From owner-freebsd-questions@FreeBSD.ORG Tue Oct 12 21:02:06 2004 Return-Path: Delivered-To: freebsd-questions@freebsd.org Received: from mx1.FreeBSD.org (mx1.freebsd.org [216.136.204.125]) by hub.freebsd.org (Postfix) with ESMTP id EA21F16A4CE for ; Tue, 12 Oct 2004 21:02:06 +0000 (GMT) Received: from smtp13.eresmas.com (smtp13.eresmas.com [62.81.235.113]) by mx1.FreeBSD.org (Postfix) with ESMTP id 3573E43D2D for ; Tue, 12 Oct 2004 21:02:06 +0000 (GMT) (envelope-from norgaard@locolomo.org) Received: from [192.168.108.62] (helo=mx01.eresmas.com) by smtp13.eresmas.com with esmtp (Exim 4.10) id 1CHTmO-0004bG-00 for freebsd-questions@freebsd.org; Tue, 12 Oct 2004 23:01:56 +0200 Received: from [62.174.254.182] (helo=top.daemonsecurity.com) by mx01.eresmas.com with esmtp (Exim 4.41) id 1CHTmP-0004IA-4Q for freebsd-questions@freebsd.org; Tue, 12 Oct 2004 23:01:57 +0200 Received: from [192.168.0.32] (charm.daemonsecurity.com [192.168.0.32]) by top.daemonsecurity.com (Postfix) with ESMTP id DFE33A1426 for ; Tue, 12 Oct 2004 23:01:56 +0200 (CEST) Message-ID: <416C4640.3020706@locolomo.org> Date: Tue, 12 Oct 2004 23:01:52 +0200 From: Erik Norgaard Organization: Loco Lomography User-Agent: Mozilla/5.0 (X11; U; FreeBSD i386; en-US; rv:1.7.2) Gecko/20040918 X-Accept-Language: en, en-us, da, it, es MIME-Version: 1.0 To: freebsd-questions@freebsd.org X-Enigmail-Version: 0.84.2.0 X-Enigmail-Supports: pgp-inline, pgp-mime Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit X-Spam-Score: 0.0 (/) Subject: VoIP: sip client X-BeenThere: freebsd-questions@freebsd.org X-Mailman-Version: 2.1.1 Precedence: list List-Id: User questions List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , X-List-Received-Date: Tue, 12 Oct 2004 21:02:07 -0000 Hi, I am trying to find a SIP client to work behind an ADSL router with NAT. I have tried linphone, but it seems not to support STUN, and I have tried kphone which crashes regularly and I have no sound. Is there another SIP client that works? Or should I try setup Asterisk or SER to proxy calls from linphone? Sorry, I'm new to VoIP and asking the _right question_ (TM) is difficult. Any suggestions, directional pointers or references would be greatly appreciated. Thanks, Erik -- Ph: +34.666334818 web: www.locolomo.org S/MIME Certificate: http://www.locolomo.org/crt/2004071206.crt Subject ID: A9:76:7A:ED:06:95:2B:8D:48:97:CE:F2:3F:42:C8:F2:22:DE:4C:B9 Fingerprint: 4A:E8:63:38:46:F6:9A:5D:B4:DC:29:41:3F:62:D3:0A:73:25:67:C2