similar to: how to configure asterisk user and group rights

I'd like to these three things about asterisk: 1. How the asterisk program can be configured to run as a different user from root. 2. what directories and files it must have read and right access to 3. Setup an asterisk group, which also has some of the rights the asterisk user has rights to, and what else it can be used for. Are these some info sources which go into these areas in depth?

Is there a key sequence to stop a call as its ringing, before the call is answered? The * key stops a call after it is answered, but I'd like a way to cancel the call during the ringing phase. /Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

I have been trying to connect via sip and things don't seem to work. What do messages like this mean? Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834 (len 361) to 216.127.66.119 returned -1: Invalid argument Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102

My CDR is displaying wildly inaccurate results. When I make a call the CDR records the time between connecting into the server and hanging up, instead of recording the time between dialling from the server to the PSTN destination via VOIP termination. It is alright to log the duration of the connection to the server, but why it does not log calls for termination via voip provider is the main

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because you current number it is not being answered, and you don't want to hangup before dialling again. /Obelix

I have compiled the OH323 module for my system. When can I find some info on how to properly configure it? I haven't read any info for its configuration, and I need some starting info. Were do I start? Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. /Obelix

Which Actions and events to the read/write options in manager.conf give access to, ie the options below. read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Are they documented somewhere? /Obelix

I am running a prepaid application with Asterisk. When authentication has to be done by DTMF everything works fine. However when the user is authenticated directly from the sip phone, the channel variables seems to disappear. Trying to retrieve the channel status always returns -1 instead of the 6 that happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME variables. The

I am interested in how much CPU and RAM asterisk requires for call handling. 1. What is the minimum CPU required for asterisk to manage 60 concurrent calls without transcoding. 2. Handle calls on a 75% no transcoding, 25% transcoding 3. How many calls can it connect per second ie from one VoIP -> VoIP? All the above refer to a VoIP setting. 4. Is there a difference between bridging and

How does TDM work, how do you connect to it? I have the impression it can't be routed like ethernet, but a cable from your switch has to be plugged into the providers equipment. I have seen the Asterisk info about TDMoE - does this mean that the Asterisk card will modulate the signal on the Ethernet cable to allow it plug directly into a proper TDM connection? Will someone please enlighten

How does one check what codec translations are in use in a call? I am connecting to sip system which says 488 "4XX Not Acceptable Here". I don't know what is stopping the call from being accepted and I'd like to know if there are codec issues involved. /Obelix ---------------------------------------------------------------- This message was sent using IMP, the Internet

My firefly softphone is having problems connecting via SIP. When I set it up, one provider does appears to connec, but trying to call results in a 'Couldn't start call' The other responses with a 401 failure code. Xten connects okay via SIP. Is there something about Firefly SIP configuration that I don't know about? IAX connects okay / Obelix

My Asterisk PBX seems unable to receive DTMF information via SIP. I have tried all the various methods, rfc2833, inband and info and they all don't seem to work. IAX2 works fine. Is there something I must be missing ? /Obelix ---------------------------------------------------------------- This message was sent using IMP, the

Hello, I use pam_winbind for users authentification and pam_mount for mounting homedirs stored on w2k3 fiel server. I still have two more problems to solve: 1. I?m not able to set proper access rights on homedir. I want to set them to 700 but after mounting the homedir access rights are set to 777. I used to use pam_mkhomedir umask=0077 attribute to set proper rights, but this have no effect

Hi! I have some problems with the download of Asterisk-addons. I try to follow instructions that I found in http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql , but nothing to do. This is my shell: [root@obelix root]# cd /usr/src [root@obelix src]# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot [root@obelix src]# cvs login Logging in to

Hi, I see that currently physical register uses marked as "undef" can still cause dependencies. Is this intentional? SU(9): %D5<def,undef> = LDrid %R0, 0, %R10<imp-def>, %R11<imp-def> # preds left : 0 # succs left : 11 # rdefs left : 0 Latency : 1 Depth : 0 Height : 0 Successors: ...

Hi all, I try to connect two asterisk-server together. There is a server (obelix) which receives a call. This call should be transfered to another server. In my dialplan at obelix I have the following: exten => 920622201,1,Dial(SIP/outbound:geheim at asterix.local:${EXTEN}) exten => 920622201,n,Hangup exten => i,1,Congestion exten => t,1,Congestion If I call the number 920622201

SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC "Keep in touch with the World" Hello, The next Asterisk Users Group meeting has been scheduled for this Saturday March 11th at 11:30am. Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online

I'm trying to get running an ethernet with two machines: a samba server running suse linux 6.4 and a client running windows 98. I follow the steps in the DIAGNOSIS.txt file, but I get stucked when executing 'net view \\SERVER' in the client. I have laready done everything it says in that file: 1) Set a lmhost file in the client. Just with this line: 192.168.0.1 ASTERIX #Asterix is