**** BEGIN LOGGING AT Thu Feb 01 02:59:57 2007 Feb 01 11:02:19 can somebody help, i can't find glibc-devel on ipkg and i believe it's needed to compile Feb 01 11:02:30 is it in some other package? Feb 01 11:16:09 ok believe i found the missing package, libc6-dev Feb 01 12:26:52 indipendent is better ;) Feb 01 22:56:23 joshin: ping Feb 01 22:56:36 rwhitby: sir yes sir? Feb 01 22:56:54 saw a couple of questions from you in the logs over the last couple of days Feb 01 22:57:10 (but of course now I forget what they were) Feb 01 22:57:41 Was wondering about the kernel and why it went from what looked like a .19+.20 hybrid back to just .19 Feb 01 22:57:52 right - that was one. Feb 01 22:58:08 the .19+.20 seemed to cause a problem with the feeds Feb 01 22:58:10 dunno why Feb 01 22:58:21 so I went back to .19 Feb 01 22:58:34 if you want the other, you can set preferred_version Feb 01 22:58:48 Yeah, for now I'm just sticking with the default. Feb 01 22:58:57 I think in general we will have slugos track the latest released version, rather than -rc candidates. Feb 01 22:59:51 I'll probably need to add a serial port to figure out the crashes I've been having. When under load, my slug seems to lose either its ability to talk via ethernet and/or the USB attached disk. Feb 01 22:59:54 Anyone here know what kernel modules I need to load for sound support with a yealink usb handset? Feb 01 23:00:24 I can reliably crash it by building gcc4. Feb 01 23:00:38 Do you have lsusb installed? Feb 01 23:01:13 (I'm trying to test the new yeaphone package) Feb 01 23:02:47 They may be listed here: http://billlions.blogspot.com/2006/08/tesco-internet-phone-with-skypelinux.html Feb 01 23:07:56 kernel-module-snd-usb-audio is one Feb 01 23:08:23 and that covered all the yeaphone interfaces. sweet. Feb 01 23:08:46 and it now appears in linphonec sourcecard list Feb 01 23:08:47 soundcard list Feb 01 23:11:54 ok, yeaphone sees the phone. Now I just need a sip provider ... Feb 01 23:18:12 install openser and be your sip provider ;-) Feb 01 23:18:49 s/your sip/your own sip/ Feb 01 23:18:50 osas meant: install openser and be your own sip provider ;-) Feb 01 23:19:03 Its nice when something is actually easier than you expect it to be. Feb 01 23:19:42 well, it all depends on your expectations, right ;-) Feb 01 23:30:50 osas: will look into that. got a newbie guide for me? Feb 01 23:31:24 ohhhh .... that's a tough question Feb 01 23:31:32 bbiab Feb 01 23:31:35 go asterisk Feb 01 23:35:16 ga asterisk Feb 01 23:35:23 s/ga/go/ Feb 01 23:35:23 osas meant: go asterisk Feb 01 23:35:34 it is easyer to setup Feb 01 23:37:06 for a slug running slugos 4.x, which version would you recommend? Feb 01 23:38:39 both are pretty stable Feb 01 23:38:44 go with 1.4 Feb 01 23:38:54 run only SIP Feb 01 23:39:06 do you have a static IP? Feb 01 23:40:00 if yes, you can easily host clients outside your network Feb 01 23:40:36 my * has 5-6 internal clients and 10 external Feb 01 23:40:37 in one scenario (at work) I do, in the other (at home), it's dynamic but I do ddns at zoneedit.com for home.rwhitby.net Feb 01 23:40:56 which ports have to be allowed through the firewall (incoming)? Feb 01 23:41:24 if you have control over the fw at work (i.e. port forwarding), then do it at work Feb 01 23:41:26 * rwhitby does bitbake asterisk in OE Feb 01 23:41:42 which version? 1.4 Feb 01 23:42:06 I gave up trying to get 1.4 to work with bb ... Feb 01 23:42:25 there seems to be a 1.4 .bb file there ... Feb 01 23:42:32 maybe now. after they fixed several build issues it will be easyer Feb 01 23:42:36 really? Feb 01 23:42:46 and it is working? Feb 01 23:42:58 I am using only the 3.10 (stable) Feb 01 23:42:59 haven't built it yet ;-) Feb 01 23:43:10 that's why I switched to optware Feb 01 23:43:25 I'm running 1.4 on 3.10 slug (optware) Feb 01 23:43:36 woks pretty well :-) Feb 01 23:44:03 my relatives oversees are happy with free phone calls ;-) Feb 01 23:44:24 back to * setup Feb 01 23:44:33 I Feb 01 23:44:45 all you need is to port forward port 5060 to the asterisk box Feb 01 23:45:09 forward a bunch of ports for rtp traffic (see rtp.conf for range) Feb 01 23:45:53 the router (billion voip 7402vgp) has two voip phones on it already - is that going to conflict with running an asterisk server too? Feb 01 23:46:37 then configure asterisk to use an external IP (see sip.conf externip or externhost) Feb 01 23:46:45 it may Feb 01 23:46:54 bbiab (again) Feb 01 23:46:58 you can run asterisk on a different port Feb 01 23:51:22 bbiab Feb 01 23:51:32 * osas need to run some errands Feb 01 23:52:35 thx **** ENDING LOGGING AT Fri Feb 02 02:59:59 2007