I just installed asteriks on my remote server ,and set up 2 account for test purpose.i added these lines to sip.conf
Code:
[benchabane]
type=friend
username=benchabane
secret=tcpip13
host=dynamic
context=mygroup
[Code]...
I compiled and installed the Asterisk ztdummy package because there is no rpm for it, unfortunately, and i even reinstalled asterisk, but i still get the "No application 'Meetme' for extension..." error when trying to conference. I do a "module show", and it lists other modules that were compiled with the zt source, but not ztdummy.Does anyone know how to fix this? This is more than a passing interest or hobby, because i need to conference about 3 to 5 people to help me test a new Website Content Management System and User Forums Management System i am about to launch as a service.
View 3 Replies View RelatedI just installed ubuntu and cannot log in. authentification failure.
View 1 Replies View RelatedI have a debian lenny server installed with postfix, pop, imap, authdeamon. Pop work great, I can authenticate in the server , read mails ..Currently I am looking to install roundcube which runs under IMAP .with imap i still have some trooubles when i try to be authetified ..I enabled debugging mode of authdeamon
Logs results:
Jul 21 12:03:49 ks308711 imapd: Connection, ip=[::ffff:93.23.239.100]
Jul 21 12:03:49 ks308711 authdaemond: received auth request, service=imap, authtype=login
Jul 21 12:03:49 ks308711 authdaemond: authpam: trying this module
Jul 21 12:03:49 ks308711 authdaemond: authpam: sysusername=informaticien, sysuserid=<null>, sysgroupid=100, homedir=/home/informaticien, address=informaticien, fullname=, maildir=<null>, quota=<null>, options=<null>
Jul 21 12:03:49 ks308711 authdaemond: authpam: clearpasswd=<null>, passwd=x
Jul 21 12:03:49 ks308711 authdaemond: pam_service=imap, pam_username=informaticien
Jul 21 12:03:49 ks308711 authdaemond: dopam successful
[Code]...
i installed asterisk with yum, the command i used is that yum install asterisk*
Now i want ro remove only asterisk, not its depencies. how i do this?
I have a USB 3g modem, which also supports voice calling, I want to implement IVR ( Interactive Voice response) with it on linux, Can I do it with Asterisk? If yes, then please help me on how to do this ( references) ?
Is there example or howtos available to do this?
Can I develop IVR system in linux without Asterisk?
Note: I am very new to Asterisk and IVR.
What are the standard procedures to monitor asterisk with nagios? I downloaded check_asterisk from URL...
View 1 Replies View RelatedI want to authenticate and authorize for SiP user of my * box, it is my project, but i did not finish it, hic hic! My asterisk box is version 1.6.0, fedora 10.the first, i downed and installed freeradius to my computer, and run "radiusd -X", it's run normally. Then, I installed radiusclient-ng0.5.6, it's rpm file. All is perfect. But the problem is i can't run asterisk box as radiusclient.
View 2 Replies View RelatedI have been reading about asterisk, I did some basic configuration, a small ivr, record messages.. but I was wondering how and what/where should I modify to use a database to save sip users and voicemail user..then we can add user/voicemail with php-myadmin I always search but I cant find about this configuration..
View 1 Replies View RelatedMy asterisk version is 1.6 and Web-MeetMe is 4.0.2 I get the following upon issuing "make" in web-meetme/cbmysql/ :
Quote:
cc -pipe -I/usr/include/mysql -L/usr/lib/mysql -fPIC -I/usr/src/asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_cbmysql.o app_cbmysql.c
app_cbmysql.c:14:22: error: asterisk.h: No such file or directory
app_cbmysql.c:25:33: error: asterisk/autoconfig.h: No such file or directory
app_cbmysql.c:26:27: error: asterisk/lock.h: No such file or directory
[Code]....
Has anyone ever had any luck with htpasswd and htaccess on asterisk, I set it up on a test apache server in VMWare just to make sure I knew what I was doing, so It was a very basic html page that I used, however, when I go to implement it on one of my Asterisk Servers, It comes up with the following page after I type a user name and password credentials in:
***************************************************************************
Internal Server Error
The server encountered an internal error or misconfiguration and was unable to complete your request.
Please contact the server administrator, webmaster@localhost and inform them of the time the error occurred, and anything you might have done that may have caused the error.
More information about this error may be available in the server error log.
Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny4 with Suhosin-Patch Server at fred Port 80
***************************************************************************
Anyone know if its compatible? and if so, is there any tricks around this?
I want to install Asterisk gui using svn command, but I have the following error:
# svn co http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui
svn: PROPFIND request failed on '/svn/asterisk-gui/trunk'
svn: PROPFIND of '/svn/asterisk-gui/trunk': Could not resolve hostname `svn.digium.com': Temporary failure in name resolution (http://svn.digium.com)
We would like to install Asterisk with a minimal centos build. We are writing kick-starts to install packages, does anyone know or have to hand a packages list which installs minimum system requirements for asterisk? Preferably this will not exceed the contents of disk 1
View 4 Replies View RelatedI have fedora 13, and installed asterisk.. Before I had centos and have my asterisk running to test and learn.. but in fedora I see there is a http miniserver for admin asterisk..I edited enable, port and ip in the file http.con but when I try URL...I got 404 page no found Asterisk server.
View 1 Replies View RelatedI first installed asterisk using sudo apt-get install asterisk on ubuntu 10.04. After doing this the SIP packages did not install for some odd reason.I uninstalled asterisk using sudo apt-get remove asterisk then reinstalled and now none of the asterisk config files are gone. deleted all files and directories I could find relating to asterisk, rebooted the machine, did a kernal upgrade, and a package update and reinstalled asterisk using sudo apt-get install asterisk, and all the config files are still missing. What am I missing here!
View 6 Replies View RelatedIs it possible to make root password visible as asterisk(or any other character) on using sudo command??urrently,it does not display anything on monitor,thus,making it difficult to count the number of keystrokes pressed...
View 5 Replies View RelatedI have two asterisk servers each one behind a linux firewall/gw. Linux is Centos 5.4, kernel 2.6.18-164.el5, iptables v1.3.5. Routes on the fws are ok and when iptables is stoped the servers are see each other, all good. But when I run iptables script in any fw, one server (not always the same) goes unreachable. I verify this with asterisk -r, then show sip trunk, and status becomes UNREACHABLE.
Iptables scripts is generated by fwbuilder. The weird part is I put only one rule to de script and it looks like Source=any, Destination=any, Service=any, Interface=any, Direction (Inbound,Outbound)=any, Time=Any, Action=ACCEPT. So as you can see I tried something like "Do not do anything at all". But anyway I run the script in any fw and one server becomes UNREACHABLE. I think the script does something wrong after all or maybe I have some missconfiguration in my asterisk conf files. The point is I am not so expert in iptables or shell scripting so I can't see anything in the iptables script. I have look for some issues like iptables blocking because of ip_conntrack table full, or "dont fragment" bit set in kernel problem, but nothing seems to be the right problem at all.
I am using SAMSUN R-519 laptop with 4 GB RAM and 2.1 GHz dual core CPU and I intend to configure asterisk on this machine for voice conferencing with 4 other PCs. I am unable to configure asterisk. I don't have any hardware regarding voip such as Digium card.
View 14 Replies View RelatedI have fedora 13, and installed asterisk.. Before I had centos and have my asterisk running to test and learn.. but in fedora I see there is a http miniserver for admin asterisk..
I edited enable, port and ip in the file http.con but when I try [url] I got 404 page no found Asterisk server...
I was just wondering if you know of any site/forum that is informative about Asterisk. I have just configured an Atcom IP04 in which Asterisk is embedded. I need to know how to configure the IP-PABX to receive fax and come up with a Skype Gateway to receive/call out on Skype.
View 2 Replies View RelatedI am encountering a strange problem on my VOIP setup Basically, I have a asterisk appliance IP04. I have setup all the extensions and everything. I use a Linksys PAP2T as an ATA remotely. Now, my problem is the ATA sometimes is okay can call SIP and PSTN but sometimes I just can't hear anything. I thought it was my ISP blocking the VOIP packets but I have tried both the SIP softphone and IAX2 softphone on my PC. For IAX2, it works perfectly however in the SIP, I can hear the other end but they cannot hear me.
These are the ports I have opened on my router
1.) UDP 5060 - SIP Port
2.) UDP 10000 - 20000 - RTP Port
3.) UDP 4569 - IAX2
Do I need to open both TCP/UDP for these ports or UDP should be enough? These are the test cases:
1.) Using my WiFi Connection and a analog phone connected to ATA --> Sometimes working sometimes not and sometimes you can call SIP but the other end cannot hear you
2.) Using IAX2 in WiFi connection --> This one works perfectly
3.) Using a mobile phone connected to WiFi Network --> The same...but you can call and go out on PSTN but the other end cannot hear you
4.) Using a mobile phone connected via 3G --> Works perfectly but as expected it is quite slow and voice quality is awful
I want to use SIP rather than IAX2 because it is widely used and since my ATA doesn't support IAX2. Are there other ports I need to open or configure?
Today my boss come to me and ask me to get a cote to upgrade our old "Cisco call manager" (2004) now "Cisco Unified Communications Manager". So I was wondering, instead of doing a costly upgrade (over 35 000�), maybe it's time to change... Does anyone of you got some insight with Asterisk in an enterprise environment? Is it reliable? Following you own judgement, what are the + and - ? If Asterix worth it, what argument (apart of the price) could I use to help the management turning on my side? Will the Cisco 7921 VOIP phone will be able to connect to it? (as we do have over 35 of them)
Enterprise environment:
- 3 sites (VPN interconnected)
- ~35 VOIP phones and ~10 landlines phones
I have to clean the voicemails from asterisk, but I want to keep at least, the voicemails for five days.I started writing a script, but I'm a bit stuck, and I've trying somethings for sometime, but still I can't get itThe hierarchy of Asterisk Voicemails is like this:
default
10000
INBOX
[code]....
I need to custom my dialplan at Linksys Pap2-NA to fit my asterisk server as i have three destinations.
Localy : xxxx
International: 00.
Services: *666x
Maybe I'm not using the right search terms, but I can't find anything on this. I have a setup of Slackware with Asterisk and FreePBX. I am recording calls on demand and can get to the recordings from the Call Monitor (web interface). I would like to be able to access the recording from the phone (Aastra 57i), much like a voice mail is accessed.
View 3 Replies View RelatedI'm having a hard time getting part of Asterisk (the open source PBX) called asterisk-addons to compile with mysql CDR support which I need to enable Realtime I believe. I've spent the whole day trying to fault find this one (including thinking I had ruined my box and creating a new CentOS build!) and am pretty worn outWhen I attempt to install asterisk-addons (I've tried 1.6.0.1, 1.6.0.1-patch and 1.6.1.0-rc3), I get the following line in a ./configure:
checking for mysql_config... /usr/bin/mysql_config
checking for mysql_init in -lmysqlclient... no
Then when I do a make menuselect, MySQL is not selectable and XXX"d out:
[code]....
I am using debian squeeze server with asterisk 1.6 installed and configured.my problem is non root users need to access the server using ssh and restart asterisk server after making changes in asterisk configuration files.As of now i am giving root username/password for this process (i know it is not at all a good idea) .now how can i create a username and configure it which can only access and modify asterisk configuration files and restart asterisk server without any other privileges.
View 1 Replies View RelatedI currently have Zarafa and I am going to install MythWeb for MythTV.To solve the permission problem for installing amportal (FreePBX), I have to change Apache's user and group to Asterisk instead of this, which I've commented out:User ${APACHE_RUN_USER}Group ${APACHE_RUN_GROUP}Once FreePBX is installed completely, I checked to make sure FreePBX is working and it worked. I used the latest version of Asterisk from the SVN trunk, so I used ./install_amp --my-svn-is-correct.However, I want to use Zarafa AND FreePBX, I just don't like having to change the Apache User and Group to asterisk.Here is a couple of examples of instructions when installing FreePBX:Is there a workaround? I really want to change it back to what it was before while FreePBX can still access their own directories.
I did not know if this question has been asked, but I thought I'd ask here. I've added "zarafa" and "mythweb" to the tags list my thread.Update: I changed my apache configuration file back to www-data for User and Group and I do have /var/lib/asterisk/agi-bin and /var/www/asterisk (where FreePBX PHP pages are installed in /var/www/asterisk) owned by asterisk:asterisk. But I get this:
Code:
Retrieve conf failed to copy file(s) from a module's agi-bin dir: copy(/var/lib/asterisk/agi-bin/directory): failed to open stream: Permission denied
[co0de]...
i just had to switch from 10 to 11 until 12 comes out, and after setting up asterisk, i noticed VERY VERY BAD sound quality, mostly comprised of severe jittering. I noticed that the way Fedora lets you work with your sound devices in 11 is way different (nicer, but obviously not complete) than ever before, and i am using the same asterisk configs as before, so i'm positive that this is NOT an asterisk issue, and does NOT really belong in this forum; however, only asterisk users will be able to replicate or identify with this issue.
Have any of you 11 users had this happen to you, and if so, did you figure out how to solve it? I'm sure that i can *try* adjusting asterisks jitter buffer settings, but like i said, it was NEVER an issue before 11... Another clue would be that i cannot get crap like gtk-recordmydesktop to produce sound (but that may very well be another issue; however i doubt it)...
I want to use asterisk as a PBX and SIP server using Skype. I have everything planned out but I am having a problem loading the "Skype for Asterisk" module provided by Digium. This module costs roughly $60 and Digium provides its customers with free support. After calling Diguim, the technician was unable to find the problem and said that I "should use a different Linux." I found this advice very hard to swallow as I have grown very fond of my openSUSE.I searched the Digium forums and found that another user had the very same problem with Ubuntu. We compared notes in this forum thread:
[URL]...
We were unable to figure out the problem until another member pointed to a Kernel bug in Ubuntu as the source of the problem. The Kernel where this bug exists is 2.6.31-11. Here is a link with some information on the bug:
