[asterisk-users] SIP REGISTRATION TIME OUT

[dave cantera]

hello,
I use both * 1.4 and *NOW...   because the *gui is incomplete in *NOW, I 
loaded 1.4 over *NOW because the gui regenerates files that, well, don't 
seem to work very well.  it seems to me the gui creates the users.conf 
file, and then a script creates or uses the users.conf to create the 
dialplan...  here is the users.conf file from *NOW...

as you can see, this file does not conform to either sip.conf or 
extensions.conf, so that is my reasoning that it is source for some 
other generator...
daveC

;!
;! Automatically generated configuration file
;! Filename: users.conf (/etc/asterisk/users.conf)
;! Generator: Manager
;! Creation Date: Sun Jan 21 15:41:42 2007
;!
[general]
;
; Full name of a user
;
fullname = New User
;
; Starting point of allocation of extensions
;
userbase = 6000
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Create SIP Peer
;
hassip = yes
;
; Create IAX friend
;
hasiax = yes
;
; Create H.323 friend
;
;hash323 = yes
;
; Create manager entry
;
hasmanager = no
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
host = dynamic
localextenlength = 4
;[6000]
;fullname = Joe User
;email = joe at foo.bar
;secret = 1234
;zapchan = 1
;hasvoicemail = yes
;hassip = yes
;hasiax = no
;hash323 = no
;hasmanager = no
;callwaiting = no
;context = international





Nicholas Campion wrote:
> The quick way to check if a user is defined is to go to the asterisk 
> console and type "sip show users" which will list all the defined 
> users and passwords.
>
> You say that it isn't a networking issue, but the fact that you are 
> behind a NAT (your local ip is 192.168.0.100 <http://192.168.0.100>) 
> is causing the problem (i think).  All of your packets are reaching 
> the server, but when it tries to respond it is sending the packets to 
> 192.168.0.100 <http://192.168.0.100> which is (obviously) not what you 
> want to happen.  The solution to this (typically) is to add "NAT=yes" 
> to sip.conf in the general section.
>
> Give that a try and see what your result is.
>
> Nick
>
> On 4/13/07, *Alex Balashov* <abalashov at evaristesys.com 
> <mailto:abalashov at evaristesys.com>> wrote:
>
>
>     On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:
>
>     > mmm are you sure that asterisk-gui generate it on the sip.conf file?
>     > cause i see a new file called users.conf, and i can see the sip
>     users
>     > on it. Anybody uses asterisk now and can check it please??
>
>        Hmm.  I use 1.4.x here and installed the stock config file samples
>     bundle, and there's no trace of users.conf.
>
>        But then again, I have never used any GUI configurator, so I'm
>     not in the
>     best position to know what sort of structure and metadata it
>     generates.
>
>     --
>     Alex Balashov <sasha at presidium.org <mailto:sasha at presidium.org>>
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