search for: vvvvc

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]: chan_iax.c:4828 set_config: Unable to load conf...

I have no problems lauching asterisk from the command line . . . asterisk -vvvvc However, I'm trying to autostart on boot up, so I'm trying safe_asterisk When I do this, I get: Asterisk ended with exit status 127. Asterisk died with code 127. Aborting. I've looked through the script but can't find what the problem is. I'm running on RedHat Fedora. Than...

...a couple of previous posts on this subject, but with no posted resolution... I'm attempting to run * as a non-root user (asterisk), following the guidelines on the wiki: http://voip-info.org/tiki-index.php?page=Asterisk%20non-root I can run * as my new user with "/usr/sbin/asterisk -vvvvc" without problem. However, I'm unable to run * using safe_asterisk with my new user: Asterisk ended with exit status 127 Asterisk died with code 127. Aborting. safe_asterisk is trying to use tty9 and I have changed the ownership appropriately: # ls -l /dev/tty9 crw-rw-rw- 1 ast...

Hi, I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there. I know that the process is killed because when I start asterisk using the command asterisk -vvvvc it starts and then it exits and the word killed is wrote on the console. Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too. Now I used the strace command on asterisk and I can clearly see at the end of the strace the line : killed by SIGKIL...

helleo ,I have one wildcard installed on the server,and configured it.there are two modules which both FXOs .all configurations complete,and when we start asterisk use "asterisk -vvvvc",it shows asterisk ready,no warnings.afterwards when the sever connected to the PSTN in the one port which configured,and have a phone line connected to the other port then run "zttool",however it shows just one channel active.I can't understand ,why? I try to dial the PSTN numb...

ive installed asterisk but below is what am getting proces gets killed.please help [root at localhost sounds]# asterisk -vvvvc Asterisk 13.13.1, Copyright (C) 1999 - 2014, 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 Licens...

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

...installation on fedora core 2 with zaptel drivers installed ok. However I seem to have lost the .pid file in /var/run or even /var/run/asterisk. If I run 'asterisk' I get: Unable to open pid file 'var/run/asterisk/asterisk.pid' : No such file or directory If I run 'asterisk -vvvvc' it seems to run fine! Also I copied over the redhat service script, this will run first time however if I do a restart it will sometimes cause a kernel panic, which I don't think is that great! The kernel panic seems rather intermittent and I think its something to do with the zaptel dri...

Hi, I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time i run asterisk -vvvvc, i get the following error: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 segmentation error [ro...

Hi everybody! I've managed to compile the bristuff patch on asterisk from Junghanns.net. I want to run this on the quadBRI card built into the PC. As I said, driver and asterisk are compiling well, but "asterisk -vvvvc" bombs with this message: [pbx_dundi.so]Nov 24 15:57:53 WARNING[1076754432]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: ast_sign_bin Nov 24 15:57:53 WARNING[1076754432]: loader.c:423 load_modules: Loading module pbx_dundi.so failed! lcn-aste...

Dear all I got one more error my asterisk E1 card connected with avaya E1 card [avaya]-------E1-----[asterisk] i got this 2 error what is start asteris on consol mode asterisk -vvvvc [Jul 27 09:51:29] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:30] WARNING[737] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jul 27 09:51:31] WARNI...

I saw a few people mentioning they were running or trying to run asterisk on Xen. Last night I checked out v1-0, compiled, ran "make install; make samples", then started asterisk with "asterisk -vvvvc". Several modules refused to load giving this error: [chan_sip.so]May 22 13:48:41 WARNING[4308]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: __i686.get_pc_thunk.dx May 22 13:48:41 WARNING[4308]: loader.c:440 load_modules: Loading module chan_si...

...t: 5 NOTICE[12301]: File chan_zap.c, Line 4233 (handle_init_event): Alarm cleared on channel 94 (last line repeated for every defined channel) == D-Channel on span 2 up == Restart on requested on entire span 2 /usr/local/sbin/astrun: line 4: 329 Segmentation fault /usr/sbin/asterisk -vvvvc Ouch ... error while writing audio data: : Broken pipe vectra:~# Any clues ? zttool sees the channel status OK. Best regards, Florian

...t; > http://atrpms.net/name/asterisk-sounds/ > > http://atrpms.net/name/spandsp/ > > http://atrpms.net/name/libpri/ > > http://atrpms.net/name/zaptel/ on the Fedora list, added the repository to yum, and downloaded, installed, then typed: # asterisk -vvvvc , hit the return, and a bunch of stuff happened, before returning to the root prompt. My first goal(s) is to be able to configure the machine to make a PC to PC call to my daughter, who lives in Minnesota. If all goes well, I can set up her computer to receive the call, using Asterisk. Is this...

...tmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there. > > I know that the process is killed because when I start asterisk using the command asterisk -vvvvc it starts and then it exits and the word killed is wrote on the console. > > Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too. > > Now I used the strace command on asterisk and I can clearly see at the end of the strace the lin...

Hi, I'm new to asterisk and try to get it work with capi.so. When I try to start asterisk with "asterisk -vvvvc" I get the following errors. I couldn't find any hint on the net what may be wrong in my configs. Has anybody got a hint? Here is the error output: [capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Jun 23 12:04:33 NOTICE[1075622560]...

...lled. Running with no digium hardware (yet) and no soundcard in asterisk box. Problem is using the sample configs with a sip phone added as follows [2203] type=friend username=2203 secret=2203 host=dynamic defaultip=192.168.0.2 dtmfmode=inband canreinvite=yes the console on * when running with -vvvvc says :- (whenb trying to dial extension 500 (the demo) from my xten sipphone) -- Executing Playback("SIP/2203-2e99", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum...

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

Hello! I am trying to install a TN405P on a P4-3GHz-HT machine running Debian Sarge with kernel 2.4.27. When I start Asterisk in -vvvvc mode it always shows == D-Channel on span 1 up == Restart on requested on entire span 1 == D-Channel on span 3 up == D-Channel on span 2 up == Restart on requested on entire span 3 == Restart on requested on entire span 2 == D-Channel on span 4 up == Restart on requested on entire span 4 -- B-ch...

...Zap/g2/${EXTEN}) H323.conf [general] port = 1720 bindaddr = 192.168.69.1 tos=lowdelay disallow=all allow=g723.1 allow=gsm -------------------------------------- I can pick up the phone and ring 099 and echo works but if I dial anything else I just get a busy signal with no errors on asterisk -vvvvc, what I need is for ANY incoming calls to make the analogue phone ring. Outgoing calls that fit the rules use h323, everything else should pick up the PSTN line and dial. I again apologise for the mess and newbness (did I just invent a word), I just need a kick start and get the basic stuff worki...