[asterisk-users] SIP REGISTRATION TIME OUT

[Manolet Gmail]

Hi, now i can log in ok on my xlite, somebody calls me and everythink
its okey. i hear and the caller hear. (the pc with the xlite have
DMZ).

But now i close xlite and put the same extension on a grandstream 286
(dont have DMZ). When somebody calls me the caller can hear me. but i
cant hear!

whats the problem? with other providers i can talk using my
grandstream 286 without give it dmz or changing the configuration on
my router.

i hopes somebody can help me!

2007/4/14, dave cantera <david.cantera at iacnet.net>:
> 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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