similar to: Using Dovecot as Asterisk PBX voicemail server

Hi, i upgraded from Debian sarge to etch, so from samba 3.0.14a-3 to 3.0.24 When someone creates/modify a file the user attribut is set to r (instead of rwx) so the next time a user opens the file is's on read only. What can i do so that the file keeps the attribut rwx for the owner ? Another question : the file has the rigthts : -r--rwx--- but although the user is member of the group

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢

I've got a fresh FC2 install and I'm trying to get the symlinks right according to the /usr/src/zaptel/README.Linux26 instructions. I've created two symlinks: /usr/src/linux-2.6 -> /usr/src/linux-2.6.5-1.358 /lib/modules/linux-2.6 -> /lib/modules/2.6.7-1.494.2.2 When I do a "make linux26", I get a million warnings and errors with the result being: make[2]: ***

Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks...

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Folks, My question concerns the SIP Notify that is being sent to ... device. You can see it in the following line: Voicemail: 0/0 Shows no Voice mail but I did leave a voice mail at the extension. Any suggestion on what I should look for in my * setup. I am not worried about the 481 coming back for the other side yet. Once I get a handle on the Notify, I'll work on the 481.

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

First off, I have to admit that I'm uncertain if this is the appropriate forum; I'd be happy for suggestions about where else to look. I'm doing this work on a stock install of CentOS-7-x86_64-Minimal-1810.iso, with no updates. I'm trying to create an RPM database from a custom set of RPMs. One RPM ('openldap-ltb' from the LDAP Tool Box project (ltb-project.org) has a

I get this error no matter what I do. I've tried switching to different locations on the TDM400p to no avail. I've got the card setup with the green module in slot1 and red in slot2. My config files are exactly like the "Configure Dev-Kit PCI (TDM11B)" example. http://www.digium.com/index.php?menu=configuration#TDM11B When I run ztcfg -v, I get: 2 channels configured.

Hi all, I have been worriyng and googling a lot but I can't find my mistake. I am trying to regiter an X-Lite Softphone to Asterisk, but I am getting a SIP/2.0 403 Forbidden response: SEND TIME: 10157385 SEND >> 10.100.249.12:5060 REGISTER sip:10.100.249.12 SIP/2.0 Via: SIP/2.0/UDP 10.100.249.86:5060;rport;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22 From: Tester

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

I need to get your thoughts on something.. :) I am trying to create a system to process the CDR call logs for department accounting.. I think there are two ways of doing it.. Either I can create an AGI that will run on the "h" extension and will lookup the last entry that matches the account code of the call that just ended in the MySQL CDR and calculate the call cost immediately..

Patrick hi. Asterisk can do that, and you don't need VOIP lines. If you connect Asterisk to the net, and all employees have a VOIP phone (either hardware or software) then you're good to go. What do you need? To begin with, install linux on an old pc (well, not too old). Then go to voip-info.org and take a look at the Asterisk wiki. Everything you need is there. And of course, we're

I'm a newbie with a TDM11B. I've read the FAQs about linking /usr/src/linux-2.6 to /usr/src/linux-2.6.8-1.521 and /lib/linux-2.6 to /lib/linux-2.6.8-1.521 but still get a million errors and eventual abort during compile. Could someone point me in the right direction? I do a yum update every day and I'm using the CVS from 9/14/04. I'm also using an ASUS CUSL2 system board.

Another point of disparity between how I see others write Puppet manifests and the Best Practice that I''ve adopted at my institution is the use of quoting. In Puppet, you can get away with not quoting values or references if there isn''t a special character or a keyword being used (e.g. package { openssh: ...} or User[agirl]). However, even though that is possible, to make

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Hi, I have asterisk behind my primary PBX connected via ISDN (chan_capi). Calling out and calling in works just fine, however I can't connect to my primary pbxs' extensions. If anyone has an example extensions.conf, I'd be grateful for a copy. I tried (the MSN of the ISDN card is set to 30) exten => 22,1,Dial,CAPI/30:22 but this does not work. Changing it to exten =>

I think you are mixing up answers and general advice. FreePBX was intended to get you over the dialplan creation hurdle (the biggest challenge for people new to Asterisk). In regards to the LinkSys they are compatible and you do find them in enterprises, but admins are trying to get rid of adapters/converters so if possible you may wish to invest in SIP devices directly instead of an adapter.