similar to: Can't connect to Asterisk cli

Hello, While connecting to my voicemail, I noticed that Asterisk may perform some tasks isn't included in the options levels that currently played! For ex: I am listening to the main menu which asking me to press 1 for new messages, 2 change folders, 0 mailbox options which is OK for these numbers, but if I pressed 7 by mistake for example it will tell me "Message deleted" which it

Hello,For the time being I am using the following line to play the original saved message by Asteriskexten => 7001,n,Playback(vm-nobodyavail)Now I am trying to use the other features for Asterisk's voicemail. I have recorded a message, and I can see it saved on the system, but still Asterisk keeps playing the original message... Is there something I can add to let the subscriber plays his

Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see

Hello, I wonder if it is configurable possible to add a new menu demo to run within voicemail context dialogue! I want to run am interactive menu before or within the normal voicemail dialogue to run a script based on the subscriber selection. My point is to get use from the authenticated password that provided by user to go though his voicemail to access this new feature as well! If not, is there

If asterisk is running as a daemon (started with no command line options), can I reconnect to it and get the vvverbose info in the console? In other words, will asterisk -rvvvc work? Sorry to ask.... I would just try it to see if it works, but I don't have an installation handy and have to do some remote troubleshooting soon. ThanksE+06. D. __________________________________ Do you

Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [cybercall at ip-208-109-177-212 ~]$ asterisk -vvvvvvc -bash: asterisk: command not found [cybercall at ip-208-109-177-212 ~]$ I checked modules and other configuration files which are installed correctly. Please help me

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

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong

Hello, I am trying to use postfix/dovecot as mail server to be the imap storage for my voicemail system.For that I installed postfix and dovecot and trying to follow the instructions in this post http://etel.wiki.oreilly.com/wiki/index.php?title=Storing_Voicemail_on_an_IMAP_server&printable=yes I should add a master user for Asterisk to your IMAP server that has access to all user's

Hi all, i'm trying to install Asterisk with Junghanns QuadBri ISDN card: I have followed all the instructions, and when I run "|ztcfg -tvv" I get | | Zaptel Configuration ======================| SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02)

Hello, I am trying to get MWI working after integrating Asterisk with CCM.I have followed the instructions in http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+IntegrationMy problem is that I don't see externnotify's script being called at all in the logs, and not sure if I miss something here! In Voicemail general I addedpollmailboxes =

Has this been solved? When I park a call, the caller hears a second of music on hold and then the whole system crashes. I can restart with a simple (asterisk -cvvv), I don't have to reboot or anything John

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [root@server2 akrall]# ps -ax|grep asterisk 20555 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ? S 0:00 asterisk -vvvg -c 20558 ? S 0:00 asterisk -vvvg -c 20560 ? S 0:00 asterisk -vvvg -c 20561 ? S 0:00 asterisk -vvvg -c 20562 ? S

When Asterisk first starts up, it will attempt to "bring up" the B channels on any PRI circuits. If you are using A@H then you can log on to the Asterisk CLI (asterisk -r) and then do "stop now" to stop asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux command line. You should see a bunch of messages on the terminal and then you'll get the Asterisk

I know you have been busy with mongrel and other projects. Is ther a test image available anywhere for Rubuntu? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060619/e09588a0/attachment.html

Hello; I am installing asterisk and dahdi on ubuntu "and I used my username bghayad to login for ubuntu and do the installation, actually I feel my problem is related to the?username and permission but I am not able how to fix it", I am facing now mainly the following two problems: The first one, asterisk is not starting automatically although I did sudo make config (for asterisk and

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 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

I installed the TDM400P and installed it on a system running on Fedora Core1, please refer to the steps below. I connected an analogue phone to the FXS port. When I pick up the speaker I don't hear the dialtone although when I press a number key I hear a DTMF generated. What could be the problem? I appreciate your help 1- First I installed the TDM400P in the PCI slot and connected