similar to: Events offered by

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.

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

> 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

Hi all, we use OpenSER together with Asterisk. All SIP users registers with OpenSER and asterisk is doing the voicemail thing. We use the Asterisk RealtimeArchitecture for voicemail users and SIP peers. The database table for the sip peers is a view from the OpenSER subscriber table. The MWI for a user will only work, if the user object (sip peer) is loaded into memory and visible with the CLI

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

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

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

Hi im running several gentoo servers with Asterisk, only using IAX2 and SIP. Recently we decided to implement h323. All the necessary dependences for oh323-0.7.3 were installed by portage (package manager of Gentoo distro), including openh323, pwlib etc. The module is successfully loaded (load chan_oh323.so) but when asterisk is stopped (stop now) or the oh323 module is unloaded (unload

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,

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

I have a script that I want to leave running in the background to handle specific manager events. I'm running into a problem where it gets stuck in the wait_response function in phpagi-asmanager.php and the PHP maximum execute timeout kills the script. The script doesn't interact with the dialplan, so I cannot launch it from within Asterisk. Any pointers would be appreciated. I did

Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd

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??????????????