similar to: Xten Lite Build 1079

I've had a few problems with my system holding the line after a call has been made, just now I rebooted and noticed the following in /var/log/messages Aug 13 17:23:15 Sheriff kernel: wcfxo: DAA mode is 'FCC' in the file wcfxo.c the following structure is initialised as below which would suggest that FCC is wrong for France or pretty well all of Europe. static struct fxo_mode {

I don't think I've ever seen so many commits to the CVS, I'm not complaining. To save lots of checkout/test cycles can a message be posted when the current surge is over? -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 <http://www.linuxautrement.com> IAX 17004902330 FWD 42651

Hello All, I am getting a set of errors when I boot Asterisk that I have not been able to solve. What is causing these error(s)? Asterisk boot output: ============== Asterisk CVS-04/10/04-21:44:51, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= [

I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said "Registration error". Why would phones loose registration to asterisk when the internet connection

Hy!     I''ve been trying to make Xen 2.0 work on a compac evo N800C laptop. But I am getting an error that say, that there is no more memory for the kernel and then the message Aiii cpu is toast, then re reboots in 5 sec. I looked into the archives and didn''t found a clue that helped me much on this one. >From what I can see from the output is that he dont see the

From the bend me over news department. 2 March 2006 Mr. Leonard Katz Executive Director Broadcasting and Telecommunications Canadian Radio-television and Telecommunications Commission Ottawa, Ontario K1A 0N2 Dear Mr. Katz: Associated with Bell Canada Tariff Notice No. 6929 1. Attached for the Commission's approval are proposed revisions to Bell Canada's Access Services Tariff Item

Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip - NS Cx5530a Southbridge National Semiconductors SC2200 - NS PC97317 in chipset - 32MB Compact Flash - 64MB Ram - 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ide>flash

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

Hi all! I get this error when trying to start asterisk: ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk What can be the problem? Thank you! Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 UK +44 870 - 3403539 FWD 64662 sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part

Hi, I could try latest 7zip (4.57) on a triple boot machine (Debian SID i386, x86_64 and native Windows XP SP2), and found something surprising in the results : amd64 at 2400MHz linux32 -> 1814 linux64 -> 1946 win32 -> 2004 So performance is much better in Linux 64 bit than in 32 bit, while both are slower than native windows system. Is it normal, I mean does anyone else obtain

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

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> -----Original Message----- > From: listas iPfone [mailto:listas@ipfone.com.br] > > Somebody knows why asterisk gives me that error wile playing .mp3 files? > > The files play well but the message aperas any way: > WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of > 4 > bytes) (No such file or directory)! Listas, You might try down-sampling

When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If

Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization... I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

After apt-get update && apt-get upgrade -y wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz tar xfvz chan_capi.02.5c.tar.gz cd chan_capi-0.2.5c make && make install shutdown -r now asterisk seg faults upon calling in via ISDN. Any ideas are greatly appreciated. rgds pos

Hi everyone, I know someone makes a product that's a POTS phone to SIP converter, where you just plug your POTS phone in one side and the network cable in the other. Has anyone successfully used any of these with Asterisk, and if so how expensive were they? I ask partly out of frustration with the FXS cards but mostly because it would make installation MUCH easier for what we're