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

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]

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.

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

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

Hi all, as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

hi all when starting kphone, it tries to register with asterisk but fails after a while. The SIP entry in * for this user is below. This is identical to the other SIP entries. The other SIP clients are MSN messenger plus one snom. these work fine. See SIP debug output attached as 'screen-exchange' thanks roy [roy] type=friend ;insecure=yes username=roy ;secret=password host=dynamic

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

Dear Asterisk community, sorry if I'm so stupid, but I couldn't register myself with Asterisk. I created the [sip-incoming] context in the sip.conf: [sip-incoming] type = peer username = elzhov port = 5062 ; my kphone listens port 5062 host = 127.0.0.1 Then run Asterisk, and checked peers that are known for Asterisk: *CLI> sip show peers Name/username

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

> > ---------- 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