[asterisk-users] SIP REGISTRATION TIME OUT

Thu Apr 19 07:31:17 MST 2007

hi, to get it work i change under sip.conf

nat: route
Allow RTP reinvite:update

with that i can hear, without dmz... but... why?

2007/4/19, Manolet Gmail <manolet at gmail.com>:
> 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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