search for: rvvvvv

...nning fc3 with current cvs-head... Is there a nice way to rotate the /var/log/asterisk/messages file without shutting down asterisk? I'm currently rotating the log files via cron, however my script requires asterisk to be shut down, which also kills any outstanding cli sessions (eg, asterisk -rvvvvv). Would like to rotate the files without killing the cli session. Any reasonable way to accomplish this? Rich

...and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the console or an xterm, the text appears correctly. Does anyone have any ideas what is causing this and how to fix it? Thanks Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

...recieve calls from the extension I am using. I didn't think this sounded like an AMP problem as the conf files has the entries in there. I looked in the astrisk "full" logfile but saw no errors pertaining to this nor did I see anything in the console when I connected with asterisk -rvvvvv and did a sip debug ip 192.168.5.114:5060 (my extensions's ip and port). Where should I begin with troubleshooting?

Ever since upgrading to 1.2.8 I've been getting occasional WARNINGS that say: WARNING[4356]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed. Is there something I need to fix or is this a benign message? Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.8.0/352 - Release Date: 5/30/2006

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 a...

Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI

...d installed it on a RedHat 9 box. After compiling just fine, running like a champ in tests, and having my extensions.conf configured to taste, I went ahead and did a live beta test this past weekend. The phone system stopped responding 3 times. The first time, I got into the asterisk console, vi -rvvvvv. I didn't see anything too out of the ordinary, except for lots of messages that all of the lines were busy (even though no calls were up and I have 2 PRI's). I thought it odd, but since I had been tweaking a config file or two, I didn't think too much about it. I just stopped * and...

...sole when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rvvvvv): == Using SIP RTP CoS mark 5 -- Executing [xxxxxxx at phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "1xxxxxxxxxx,1") in new stack -- Goto (phones,1xxxxxxxxxx,1) -- Executing [1xxxxxxxxxx at phones:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "oldcid...

..."Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" when it attempts to stop asterisk. Then I > run amportal start and it says it is already running (obviously if it never stopped). Also when attempting to access the asterisk > console as root, 'asterisk -rvvvvv' for example, I get the same message. I su to the asterisk user and get the same message. > > On a probably unrelated note, if I attempt to start asterisk via '/etc/init.d/asterisk start' (how I had it setup before > installing freepbx) it starts then stops, exiting error code...

...outgoing calls from an office snom 190 phone to my phone beside me on the desk. A timeout of 3 secs for POTS or 15 secs for my mobile guaranteed that no voicebox would take over but I heard a short ring when calls got through, to add a real life ringtone to remote visual feedback from asterisk -rvvvvv. httpsnom-dialtest ------------------------- #!/bin/bash # Created 070107 by AvH # $1 is the extension to dial if [ "$1" = "" ] then echo enter number please ; exit fi # command for snom 190 phone, taken from # http://80.237.155.31/kb/index.php?View=entry&CategoryID=21...

...of Astlinux and installed on a WRAP board and I'm > totally stuck! > > I'm using sipgate.co.uk for incoming calls, but when I make a test > call from the PSTN, the call just dies without connecting to my > Astlinux box. (I'm monitoring asterisk console via 'asterisk -rvvvvv' > and see nothing). > > I wondered if it might be a problem with Asterisk not listening > properly, or perhaps a problem with my home firewall. Would anyone be > kind enough to advise me as to where I may have gone wrong? > > Thanks, Chris. > > My sip.conf looks l...

Hi, i've just got 2 grandstream phones and when I try to connect them with * I get the following: -- Playing 'demo-abouttotry' (language 'en') WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for seqno 59134 (Response) I've seen there was some discussion on this already but i

http://bugzilla.mindrot.org/show_bug.cgi?id=1161 Summary: scp -r fails Product: Portable OpenSSH Version: 4.3p1 Platform: ix86 OS/Version: Cygwin on NT/2k Status: NEW Severity: normal Priority: P2 Component: scp AssignedTo: bitbucket at mindrot.org ReportedBy: gregt at post.pl This is on

hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vvvvvcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _________________________________________________________________ MSN Hotmail :

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also