[Asterisk-Users] Asterisk behind NAT << How to do it.

[Craig Waddington]

Hi

I have SIP working on NAT using X-lite phones. 

On my Cisco 827H ADSL router I forwarded ports 5060, 16394, 16384 to my
* (10.1.0.0).

16394,16384 being RTP.

In X-lite set the RTP port to use 16394 instead of the default 8000.

Works great over the internet. Didn't need patches or anything else.

I hope that helps you.

-C


www.ntfs.org




-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Balaji NJL
Sent: 27 December 2003 08:34
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

Hi All,

i tried to apply this patch and i got the following
error. The chan_sip.c
version i hv is 1.265

hv any one tried this patch on this latest chan_sip
version.

thanks,
-B

chan_sip.o: In function `load_module':
chan_sip.o(.text+0x15ebf): undefined reference to
`ast_rtp_proto_register'
chan_sip.o(.text+0x15ee0): undefined reference to
`ast_register_application'
chan_sip.o: In function `delete_users':
chan_sip.o(.text+0x15fc1): undefined reference to
`ast_free_ha'
chan_sip.o(.text+0x1604d): undefined reference to
`ast_sched_del'
chan_sip.o: In function `prune_peers':
chan_sip.o(.text+0x16167): undefined reference to
`ast_sched_del'
chan_sip.o(.text+0x1618d): undefined reference to
`ast_sched_del'
chan_sip.o: In function `unload_module':
chan_sip.o(.text+0x162bd): undefined reference to
`ast_channel_unregister'
chan_sip.o(.text+0x162ce): undefined reference to
`ast_unregister_application'
chan_sip.o(.text+0x16337): undefined reference to
`ast_softhangup'
chan_sip.o(.text+0x1636c): undefined reference to
`ast_log'
chan_sip.o(.text+0x163ab): undefined reference to
`pthread_cancel'
chan_sip.o(.text+0x163be): undefined reference to
`pthread_kill'
chan_sip.o(.text+0x163d1): undefined reference to
`pthread_join'
chan_sip.o(.text+0x16418): undefined reference to
`ast_log'
chan_sip.o(.text+0x164b8): undefined reference to
`ast_log'
collect2: ld returned 1 exit status
make: *** [chan_sip.so] Error 1

----- Original Message ----- 
From: "listas iPfone" <listas at ipfone.com.br>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, December 09, 2003 2:10 AM
Subject: Re: [Asterisk-Users] Asterisk behind NAT <<
How to do it.


> Hi
>
> The version 1.260 of chan_sip.c already have that
patch?:
>
>
http://bugs.digium.com/file_download.php?file_id=430&type=bug
>
> thanks!
>
> Miklos
>
>
> ----- Original Message ----- 
> From: "Leif Madsen" <leif at hacklocalhost.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, November 28, 2003 2:10 AM
> Subject: [Asterisk-Users] Asterisk behind NAT << How
to do it.
>
>
> > Thanks to ww and his patch on bug #104, I have
successfully implemented
> > Asterisk behind NAT without using STUN or anything
crazy.  It's quite
> > straight forward.
> >
> > Until this gets tested enough and put into CVS,
you will have to patch
> > your chan_sip.c file to do this.  I'm sure within
the next few days this
> > will get put merged into CVS if no one finds any
problems.
> >
> > I tried this on chan_sip.c version 1.249 (the
version the patch was
> > written for) and the latest as of today 1.258. 
Both work great.
> >
> > Open ports 5060 and your RTP range (found in
/etc/asterisk/rtp.conf).
> > Default is 10000 -> 20000
> >
> > Forward ports 5060 and your RTP range to your
internal Asterisk box.
> >
> > For your sip.conf, you need to add three lines:
> >
> > ; sip.conf snippet
> > [general]
> > port=5060                       ; make sure you
have this line :)
> > inside_net=192.168.1.100        ; this is the
internal ip address of
> > the                                ;
> > asterisk server
> > inside_mask=255.255.255.0       ; internal ip
mask.  /24 as this example
> > outside_addr=216.239.33.100     ; this can also be
a FQDN! ie.
> >                                 ; my.domain.com
> > ; ... plus whatever else you have in your sip.conf
> >
> > Download the patch at:
> >
http://bugs.digium.com/file_download.php?file_id=430&type=bug
> >
> > Either update your Asterisk or verify you have at
least version 1.249 of
> > chan_sip.c:
> >
> > cd /usr/src/asterisk/channels/
> > cvs status chan_sip.c
> >
> >
===================================================================
> > File: chan_sip.c        Status: Locally Modified
> >
> >    Working revision:    1.258
> >    Repository revision: 1.258
> > /usr/cvsroot/asterisk/channels/chan_sip.c,v
> >
> > While in pwd /usr/src/asterisk/channels/
> > patch -p0 < /path/to/patch
> >
> > Nothing should fail.
> >
> > cd /usr/src/asterisk/
> > make
> > cp /usr/src/asterisk/channels/chan_sip.so
/usr/lib/asterisk/modules/
> >
> > Restart your Asterisk and try it.  If you want to
call a NAT'd Asterisk
> > box, my Free World Dialup number is 18924. 
Currently online.
> >
> > -- 
> > Leif Madsen <leif at hacklocalhost.com>
> > http://www.hacklocalhost.com
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users


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