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Messages - half_life

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Go to Users and groups.  Then LDAP Settings.  Third item down is the password assuming that you are using 2.0.

I can understand what you are saying.  I was just informed in another thread that (a show stopper in my opinion) problem with asterisk voicemail crashing the VOIP module would not be addressed until the 2.1 rollout.  The schedule for beta doesn't occur until June with a Sept.  time frame for gold release of 2.1.  That is a long time for a bug to exist in a major component.  I thought the whole idea of the product was to cater to the non tech business person.  I am the IT manager at my workplace and we have 3 employees.  Me, myself, and I.  If I get taken out of commission I need the interface for things to be easy and just work.  I thought that I had found an ideal solution until this.  Now I am going to split out the VOIP parts and roll my own setup.  That means that I can't just tell my boss to buy an Enterprise support contract if something should happen to me.  Other than the video issue in the VM, I really haven't had much trouble with Zentyal until this.

Installation and Upgrades / Re: Asterisk Voicemail oddity
« on: February 21, 2011, 05:53:17 pm »
Forgive me but the 2.0 series just went gold in Sept.  The first round of betas won't be until June.  Are you really saying that you are going to let a show stopper sit there for that long.  For me the answer is straight forward,  rip out the ebox asterisk component and use plain vanilla.  Where does that leave your target customer though? 

Just my two cents


Installation and Upgrades / Asterisk Voicemail oddity
« on: February 21, 2011, 07:16:53 am »
Here is the details,  when I call an extension and leave voicemail everything works as expected.  When I dial the main voicemail extension to retrieve voicemail the asterisk instance crashes on hangup.  It doesn't matter if I actually punch in credentials or just hang up.  Also, waiting for a minute before hanging up makes no difference. 

System info

Core version 2.0.15
all modules and options loaded

What might make it unique :
VM under Xen 4.0.1
2 g Ram
4 pinned CPU's

The system has 24g ram as a whole and 16 cores in two dies

Everything is 64bit AMD

I should mention also that there is a warning on the asterisk cli

53:09] ERROR[3970]: res_config_ldap.c:1317 update_ldap: Couldn't modify dn:uid=dhoff,ou=Users,dc=Zentyal,dc=gltconline,dc=com because Naming violation

This is not my first rodeo with Asterisk and still this one is confusing to me .  When using sip debug mode it really doesn't throw meaningful errors just crashes.

Installation and Upgrades / Re: Hardware VOIP phones
« on: February 21, 2011, 02:43:28 am »
I have tested the ldap changes with a soft-phone.  I changed:

md5secret = eboxRealmPassword


#md5secret = eboxRealmPassword
secret = AstAccountSecret

note that I changed from md5 hash to plain text (not reccomended on public networks)

I used Luma (ldap browser/editor ) to add AstAccountSecret to an account with a plaintext password.

To finish the job with a hardware phone I would add my user account name and plaintext password to the phone (in my case via a few config files served up by an tftp server since they are Cisco).  I leave it as an exercise to the reader how to use md5 passwords and configure other phones.  I will re-post a complete explanation with Cisco config file examples after I have done more testing.

Since my needs involve more than basic asterisk functions (as provided in Zentyal), I will be overriding parts of the default config (multiple DID's and several queues plus an operator panel) I can post a more detailed explanation of how I did it if anyone is interested.   Because I am mindful of the fact that others in my company aren't linux savvy, I am using GUI's wherever possible such as a visual dialplan tool etc. 

You are very welcome.

Installation and Upgrades / Re: Hardware VOIP phones
« on: February 21, 2011, 01:08:09 am »
Can you see any problem with my solution in the mean time? Might it break an upgrade for instance?  I have toyed around with the idea of adding my phone xml/flat config files back into the ebox .mas file paradigm.

Installation and Upgrades / Hardware VOIP phones
« on: February 19, 2011, 06:44:32 am »
Folks, I have searched the archives and can't find anything that comes close to answering this so here I go.   I have about 20 Cisco VOIP phones that I will be setting up in Asterisk in the next few weeks.  IN plain vanilla Asterisk this is a breeze.  Now enter the twist that has me scratching my head.  Each SIP client should be authenticated via password.  Zentyal has chosen to use end-user passwords.  Phones like Cisco 79XX use XML configuration files to set themselves up.  The password for authentication therefore gets put in an XML file for the phone as a standard practice.  When the user changes their domain password,  how do I get that info back around to the phone?

I am the IT manager for my company and have chosen Zentyal as my server/gateway of choice because I need a straight forward setup if I ever happen to have an accident.  It would be nice to keep everything together under one GUI if I can.  Can anyone give me a hint?


Update: Feb20 18:42 EST

On further digging around it is looking like the solution is to modify the res_ldap.conf
   md5secret = eboxRealmPassword
  md5secret = SomeOtherPassword

and append SomeOtherPassword =  SomeFixedPassword to each users LDAP record.

More on this as I experiment with it.  Anyone with thoughts on the matter feel free to jump in.


Installation and Upgrades / Re: basic settings, GATEWAY and ROUTE
« on: February 16, 2011, 03:56:36 am »

On your internet side (eth0) if you are being fed via dhcp you don't need any gateway information.  This is provided by your ISP.  On your internal side (eth1) You would turn on dhcp service for your network
so all you need here is your fixed ip address for your server

DHCP (in the infrastructure group) 

Gateway Zentyal
 Probably every setting on this page that has a Zentyal choice ..... that is the one you want.  This is enough to get the system up and running serving IP's to machines..... Have fun


Installation and Upgrades / Re: Asterisk several SIP Provider?
« on: February 13, 2011, 12:12:40 am »
Probably the easiest way of doing this would be to add an #include statement to the /etc/asterisk/sip.conf pointing to another file say external-sip.conf.  Put the register lines in this file.  

For instance :

#include /etc/asterisk/external-sip.conf
;here is an example setup for telasip
;callerid = "Joe Smith"<4345099999>
disallow = all
allow = ulaw
allow = gsm
canreinvite = no
qualify = yes
context = telasip-in
dtmfmode = rfc2833
fromuser = my-username
host =
insecure = very
secret = password
sendrpid = yes
type = peer
username = my-username


register => username:password@sip-provider-gateway/(incoming extension to go to optionally)
if you don't add the extension it will look at the sip-provider-gateway entry in sip.conf (or external-sip.conf)  to figure out which context to send the incoming call to where you can sort based on what number is being called.

Example incoming from telasip:


exten => _4345099999,1,Goto(privacy_gard,s,1)


The reason for using an #include statement is to separate your custom changes from what Zentyal manages.  It reduces the changes that you make via sip.conf.mas to just one line.  This also works for extensions.conf and most of the other files that asterisk uses.  You can optionally use a visual dialplan tool like Apstel's Visual Dialplan product to manage your dialplan leaving Zentyal to manage users and extensions plus Queues.  This gives a totally GUI oriented approach.

I hope this is helpful


The same thing happens when using Xen.  I had assumed that it was a problem with vnc.  I do one of two things:

1 :  login via https://my-zentyal-ip/ from another machine
2:  ssh -X into my zentyal ip  and start firefox from the commandline

I am using Zentyal in a Xen production environment with most modules activated.

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