thr3ads.net - asterisk users - [Asterisk-Users] Asterisk behind NAT << How to do it. [Nov 2003]

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Leif Madsen

2003-Nov-27 21:10 UTC

[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@hacklocalhost.com>
http://www.hacklocalhost.com

Cees de Groot

2003-Nov-28 02:15 UTC

head link

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

Leif Madsen  <asterisk-users@lists.digium.com>
said:>outside_addr=216.239.33.100     ; this can also be a FQDN! ie.
>                                ; my.domain.com
>Which might be a problem for dynamic environments, but still nice that
someone implemented this patch. I'm sure going to take a look at it -
we're currently happy with DIAX calling in to my NAT'ed * setup, but it
would be handy to be able to use SIP as well - thanks!

(that leaves me with two options - I noticed my Speedtouch 510 also
supports SIP in its NAT stuff)

-- 
Cees de Groot               http://www.tric.nl     <cg@tric.nl>
tric, the new way           helpdesk/ticketing software, VoIP/CTI, 
                            web applications, custom development

Darren McIntosh

2003-Nov-28 16:23 UTC

head link

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

> Message: 9
> From: Leif Madsen <leif@hacklocalhost.com>
> To: asterisk-users@lists.digium.com
> Organization: http://www.hacklocalhost.com
> Date: 27 Nov 2003 23:10:42 -0500
> Subject: [Asterisk-Users] Asterisk behind NAT << How to do it.
> Reply-To: asterisk-users@lists.digium.com
>
> 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@hacklocalhost.com>
> http://www.hacklocalhost.com
>I can confirm this works for my NAT'd setup as well. Just one comment though
that the inside_net variable is your internal subnet address not the
asterisk server address.

cheers,
darren

Arnold Ligtvoet

2003-Dec-02 13:55 UTC

head link

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

> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Leif Madsen
> Sent: vrijdag 28 november 2003 5:11
> To: asterisk-users@lists.digium.com
> 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.
>
Hi Leif,

I tried the patch. Installed it exactly as described per your email. Thought
that you might be interested that it works for me as well. Like a charm, I
can finally call FWD numbers like 10001 and 612 (speaking clock demo).

BTW: For anybody wanting to install this, if your version of chan_sip.c is
older than the one described, first use 'cvs update -C
asterisk/channels/chan_sip.c'.

Thanks,
Arnold Ligtvoet.

Todd Wallace

2003-Dec-03 14:12 UTC

head link

[Asterisk-Users] phone port on the x100p

Can the phone port on the x100p be an addressable extension on asterisk?  I
want to plug our conference phone into that phone jack as it is an analog
phone.


Todd Wallace

Richard Alexander

2003-Dec-03 14:29 UTC

head link

[Asterisk-Users] phone port on the x100p

> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of Todd Wallace
> Sent: Wednesday, December 03, 2003 4:12 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] phone port on the x100p
> 
> Can the phone port on the x100p be an addressable extension on
asterisk?> I
> want to plug our conference phone into that phone jack as it is an
analog> phone.
> 
> 
> Todd Wallace
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
In a word.... No. 

You can use it to check the POTS line and not much else.

Leif Madsen

2003-Dec-07 14:22 UTC

head link

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

On Wed, 2003-12-03 at 15:34, William Waites wrote:
> > > localnet= internal ip of * machine?
> 
> localnet should be the internal network address not the internal
> ip address. i.e. if your asterisk server is 192.168.0.245, localnet
> should be 192.168.0.0
Agreed, I was wrong before :)

-- 
Leif Madsen <leif@hacklocalhost.com>
http://www.hacklocalhost.com

listas iPfone

2003-Dec-09 03:10 UTC

head link

[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@hacklocalhost.com>
To: <asterisk-users@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@hacklocalhost.com>
> http://www.hacklocalhost.com
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Balaji NJL

2003-Dec-27 01:33 UTC

head link

[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@ipfone.com.br>
To: <asterisk-users@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@hacklocalhost.com>
> To: <asterisk-users@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@hacklocalhost.com>
> > http://www.hacklocalhost.com
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
http://lists.digium.com/mailman/listinfo/asterisk-users> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users


__________________________________
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New Yahoo! Photos - easier uploading and sharing.
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Craig Waddington

2003-Dec-27 04:43 UTC

head link

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

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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Balaji NJL
Sent: 27 December 2003 08:34
To: asterisk-users@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@ipfone.com.br>
To: <asterisk-users@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@hacklocalhost.com>
> To: <asterisk-users@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@hacklocalhost.com>
> > http://www.hacklocalhost.com
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
http://lists.digium.com/mailman/listinfo/asterisk-users> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users


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_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users

Balaji NJL

2003-Dec-27 10:35 UTC

head link

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

thats cool. i ll try that too. Whats ur * version.

if thats the case what is this patch for. Is bug 104
already approved and in
production.

-B

----- Original Message ----- 
From: "Craig Waddington" <craig@xmbsystems.com>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT <<
How to do it.

> 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@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On
Behalf Of Balaji NJL> Sent: 27 December 2003 08:34
> To: asterisk-users@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@ipfone.com.br>
> To: <asterisk-users@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@hacklocalhost.com>
> > To: <asterisk-users@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@hacklocalhost.com>
> > > http://www.hacklocalhost.com
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > >
>
http://lists.digium.com/mailman/listinfo/asterisk-users> > >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users>
>
> __________________________________
> Do you Yahoo!?
> New Yahoo! Photos - easier uploading and sharing.
> http://photos.yahoo.com/
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users


__________________________________
Do you Yahoo!?
New Yahoo! Photos - easier uploading and sharing.
http://photos.yahoo.com/

Balaji NJL

2004-Jan-11 03:29 UTC

head link

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

Hi All,

i just applied this patch. i need to test whether its
working. Can someone
connect to my server and leave me a vm at extension
2000.

Server : ojoobala.com

Phone
    Extension : 2005
    pwd   : mytest
    auth: md5.

pl leave a vm on extension 2000.

thanks a lot,
-B
----- Original Message ----- 
From: "listas iPfone" <listas@ipfone.com.br>
To: <asterisk-users@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@hacklocalhost.com>
> To: <asterisk-users@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@hacklocalhost.com>
> > http://www.hacklocalhost.com
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
http://lists.digium.com/mailman/listinfo/asterisk-users> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users


__________________________________
Do you Yahoo!?
New Yahoo! Photos - easier uploading and sharing.
http://photos.yahoo.com/

Balaji NJL

2004-Jan-11 03:34 UTC

head link

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

i like the idea of not requiring to open 10000 ports
in the firewall.

Do i need to change rtf.conf to from 10000 - 20000 to
16384 and 16394.

thanks,
-B 
----- Original Message ----- 
From: "Craig Waddington" <craig@xmbsystems.com>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT <<
How to do it.

> 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@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On
Behalf Of Balaji NJL> Sent: 27 December 2003 08:34
> To: asterisk-users@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@ipfone.com.br>
> To: <asterisk-users@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@hacklocalhost.com>
> > To: <asterisk-users@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@hacklocalhost.com>
> > > http://www.hacklocalhost.com
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > >
>
http://lists.digium.com/mailman/listinfo/asterisk-users> > >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users> 
> 
> __________________________________
> Do you Yahoo!?
> New Yahoo! Photos - easier uploading and sharing.
> http://photos.yahoo.com/
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
__________________________________
Do you Yahoo!?
New Yahoo! Photos - easier uploading and sharing.
http://photos.yahoo.com/

Craig Waddington

2004-Jan-11 05:05 UTC

head link

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

Balaji.

I just left rtf.conf at default. Though I guess it wouldn't hurt to
change it to test.

Does it currently work for you with the settings I provided?

Craig.


www.ntfs.org


-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Balaji NJL
Sent: 11 January 2004 10:35
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

i like the idea of not requiring to open 10000 ports
in the firewall.

Do i need to change rtf.conf to from 10000 - 20000 to
16384 and 16394.

thanks,
-B 
----- Original Message ----- 
From: "Craig Waddington" <craig@xmbsystems.com>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, December 27, 2003 3:43 AM
Subject: RE: [Asterisk-Users] Asterisk behind NAT <<
How to do it.

> 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@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On
Behalf Of Balaji NJL> Sent: 27 December 2003 08:34
> To: asterisk-users@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@ipfone.com.br>
> To: <asterisk-users@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@hacklocalhost.com>
> > To: <asterisk-users@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@hacklocalhost.com>
> > > http://www.hacklocalhost.com
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > >
>
http://lists.digium.com/mailman/listinfo/asterisk-users> > >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users> 
> 
> __________________________________
> Do you Yahoo!?
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> http://photos.yahoo.com/
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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