the same box, control my clients bandwidth. I have this classes created: ---------------------------------------------------------------- UP="eth0" # wan infocontabil DL01="eth2" # lan clientes $TC qdisc del dev $DL01 root 2> /dev/null >

Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only