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...wing 'top' show ; 5737 root 25 0 3832 2588 2424 R 99.7 0.5 2:54.00 mpg123 5397 root 15 0 167m 18m 6212 S 2.7 3.7 0:10.95 X 5627 root 15 0 47988 6604 3624 S 1.3 1.3 0:15.64 asterisk 4673 root 17 0 8588 6820 1620 S 0.3 1.3 0:02.05 hald 5509 zenkato 15 0 24168 9.8m 7068 S 0.3 2.0 0:00.47 metacity 5551 zenkato 15 0 58500 17m 10m S 0.3 3.4 0:01.20 gnome-terminal ....(snip)... So, I have to do 'kill -9 pid-of-mpg123'. * goes to segmentation fault. Is this safe way to stop asterisk? When UA could not send 'hangup...

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

...usecount(void) { int res; STANDARD_USECOUNT(res); return res; } char *key() { return ASTERISK_GPL_KEY; } ------------------------------- Is this "broken pipe" means that I don't have "stdinpcim" on /dev? My system has "stdin" but no "stdinpcm". [zenkato@p4c8 dev]$ ls -al|grep stdin lrwxrwxrwx 1 root root 15 8 27 2005 stdin -> /proc/self/fd/0 Sorry I posted this on 'asterisk-users' ml, but there is no response there. So, I posted this on this ml. Regards, Zen Kato

..., but rxfax doesn't work. After some FAX sounds, it hangup! Could someone tell me how to debug? The following is the * CLI> log to 192.168.0.161:43222 -- Executing NoOp("SIP/4881-bde9", "") in new stack -- Executing RxFAX("SIP/4881-bde9", "/home/zenkato/voip/asterisk/fax/tif/send22.tif") in new stack Sip read: ACK sip:4883@192.168.0.3:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK3d3700001a29ffff From: <sip:4881@192.168.0.3:5070>;tag=bdf000008f360000 To: <sip:4883@192.168.0.3:5070>;tag=as4090e42f Contact: <s...

I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz and referenced http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices. But I could not succeed to start ices-2.0.1 as follows; -- Attempting call on Local/33102@stream for 33100@stream:1 (Retry 1) -- Executing Answer("Local/33102@stream-3fa5,2", "") in new stack > Channel

Hi, When we use svn branches-1.4 such as: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 # svn checkout http://svn.digium.com/svn/libpri/branches/1.4 libpri-1.4 how to write the others such as dahdi-linux and dahdi-tools? Regards, Zen

Hi, I input "0203#" after "mailbox?" voice prompt from Voicemail cmd on extensions.conf such as exten => 0021,1,Ringing exten => 0021,2,Wait(1) exten => 0021,3,Voicemail exten => 0021,4,Hangup *CLI> -- Executing [0021 at sip:1] Ringing("SIP/0103-09a308b0", "") in new stack -- Executing [0021 at sip:2]