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Old 01-01-2013, 07:20 PM   #1
baronobeefdip
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Registered: Jul 2009
Distribution: Debian Squeeze
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Asterisk Trouble


I have setup an asterisk server on a Debian Squeeze machine and setup some softphones on the clients as tests (ekiga), When I dial the users the softphone keeps telling me that the "user wasn't found" and I don't know why and it's confusing since I can't figure out what I did wrong.

Here is how I configured the users.
in the /etc/asterisk/sip.conf file i added these two entries at the bottom.
Code:
[josh]
type=friend
host=dynamic
secret=password
context=users
callerid="Josh"<100>
mailbox=100@default

[tux]
type=friend
host=dynamic
secret=password
context=users
callerid="Tux"<101>
mailbox=101@default
Then I added the following lines in the /etc/asterisk/extensions.conf file
Code:
exten => 100,1,Dial(SIP/josh,20)
exten => 101,1,Dial(SIP/tux,20)
On all of the ekiga softphones I added the accounts as sip accounts like this
Code:
name=(a name)
registrar=(the ip address of the pbx server)
user=(the name given in sip.conf)
authenticated User=(left blank)
password=(the password given which was password)
timeout=(left as is)
Did the same for the other user on the other softphone and attempted to call them. When I was on the tux phone I attempted to call the other user by typing in "sip:josh" in the text box at the top of the client and at the bottom of the window it tells me that "user wasn't found". Next I tried using josh's number that was given in the extensions.conf file like this in the text box at the top "sip:100". I still got the same outcome which was user not found. What did i do wrong or forget to do here, I am kinda new to PBX but have been trying for a while and every book,internet guide give very few troubleshooting tips for solving this kind of problem
 
Old 01-02-2013, 02:29 AM   #2
hamzar.pm
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Registered: Aug 2009
Location: India,Kerala,Cochin
Distribution: Fedora 11 ,Centos, ubuntu, Redhat
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Hi , instead off digging in asterisk CLI go with gui
here you can download it
http://downloads.asterisk.org/pub/te...-gui/releases/
its a web based one so that you can configure your extensions and every features very simply...
i had so many issues with my asterisk installation .. this solved all my issues...
 
Old 01-02-2013, 02:37 PM   #3
baronobeefdip
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I don't like the GUI, It's too difficult to setup plus the lab environment I am working in prefers text based and they are too stubborn to try cli or text based. no matter what you tell them they will not budge. I don't want any reccomendations for GUI interfaces I want to know what I am doing wrong in this configuration. Up to now I have been configuring it through a network and not locally, I would like to maintain that method of configuration and I don't want the help of any GUI or web-based environment to get me a working asterisk installation. I have been interfacing with the server via ssh this whole time and I am confused as to what I am doing wrong to make this thing not work.

Last edited by baronobeefdip; 01-02-2013 at 02:39 PM.
 
Old 01-02-2013, 03:29 PM   #4
amlife
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Location: Canada
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what do you see in the logs? try and tailf the logs when you try and dial or connect one of the users.

to me it sounds like a misconfiguration issue!

You don't like the gui? or hard to setup? don't get me wrong! but you are heading the wrong direction! to setup a stable environment try and find the easiest way to keep your services run smoothly! and try to reduce chances of making mistakes and reduce the possibility of things that could go wrong! always think of Murphy's law!

use easy asktrisk or tripbox or Elastix they all come with gui right of the bat!

good luck
 
Old 01-03-2013, 12:24 AM   #5
hamzar.pm
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Okey if you are not prefering the gui.. go with asterisk cli and start the asterisk cli in verbose (asterisk -vvvr)
and try calling extensions... you will be able to c the mistakes or error msgs and all ...
 
Old 01-03-2013, 12:05 PM   #6
baronobeefdip
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Quote:
Originally Posted by amlife View Post
what do you see in the logs? try and tailf the logs when you try and dial or connect one of the users.

to me it sounds like a misconfiguration issue!

You don't like the gui? or hard to setup? don't get me wrong! but you are heading the wrong direction! to setup a stable environment try and find the easiest way to keep your services run smoothly! and try to reduce chances of making mistakes and reduce the possibility of things that could go wrong! always think of Murphy's law!

use easy asktrisk or tripbox or Elastix they all come with gui right of the bat!

good luck
I have a Debian box that has a web server, DNS and SSH service installed and I don't want to blow everything away because I need to install a dedicated asterisk system because I won't always have physical access to the machine and I would like to have the asterisk service on the Debian server I already have and work with it. It's the only machine I have at the moment that I am allowed to keep running for days on end but I switch it off for the cost of electricity.
 
Old 01-03-2013, 01:27 PM   #7
baronobeefdip
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Your reccomendation for using trixbox inspired me to use it and then take a peek at the configuration files as it adds extensions and see how they do it. I have a great example to compare extension to with the sip.conf file but I need some help finding the dial plans. Where are the dial plans added and how can I find the ones I added for example here is one of my sip.conf entries
Code:
[100]
deny=0.0.0.0/0.0.0.0
type=friend
secret=Defense08
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=100@device
host=dynamic
dtmfmode=fgc2833
dial=SIP/100
context=from-internal
careinvite=no
callgroup=
callerid=Device<100>
accountcode=
call-limit=50
Where is the dialplan for this sip entry going to be, and what will it look like (make it easy for me to find it)
 
Old 01-03-2013, 03:11 PM   #8
amlife
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Quote:
Originally Posted by baronobeefdip View Post
I have a Debian box that has a web server, DNS and SSH service installed and I don't want to blow everything away because I need to install a dedicated asterisk system because I won't always have physical access to the machine and I would like to have the asterisk service on the Debian server I already have and work with it. It's the only machine I have at the moment that I am allowed to keep running for days on end but I switch it off for the cost of electricity.
You can run the GUI on different port say 8080 or something, and protect access using ssl certificate and password.
 
Old 01-03-2013, 03:45 PM   #9
baronobeefdip
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I wanted to know where the system stores the dial plans that have been added through the gui
 
Old 01-03-2013, 07:00 PM   #10
baronobeefdip
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I would at least like to know what the dial plan for the generic SIP device looks like in the extensions.conf file when you set it in trixbox. I am having to keep following include statements and whatever else and I am getting nothing (unless this next line of code is the diaplan for the 100 extension, first here is the SIP.conf file entry
Code:
[100]
deny=0.0.0.0/0.0.0.0
type=friend
secret=herbie
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=100@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/100
context=from-internal
canreinvite=no
callgroup=
callerid=device <100>
accountcode=
call-limit=50
And the from the best that I can find here is a dialplan that I need confirmation on if it is the right kind of statement for the SIP device specified above.
Code:
exten => 100,1,Macro(exten-vm,novm,100)
exten => 100,n,Goto(${IVR_CONTEXT},return,1)
exten => 100,hint,SIP/100&Custom:DND100
Is this the dialplan for the above SIP device or is it going to be somewhere else?
 
Old 01-04-2013, 12:14 AM   #11
baronobeefdip
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Registered: Jul 2009
Distribution: Debian Squeeze
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Quote:
Originally Posted by amlife View Post
You can run the GUI on different port say 8080 or something, and protect access using ssl certificate and password.
I still prefer the text based configurations, SSL certificates and alternate ports seem to be more work than simply figuring out why the dial plans I am putting into the system aren't working. I have changed the permissions of the files before anyone asks and they still aren't working. They look at simple as this
Code:
exten => 100,1,Dial(SIP/josh,20)
exten => 101,1,Dial(SIP/tux,20)
This code should direct the call to Josh if Tux punches in 100 in the dial pad but the phone isn't ringing and a call isn't being initiated. I did check the permissions on the extensions.conf file and changed them to 777 (temporarily, I am aware of the implications of using a chmod setting like that). And I still get the 404 error I have been getting all along. Up to now nobody has been giving me a strait simple answer as to what I should put into the extensions.conf file with the settings I have setup in the sip.conf file because the settings that I have in the extensions.conf file right now are obviously not working no matter what permissions I set them to.
 
  


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