similar to: h323 configuration

Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c)

I extracted the following weird error from the sys-log regarding pdu and schannel (have the same PID): Feb 24 10:40:37 s-svr01 smbd[31833]: [2004/02/24 10:40:37, 0, pid=31833, effective(65534, 65533), real(65534, 0)] rpc_server/srv_pipe.c:api_pipe_netsec_process(1371) Feb 24 10:40:37 s-svr01 smbd[31833]: failed to decode PDU Feb 24 10:40:37 s-svr01 smbd[31833]: [2004/02/24 10:40:37, 0,

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

Hi asterisk users! Here?s my issue, I?ve deleted the ?s? extension cause I don?t want any action to be taken on incoming calls as my pbx is for home use, but I would like to ring all my VoIP extensions at the same time the PSTN line rings and to be able to pick up the call in any extension, honestly I don?t know if this is possible, some ideas ??? Thanks in advance! -------------- next

Hi all, Running CentOS 4.2 with the following conditions: [root at gatekeeper bin]# uname -srvmpi Linux 2.6.9-22.0.1.EL #1 Thu Oct 27 12:26:11 CDT 2005 i686 i686 i386 [root at gatekeeper bin]# uptime 07:54:41 up 31 days, 13:45, 6 users, load average: 0.00, 0.02, 0.00 [root at gatekeeper bin]# rpm -q ntop ntop-3.1-1.2.el4.rf [root at gatekeeper bin]# df -h S.ficheros Tama?o Usado

I've found this in my dovecot.log: ,--[ /path/to/dovecot.log ]-- | Jul 11 01:44:29 dovecot: Info: Dovecot v1.2.1 starting up | ... | Jul 11 01:50:21 auth(default): Info: client in: AUTH 1 PLAIN service=sieve lip=192.168.111.222 rip=192.168.111.122 lport=12000 rport=35084 resp=<hidden> | Jul 11 01:50:21 auth(default): Info: sql(j.doe at

Please read the following notes regarding your project at freshmeat.net: - olly at survex.com has requested ownership of the freshmeat record for your project. Do you approve this change? If you are not sure what this means, please see http://freshmeat.net/faq/view/46/. Sincerely, Ray Shaw <news-admins at freshmeat.net> --- Headers Follow --- >From noreply at freshmeat.net Mon Sep

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

The Wine development release 1.5.13 is now available. What's new in this release (see below for details): - Client-side window rendering using the DIB engine. - Raw input support for keyboard and mouse. - Support for transparent window areas using color keying. - Proper C++ RTTI support on 64-bit. - Implementation of logical processor information. - Support for fonts with