search for: rvvv

...achine names changed to protect the innocent :-) firewall allows ssh connections if inititiated from I to O, but not if the other way. both machines have an /etc/rsyncd.conf of: [rt] path = /tmp/rsync_test comment = Test area O runs rsync daemon, I initiates a rsync cammnad like rsync -rvvv --delete --rsh=ssh O::rt /tmp/rsync_test which works great, but when flipped and run on machine I, like so: rsync -rvvv --delete --rsh=ssh /tmp/rsync_test O::rt we get rsync -rvvv --delete --rsh=ssh /tmp/rsync_test O::rt opening connection using ssh O rsync --server --daemon . root@est.llnl.gov&...

Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with "asterisk -rvvv". I need it in debugging purpose for tracking some bug. Thanks Enrico. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3473 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/...

...rience. And that bug was fixed.<br> <br> Every time I start Asterisk (just by issuing /usr/sbin/asterisk), the bash console text turns white. I'm using rxvt, so this makes everything pretty much invisible. If I login into the Asterisk console (asterisk -rvvv) - the text turns black again. This is not critical, but quite annoying. I've experienced it both with 1.6 and 1.8 installations. Maybe, just maybe this is a problem with rxvt - but I use it for absolutely everything on the command line - and no other application has pro...

...the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear, while the speaker at the internal side hears perfectly. CPU usage is quite low, and asterisk -rvvv does not show anything particular Any suggestion

Hi, I am using asterisk 11.1.0. How to display the caller number (from asterisk -rvvv terminal) in the first step of the extension (before doing any action) ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130417/8e6cbb1a/attachment.htm>

>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel (you can also use 'w' to produce .5 second

...er-directory filters are skipped when deleting under certain circumstances (options -rFR --delete, more than one sync source, rsync 2.6.9 and 3.0.0cvs). This eventually leads to a segmentation violation (double free). To reproduce: $ mkdir -p dst a/aa/aaa a/ab b/bb/bbb/bbbb $ rsync --delete -F -R -rvvv a b/bb/bbb/bbbb dst/ $ rsync --delete -F -R -rvvv a b/bb/bbb/bbbb dst/ While the first rsync succeeds, the second run aborts (--delete-before and --delete-after exhibit the same problem). Here's the log of the second invocation: building file list ... [sender] make_file(a,*,2) [sender] push...

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the

...# wget http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.6.0/x86-32/codec_g729a-1.6.0_3.0.3-opteron.tar.gz # tar xvf codec_g729a-1.6.0_3.0.3-opteron.tar.gz # cp codec_g729a-1.6.0_3.0.3-opteron/codec_g729a.so /usr/lib/asterisk/modules/ # asterisk -rx "restart now" # asterisk -rvvv Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License...

...igium.com/pub/telephony/codec_g729/asterisk-1.6.0/x86-32/codec_g729a-1.6.0_3.0.3-opteron.tar.gz > # tar xvf codec_g729a-1.6.0_3.0.3-opteron.tar.gz > # cp codec_g729a-1.6.0_3.0.3-opteron/codec_g729a.so > /usr/lib/asterisk/modules/ > # asterisk -rx "restart now" > # asterisk -rvvv > Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for > details. > This is free software, with components licensed under the G...

Hi All; I installed Asterisk on a new Server, it is a Dell Server and has 4 Ethernet ports. I gave IP address 192.168.0.3 for one Ethernet port. I am able to login for asterisk using /usr/sbin/asterisk -rvvv and from there (in the command line) I can type a commands. I have an Polycom IP Phone that is able to register for other Asterisk boxes (and some of them is 1.8.3) but with this new server, I do not see any messages coming to the consol when I give the IP address of this new asterisk server !!...

...tions = h I c a delete [temp] comment = WSS Polling TEMP Main path = /cygdrive/c/temp exclude from = temp_exclude write only = true transfer logging = yes ...in the "temp_exclude" file I have: + /*.zip + /*.xfo + /*.trg - /* ...I''m calling rsync from the client with: rsync -rvvv /client/temp/ 192.168.1.10::temp ...inside of the local "/client/temp" directory I have serveral zip files but for some reason it skips them? The rsync.log, I get, "skipping server excluded file..." and the list of the zip files? I want to be able to send to the server the...

Hi ALL, still having trouble getting Ast 12 to run. I got it compiled and built but now when I try to run, I'm getting a missing library error that seems to be in error (see below). The .so file DOES exist with correct permissions. [root at Asterisk12 ~]# asterisk -rvvv asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory BUT: [root at Asterisk12 ~]# find / -name libasteriskssl.so.1 /usr/lib/libasteriskssl.so.1 /usr/local/src/asterisk-12.0.0-beta1/main/libasteriskssl.so.1 [root at Asterisk...

Hi, I just did a test install of Ast 11, and have trouble getting the same logging information that Ast 1.x provided. I'm looking specifically for the logging around call progress / dialplan actions. In ASt 11 I've done the same thing that I did before: core set verbose 60 I also tried overwriting the logger.conf with the distribution one from Ast 11, and setting option

What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI> A No such command 'A' (type 'help' for help)

Hi i generate a call from the dialplan in this mode: exten => 1002,1,Answer() exten => 1002,2,Dial(SIP/user at host) the call is generated, but after some seconds it is interrupted, here the asterisk log: *CLI> -- Executing Answer("SIP/host1-0819d0d0", "") in new stack -- Executing Dial("SIP/host1-0819d0d0", "SIP/caller at host") in new

Our office does not have a PA system, and in our current phone system we have a certain extension that we dial that pages over the speaker of all the phones in the office. Does Asterisk support this feature? If so, could someone tell me the best way to set this up in AsteriskNOW? Thanks, Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ====ok inbound and outbound the calls between x-lite and snom300====> ok inbound and

Hi all! I'm new to asterisk and I'm trying a simple config with: - Debian GNU/Linux (unstable) - last version of Asterisk - a X100P card I have a problem with dial out from a SIP software phone (XLITE) to a public number (ex. my mobile phone), asterisk start the call, but nothing happen... If I run "ztmonitor 1" I can see the right RX level and if I try to make a call with an

...lec DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) I try to change to /etc/dahdi/system.conf to fxsks=1 echocanceller=hwec,1 Still doesn't work and this error still occured: DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) [root at callcenter dahdi]# asterisk -rvvv Asterisk 1.8.7.0, Copyright (C) 1999 - 2011 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s. This is free software, with components licensed under the GNU General Public Licens...