similar to: Writing Hangup causes to CDR record

> Message: 19 > Date: Tue, 19 May 2009 22:20:59 +0300 > From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> > Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help > To: asterisk-users at lists.digium.com > Message-ID: <20090519192059.GB3227 at xorcom.com> > Content-Type: text/plain; charset=us-ascii > > On Tue, May 19, 2009 at 11:11:40AM -0700,

"What is CentOS? CentOS is an Enterprise Linux distribution based on the freely available <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red Hat Enterprise Linux. Each CentOS version is supported for 7 years (by means of security updates). A new CentOS version is released every 2 years and each CentOS version is regularly updated (every 6 months) to support newer

"*What is CentOS?* CentOS is an Enterprise Linux distribution based on the freely available sources from Red Hat Enterprise Linux<ftp://ftp.redhat.com/pub/redhat/linux/enterprise/>. Each CentOS version is supported for 7 years (by means of security updates). A new CentOS version is released every 2 years and each CentOS version is regularly updated (every 6 months) to support newer

"What is CentOS? CentOS is an Enterprise Linux distribution based on the freely available <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red Hat Enterprise Linux. Each CentOS version is supported for 7 years (by means of security updates). A new CentOS version is released every 2 years and each CentOS version is regularly updated (every 6 months) to support newer

I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file =================

Hi there, we have a problem with dahdi and overlapdial. We are running an E1 in Germany and are in need of overlapdial. The E1 is connected to a Sangoma A101. As soon as overlapdial is set to "yes" we have problems with incoming audio on the dahdi channels. When set to "no" all audio is fine. Basically we can choose between being able to receive calls or to place calls

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Days ago I bought a OpenVox A400P card with a port FXS and another FXO that I am testing with my Asterisk installation in my house. I'm using Asterisk 1.4.24.1 with DAHDI Linux 2.1.0.4 and DAHDI Tools 2.1.0.2 on Debian GNU/Linux Lenny. I was reading "The future of telephony" and this [1] document looking for information about

Hi, I'm using a revision 6822-enabled Dahdi-Tools (see https://issues.asterisk.org/view.php?id=13897) with a Junghanns QuadBRI. 1. Do I still need qozap driver ? If positive, how is it recommended to get it ? 2. Which line should be included in /etc/dahdi/modules to have the appropriate driver loaded ? 3. The process I'm planning to use is : A- Hand edit /etc/dadhi/modules,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Making some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I am trying to redirect to a local extension the incoming calls that I receive to an account which I have in iptel.org, but when receiving I'm obtaining this error: alderamin*CLI> -- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820", "SIP/300|30|tTrm") in new stack [Feb 19 19:22:50] WARNING[19254]:

Hello everyone, I have a Atcom AX-1600P(1) card with a FXO module and I can't configure it. I have four extension with this PJSIP settings: --- /etc/asterisk/pjsip.conf --- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=6001 aors=6001 direct_media=no rtp_symmetric=yes force_rport=yes

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

Hi All This may be a silly question but... I have this dahdi_genconf generated file: ; Autogenerated by /usr/sbin/dahdi_genconf on Fri Jul 4 22:05:29 2014 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

Was'nt sure if this mail got through earlier: I have been having a weird issue with my telco's ISDN PRI occasionally resetting on a incoming call, i suspect it to possibly be a timing issue since some of the incoming call work. This problem happens very frequently. I am using asterisk-1.6.0.1 with libpri-1.4.9, the asterisk server is connected viw TDMoE to a Redfone Fonebridge into

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

su testuser11 cd /storage/CME/test No problem. But when I try to access the same directory in windows I get these entries in my logs.... /var/log/samba/log.smbd ------------------ [2010/01/04 16:08:25, 1] smbd/sesssetup.c:reply_spnego_kerberos(350) Failed to verify incoming ticket with error NT_STATUS_LOGON_FAILURE! log.winbindd reports no errors so it seems that the SIU/UID mapping

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

Same question for asterisk-users as well: ----- Forwarded message from Tzafrir Cohen <tzafrir.cohen at xorcom.com> ----- Date: Wed, 7 Mar 2012 21:14:04 +0200 From: Tzafrir Cohen <tzafrir.cohen at xorcom.com> To: asterisk-dev at lists.digium.com Short version: it's now time to remove. Anybody actually uses latest DAHDI with RHEL4? See inline, On Thu, Dec 29, 2011 at 07:42:39PM

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in