[Wengophone-devel] Re: WengoPhone-2.2-minsizerel-alsa-12867 problems

2007-10-11 Thread Aurélien Gâteau
On Wednesday 10 October 2007 20:38:38 Alejandro Cabrera Obed wrote: Dear Aurelien and people, I've installed the last alpha version of wengophone (WengoPhone-2.2-minsizerel-alsa-12867) on our Debian Etch boxes but we have three problems: 1) The main bar is diffused (the part of the bar

[Wengophone-devel] MSN problem

2007-10-11 Thread Dave Neary
Hi all, I posted a patch for the MSN connection issue which should resolve the problem for people behind firewalls as well. It's attached to ticket #1766 http://dev.openwengo.org/trac/openwengo/trac.fcgi/ticket/1766 It would be great to have some feedback on it - it works for me on the 2.2

[Wengophone-devel] Re: WengoPhone-2.2-minsizerel-alsa-12867 problems

2007-10-11 Thread Alejandro Cabrera Obed
Julien Bossart wrote: Aurélien Gâteau a écrit : On Wednesday 10 October 2007 20:38:38 Alejandro Cabrera Obed wrote: Dear Aurelien and people, I've installed the last alpha version of wengophone (WengoPhone-2.2-minsizerel-alsa-12867) on our Debian Etch boxes but we have three problems:

Re: [Wengophone-devel] Patch for GCC - Boost check is in

2007-10-11 Thread Vadim Lebedev
What is the recommended setup on Feisty? The standard GCC is 4.1.x there Vadim Aurélien Gâteau wrote: For your information, I just committed the patch which implements a stronger check for Boost and GCC versions, as discussed before. Compilation will stop if you are using Boost 1.33.1 and

Re: [Wengophone-devel] Patch for GCC - Boost check is in

2007-10-11 Thread Dave Neary
Hi Vadim, Vadim Lebedev wrote: What is the recommended setup on Feisty? The standard GCC is 4.1.x there Install gcc-4.0, and set CC=gcc-4.0 or remove /usr/bin/gcc (which is a sym link to gcc-4.1) and ln -s gcc-4.0 gcc (do the same for g++ and cpp). Cheers, Dave. -- Dave Neary OpenWengo

[Wengophone-devel] Wengophone 2.2 and Asterisk problem

2007-10-11 Thread Alejandro Facultad
I'm using wengophone 2.2 for Linux and I log into an Asterisk SIP server. I have configured the Asterisk to permit canreinvite=yes so SIP control packets go among clients and Asterisk and RTP data packets go among clients directly (bypassing Asterisk). But when I establish a call between two