search for: vvvvvc

I used the command asterisk -vvvvvc to see console messages and it works fine. Now, I want to turn off this feature. How please?

...good documentation available, I was able to install and configure the hardware as well as software. As specified in the Getting Started with Asterisk Guide, I first installed zaptel driver, then libpri and finally Asterisk. All of them successfully compiled, but when I execute asterisk # asterisk -vvvvvc it doesnt start up, after printing plenty of debug messages it finally quits with giving the message Can't open file /dev/vpb0 or /dev/vpbph0 These drivers were installed in the system for the voice tronix cards, which we trialled previously. They (voice tronix cards and driver) are no longe...

...;SIP/3002-c418' If I do a "ps ax" to see what's running, I see: 16243 pts/1 S 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 For-You. If I go to the * console and listen to "For-You", it plays through the speakers just fine. However, after starting * with -vvvvvc, I see: [chan_oss.so] => (OSS Console Channel Driver) WARNING[1074504864]: File chan_oss.c, Line 974 (load_module): XXX I don't work r ight with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.c...

...me-sound-recorder 2. Make sure your identity is configured in sip.conf and extension.conf correctly 3. Make sure kphone is registered with Asterisk File->Identity - see whether 'Unregister' is there, (means you are registered ) 4. Watch for Asterisk Messages for any clue. ( asterisk -vvvvvc ) 5. Make sure the destination extension you are dialing from kphone has proper dialplan sequence in extension.conf 6. If you have OSS sound configuration, immediately switch to ALSA. - visit alsa-project.org and search docs for your card type. Compile and install the packages. ( this OSS wo...

...talled Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition bookThe installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu at vbefe01:/etc/asterisk$ sudo asterisk -rubuntu at vbefe01:/etc/asterisk$ sudo asterisk -vvvvvcubuntu at vbefe01:/etc/asterisk$ sudo asterisk -cvvvubuntu at vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump$ asterisk -vvvvvvcIllegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks....

Hi, I am using asterisk CVS 2004-06-16 with oh323-0.6.3a I have a strange problem if I start asterisk with oh323 loaded /usr/sbin/asterisk -vvvvvc once I am in the console and issue "restart now" or "reload" asterisk hangs and it not stoping or restarting at all, below is the console logging when it happens, as you can see it stucks on "Destroying any remaining musiconhold processes" [chan_oh323.so] => (Ope...

...occurred when I was re-registering multiple endpoints one after another as fast as I could do it manually (by operating the respective phones) and at the same time tried to make calls from these phones. I have attached what was happening before the segfault in the CLI; I had started Asterisk with -vvvvvc. Unfortunately, I can't reproduce the situation, so if we want to know the cause of the problem, I guess it's necessary that a developer analyzes the attachment and the respective part of the source code. I think I am not (yet ?) so deep in Asterisk that I could help debug the problem, but...

Hi all, Anyone have any great ideas on getting asterisk 11 to work with pulseaudio? I chown user:group /var/lib/asterisk /var/run/asterisk /var/spool/asterisk /var/log/asterisk /usr/lib/asterisk /etc/asterisk change user and group in asterisk.conf when is run: su user -c "asterisk -vvvvvc" alsa.conf reports malsa_card_init^[[0m: snd_pcm_open failed: Connection refused What am I missing to get asterisk 11 to run as a user? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachm...

Hi, I have configured tdm400p alone and it works fine for me. But now i tried to add 2 x100p in the same machine with the following configs, but asterisk refuses to load, can someone guide me what i have done wrong. Following r the configs and output. TIA, Azher >> ./asterisk -vvvvvc [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXS Kewlstart signalling -- Registered channel 2, FXS Kewlstart signalling WARNING[16384]: File chan_zap.c, Line 600 (zt_open): Unable to specify channel 3: No such devic...

...text/plain; charset="iso-8859-1" hello all, I have just installed Asterisk@home version 2.1. It keeps working fine for couple of days. But after couple of days I start getting the following error as the Asterisk does not start automatically so I try to start it with "asterisk -vvvvvc". Any ideas? and how to fix this error. Thank you for your help. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-auto.conf': Found == Parsing '/etc/asterisk/zapata_additional.conf': Found [root@asterisk1 ~]# Ouch ... error whil...

hi, i use agi to send message back to Asterisk by STDERR, but why i could't see the message in asterisk CLI? i start asterisk use " asterisk -vvvvvc" in order to see all message. Thanks -- Best regards, Sucan

...t; The only thing I see is: [ Context 'parkedcalls' created by 'features' ] '700' => 1. Park() [features] -= 1 extension (1 priority) in 1 context. =- extensions.conf has the MANY contexts in it. When I run "asterisk -vvvvvc" I see the line that its loading extensions.conf and no errors reported. Where is my dialplan? Thanks, jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140114/58ee168f/attachment.html>

If I login as a user and run asterisk it connects to pulse and runs. if I login as root and run the command su user -c "asterisk -vvvvvc" it does not connect to pulse and run. I thought su ran as that user - am I missing something to get the su command to run correctly and connect to pulseaudio as that user. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium....

I'd like to start playing with 1.8, however I don't want to potentially damage anything on my existing 1.6.2 install on my production server. I'd like to test 1.8 against my existing configs leaving my 1.6.2 install untouched. Looking at the output of ./configure --help suggests that it's possible to install Asterisk into another prefix of my choosing, but as this is

I just loaded a new asterisk install (1.4.19) and found that the sip, iax, and extentions commands are missing from the cli and are not listed in help either? Any idea what could have happened or where these commands may have gone? Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 ~ I would like to : set up a server on Linux on which my friends can connect with msn or netmeeting, suporting at least sound conferance, and optionally video, but I dont want asterisk server to lock up the sound card; and then, I want to be able to connect that server with a free Linux tool; I had a look at http://www.voip-info.org/wiki-Asterisk but

I'm just wondering if any one else has tried to successfully install Asterisk on Ubuntu inside VM. I've installed Ubuntu without incident or error. Even the install of Asterisk is relatively straightforward as it is maintained in one of the repositories. But when I attempt to start Asterisk I get a nice Segmentation Fault. I've narrowed down the problem somewhat. If I disable modules

Hi all, I have installed FAX app as described in several mails. When a fax call is received, I get the following in the * console: -- Starting simple switch on 'Zap/1-1' -- Executing Answer("Zap/1-1", "") in new stack -- Executing Ringing("Zap/1-1", "") in new stack -- Executing Wait("Zap/1-1", "2") in new stack

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found...

Hello All, I am having an odd problem with Armin's chan-capi_cm on builds higher than 7955. It would seem that this happens on anything higher than 7955. What is happening is the isdn is ringing, then asterisk does a goto-if and just hangs. Asterisk itself is ok, but the isdn then rings out or busys out on the other side. Outgoing works fine, this only seems to effect incoming. I