similar to: -x option inoperative with "bind" mounts

Hello, I use rsync for incrmental backups in the following way: rsync \ $EXCLUDE \ -e ssh \ --archive \ --numeric-ids \ --compress \ --relative \ --one-file-system \ --cvs-exclude \ --delete \ --delete-excluded \ --bwlimit=0 \ --backup \ --suffix='' \ --backup-dir=$BACKUP_ROOT/$BACKUP_HOST/${TODAY} \ $BACKUP_HOST_AND_PATH

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? Thank you.

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

Hi, This used to work fine in 1.4: exten => 2131/,1,NoOp(reject3: ${CALLERID(num)}) exten => 2131/,n,Playback(no_unknow_callerid_here) exten => 2131/,n,Hangup And now, after upgrading to 1.6.1.x it matches every callerid. Did something change? Thanks,

Hi, Is there a way to have markdown automatically convert obvious (http, mailto) URL's to links? i.e: http://example.com -> <a href="http://example.com>http://example.com</a> Thanks, -- http://www.critikart.net

Hi, If several phones register to the same sip.conf section what will happen with a "Dial SIP/shared" in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy!

Hello list members! I joined this list on May 1st and I can see I have a lot to learn. Thanks to everyone - especially those *@samba.org folks. You are so appreciated. Background in my plea for help: I'm the IT guy for a family owned community bank in Southern Colorado. I am very tired of the limitations and security problems of Micro$oft and have committed to learning and using Linux

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

Hi, In our backup script we sometimes would like to override the common (i.e: static) excludes filter list. For example we exclude "- *.ext" for all backups but would like to include "+ *.ext" only for 'local' backups. Are such entries supposed to cancel each other? How can one override an earlier exclude in a filter list? Thanks,

Hi Everyone, I''m trying to figure out how to do SQL queries in the Controller. I know this is not recommended, but I can''t use ActiveRecord because I am dealing with thousands of a different tables, one for each user, and I don''t think AR can be made to handle that situation. You can assign the table name; but that''s a class variable so any particular

On Mon, 2019-05-13 at 08:28 -0400, Bee.Lists wrote: > Hi folks. Just wondering how I can implement an automatic .bash_profile for root. I have to load my > user .bash_profile every time I get into root, and I would like a better solution. There is no /home/ > for root, so I?m a bit confused if this is even allowed. > > Any insight appreciated. > > > Cheers, Bee >

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

If like me you run * on a VPN (or multihomed) gateway and want to serve remote SIP clients, make sure you have bindaddr = 192.168.0.1 ; or whatever is your box's private IP otherwise * might bind to its public IP and send it as return address in the SIP call setup, which will (should) be rejected by your firewall. To * experts: might this setting interfer with NATed SIP clients? -- I

Hi, After upgrading to 1.5.2 I no longer can directly access to my voicemail by pressing the "Message" button, I have to go through the "urgent,new,old" report first. The oneTouchVoicemail parameter is set to 1 but not taken into account apparently. Anyone noticed that problem?

Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea?

Hi, In the samba debian package there used to be a very convenient utility to scan networks for open samba shares and weak passwords. Its name AFAIR is nat. Where can one find this tool? Is it still maintained? Is there a better alternative? Thanks, -- THERAMENE: Il veut les rappeler, et sa voix les effraie ; Ils courent. Tout son corps n'est bient?t qu'une plaie.

Hi, It seems rsync no longer resumes partial transfers after a SIGINT (CTRL-C). I tried the following: % rsync -avzP ~/video/Gone_In_60_Seconds_-_DivX.avi 192.168.0.3:/backup/DivX building file list ... 1 files to consider Gone_In_60_Seconds_-_DivX.avi 262144 0% 10.34kB/s 19:40:22 [CTRL-C] [testing remote size] % rsync -avzP 192.168.0.3:/backup/DivX/Gone_In_60_Seconds_-_DivX.avi