similar to: Unable to build DAHDI-Linux in mock chroot

i currently have a samba pdc, samba bdc and samba member server all running samba-3.0.23c-1.fc5. up until the 3.0.22 releases, i never had any problems with users authenticating to member servers. problem now is, a user from windows xp professional (which is part of the domain) can auth to the pdc and bdc, but not to the domain member server. the same thing happens from windows xp home (even

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: > I want to change call files, which has caller id in them, to call > originate from dial plan. > But I don't see such parameter here > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate > > How can I pass callerid to following: > > exten => 6003,n,Originate(SIP/6003 at

On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote: > 25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: > I want to change call files, which has caller id in them, to call > originate from dial plan. > But I don't see such parameter here >

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1-rc1 DAHDI-Tools-v2.9.1-rc1 dahdi-linux-complete-2.9.1-rc1+2.9.1-rc1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1-rc1 DAHDI-Tools-v2.9.1-rc1 dahdi-linux-complete-2.9.1-rc1+2.9.1-rc1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release

Hi, I got some errors about not being able to create subdir [already existing] on a 'make update' in my zaptel 1.4. I removed the directory and did a new svn co of zaptel 1.4 [ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ] now I get: .... /usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h make -C firmware hotplug-install DESTDIR=

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1 DAHDI-Tools-v2.9.1 dahdi-linux-complete-2.9.1+2.9.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release includes a firmware

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1 DAHDI-Tools-v2.9.1 dahdi-linux-complete-2.9.1+2.9.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release includes a firmware

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1-rc2 DAHDI-Tools-v2.9.1-rc2 dahdi-linux-complete-2.9.1-rc2+2.9.1-rc2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.1-rc2 DAHDI-Tools-v2.9.1-rc2 dahdi-linux-complete-2.9.1-rc2+2.9.1-rc2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This release

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal "*" followed by

I am a fairly novice Asterisk 1.4 user who used to use CallWeaver, based on asterisk 1.2. I used Realtime MySQL with CallWeaver and am currently using the very same MYSQL tables (and columns) with Asterisk 1.4.11 and things are working well. The questions I have are, since new configuration variables have been added into Asterisk 1.4, can I simply add columns in my MySQL sippeers table for

Working with asterisk 1.4; using the AMI Originate command, it is possible to do something like: Variable: CDR(accountcode)123456 Or must the variable names be "var[n]" where n is a number? I'd like to set the accountcode for a Local channel that originates a call. Thanks. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE

I will soon be implementing a Fedora Core 4 (or 5), 2.8MHz Xeon, 1GB RAM, 2.5TB storage, gigabit ethernet server which will hold backup copies of about 200 DVDs. These DVDs will be played through a gigabit LAN to another Fedora Core 4 (or 5) workstation using xine which will then output video to the projector in my living room. I am not asking to start a competition or war on the list. My

with newest Asterisk version.=20 When holidays will end more and more people will start to complain about = this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -----Original Message----- From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony = Messina Sent: Sunday, December 30, 2007

After updating to 1.4.15, I have the following issue: When I try to use the "M" macro option in the Dial() option, I get the following in the console: -- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT") -- Called g2/w5051234 -- Zap/3-1 answered Zap/1-1 [Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to start

After upgrading to 1.8.3.2 today, I notice that my Aastra 480i SIP phones no longer initiate hold music when a call is placed on hold. I seem to be having the same issue as the person here: http://forums.digium.com/viewtopic.php?f=1&t=77553 Has anyone else run into this issue? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC