similar to: What and When is the next version of Asterisk?

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

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

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.

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

> Hi folks, > My SMB server is a Helios Ethershare 3.1 on a Tru64 Alpha machine. > We are now integrating more and more gentoo-Linux Servers. These are the > SMB Clients. > > On Obelix (one of our gentoo-Servers) I can mount the SMB Volume from the > Ethershare Server. > obelix nc4smb # whoami > root > obelix # mount -t smbfs -o username=m.heckmann

I have a legacy pbx that I know nothing about. The PBX is a merlin legend system (now owned by avaya). I have found information on the list that I can use a 100D T1 card to bridge the PBX to an asterisk system. We are looking to head this route as well as replacing are AUDIX VM system with Asterisk. Has anyone done this that wouldn't mind sharing some of the merlin specific setup?

I am having a problem with the Vmail Web app and was hoping someone has some advice. I have been able to install vmail.cgi. I am able to bring up the vmail web page and login with out problems. However, I never see any vmails listed even when there are vmails present in the directory [root@linux-home INBOX]# ls -l total 40 -rwxrwxrwx 1 root root 255 Dec 2 14:45 msg0000.txt

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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am stuck in a situation where I have two te110p's and really need a te210P. Anyone interested in a trade? They are in working condition guaranteed. If this is the wrong forum for this, I am sorry... I will take it off list. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird -

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have a 1650 running successfully in production mode with 2-4 PRI's? I want to make sure I don't have a motherboard compatibility problem before I buy one of these. We are going to be using a Digium TE210P to start off with and probably moving to the TE411P down the road aways? thanks, Sean -----BEGIN PGP SIGNATURE----- Version:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ok... I am having a serious brain fart this evening. IIRC, the next sip draft addresses shared lines and I thought I remembered something on the list about support for it in the near future. I also thought that chan_sccp supported it with in asterisk. Am I loosing it here? Is there an implementation for shared line support in asterisk? I know that

Hi. My Situation: - One Machine with samba 3.0.14 acting as DC, DHCP3SERVER, BIND9 and dynamic DNS - One Machine with w2k3 server standard edition acting as DOMAIN MEMBER, TS and Citrix Access Essentials. - Domain Logons are working perfect. - Name resolving works fine. Reverse, Forward, NB, FQDN, IP ... - RDP Connection to the TS with local useraccount on the TS works fine - ICA Connection to

Hello, I've got a problem using the plugin managsieve in dovecot (1.0.10 on gentoo linux). I use local unix accounts on a linux machine, one for each user with a .maildir in there home directory. Here is a snippet of the logging. Feb 14 20:17:48 obelix dovecot: Dovecot v1.0.10 starting up Feb 14 20:18:00 obelix dovecot: imap-login: Login: user=<michael>, method=PLAIN,

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

I run wine on tow almost identical computers. One is a Dell xps 300 and the other on a new one, with amd 900 MHZ processor, 512 MB internal memory. nVidia Geforce grafical interface. Both run Suse 7.1, kernel 2.2.18, wine 200103026. If I run a win 311 programm on the Dell computer it runs fine. If I run the same program on the new (Obelix) computer it runs out of memory. The Del has 96 MB

OK... maybe I got a little anxious and ran out and bought a Tyan GX28 with dual Opteron (dual core) processors. (It is a nice server ;) ) I did neglect to find out that you can not manually set the IRQ's on this motherboard. I am now stuck sharing an IRQ with the ethernet controller and no foreseeable end to my dilemma. I have a Digium TE210P and zttest consistently runs at 99.97% which

Hi, I have install asterisk and it works fine. But when I use Sjphone and I use Ethereal a Client send "Request:OPTIONS sip:obelix.foo" and Server answer "Status: 404 Not found". But i can talk with two client and asterisk. When I use Xlite i don't have this request it's clean. I don't understand??????????????

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.

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system