thr3ads.net - asterisk users - [Asterisk-Users] SIP through freeBSD NAT [Jan 2006]

If this information is useful, please help other people find it:
Share via:

Alyed Tzompa

2006-Jan-02 12:32 UTC

[Asterisk-Users] SIP through freeBSD NAT

Hi everyone

 My problem is the following:
 I'm trying to make a call from a sip phone (SJphone) behind a Restricted
Cone NAT towards and Asterisk behind another NAT
(a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the
remote Asterisk.

 My soft phone registers without problems to the Asterisk but once the record
starts to play I get a hangup. SJphone outputs
"End reason:  Unable to agree on media streams". 

 I'm forwarding SIP and IAX ports from the remote NAT towards the Asterisk
box (i've tryied it with IAX with no problems) using
the following config in the remote NAT:

 /etc/pf.conf
 .....
 # outgoing UDP port 5060 connections use source port 5060 on firewall
 nat on $ext_if inet proto udp from any port = 5060 to any -> ($ext_if) port
5060

 # Redirect all trafic from NAT:asterisk_port to 192.168.1.5:asterisk_port
 rdr on $ext_if inet proto { tcp, udp } from any to any port 4569 ->
192.168.1.5 port 4569
 rdr on $ext_if inet proto { tcp, udp } from any to $ext_if port 5060 ->
192.168.1.5 port 5060
 rdr on $ext_if inet proto { tcp , udp} from any to any port 10000:20000 ->
192.168.1.5 port 10000:20000

 # Let the Internet see our services
  pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port
4569 keep state
 pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port
5060 keep state
 .....
 ------------------------------------------------------------------

 I think the problem might relay in this "pass in log-all" since once
I commented the last line and the SJphone was unable to
register, but I haven't found a way to set up a range using this
"pass" command (it complains saying that the " : " is valid
only
with the "rdr " command) but I haven't found info explaining why I
should (or shouldn't) use this "pass" command.

 My Asterisk config is:

 sip.conf
 [general]
 port=5060
 externip = www.theip.net
 localnet = 192.168.1.0
 localmask = 255.255.255.0
 allow=all

 [user]
....
 nat=yes
 canreinvite=no
 host=dynamic
 --------------------------------------------
extensions.conf

 exten => 400,1,Background(/var/lib/asterisk/sounds/myrecord)
 exten => 400,2,Hangup
 exten => 400,102,Hangup
 ---------------------------------------------

 Thanx a lot!
 ww6 
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060102/6cc76198/attachment.htm

Eric "ManxPower" Wieling

2006-Jan-02 22:30 UTC

head link

[Asterisk-Users] SIP through freeBSD NAT

Alyed Tzompa wrote:> sip.conf
> [general]
> port=5060
> externip = www.theip.net
> localnet = 192.168.1.0
> localmask = 255.255.255.0
> allow=all
Don't use allow=all.  Use disallow=all and then allow= line for the 
specific codec you want to use.

Alyed Tzompa

2006-Jan-03 11:10 UTC

head link

[Asterisk-Users] SIP through freeBSD NAT

made the changes in sip.conf so now it reads:

 disallow=all
 allow ilbc

 now I when the call is placed it is not hanged up, but I cannot hear anything.
I think it's becasue Asterisk is sending the RTP's to a wrong address
(my
 internal IP).
 Looked at the sip debug and got the following:

     -- Executing BackGround("SIP/alyed-5a8d",
"/var/lib/asterisk/sounds/testt") in new stack
 We're at 200.78.243.12 port 13458
 Answering with preferred capability 0x400(ILBC)
 Answering with non-codec capability 0x1(G723)
 Reliably Transmitting (NAT):
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP
90.0.0.10;branch=z9hG4bK5a00000a000000c043bab4f9390f1bef000002ef;received=201.127.53.246;rport=5060
 From:
"unknown"<sip:alyed@www.myip.net:5060>;tag=2438130825771721203
 To: <sip:400@www.myip.net:5060>;tag=as7222f729
 Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
 CSeq: 2 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact: <sip:400@200.78.243.12>
 Content-Type: application/sdp
 Content-Length: 220

 v=0
 o=root 17028 17028 IN IP4 200.78.243.12
 s=session
 c=IN IP4 200.78.243.12
 t=0 0
 m=audio 13458 RTP/AVP 97 101
 a=rtpmap:97 iLBC/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -

  to 201.127.53.246:5060
     -- Playing '/var/lib/asterisk/sounds/test' (language 'en')
 Integra2*CLI>

 Sip read:
 ACK sip:400@200.78.243.12 SIP/2.0
 Via: SIP/2.0/UDP
90.0.0.10;rport;branch=z9hG4bK5a00000a000000c043bab4f944b4f6f3000002f2
 Content-Length: 0
 Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
 CSeq: 2 ACK
 From:
"unknown"<sip:alyed@www.myip.net:5060>;tag=2438130825771721203
 Max-Forwards: 70
 To: <sip:400@www.myipl.net:5060>;tag=as7222f729
 User-Agent: SJphone/1.60.299a/L (SJ Labs)

 9 headers, 0 lines

 any ideas? 

----------------------------------------
Return-Path: <eric@fnords.org> Mon Jan 02 22:32:10 2006
Received: from bourbon.fnords.org [209.16.72.158] by mail11.webcontrolcenter.com
with SMTP;
 Mon, 2 Jan 2006 22:32:10 -0700
Received: from [172.18.3.242] (24-179-48-91.static.slid.la.charter.com
[24.179.48.91])
 (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits))
 (No client certificate requested)
 by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88;
 Mon, 2 Jan 2006 23:32:08 -0600 (CST)
Message-ID: <43BA0BF1.3070104@fnords.org>
Date: Mon, 02 Jan 2006 23:30:25 -0600
From: "Eric \"ManxPower\" Wieling" <eric@fnords.org>
User-Agent: Thunderbird 1.5 (Windows/20051201)
MIME-Version: 1.0
To: alyed.tzompa@simitel.com,
 Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
References: <da06c3886a364e62ba1f92b767917836@simitel.com>
In-Reply-To: <da06c3886a364e62ba1f92b767917836@simitel.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Content-Transfer-Encoding: 7bit
X-SmarterMail-Spam: SPF_None 

Alyed Tzompa wrote:> sip.conf
> [general]
> port=5060
> externip = www.theip.net
> localnet = 192.168.1.0
> localmask = 255.255.255.0
> allow=all
Don't use allow=all. Use disallow=all and then allow= line for the 
specific codec you want to use.


-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060103/3389e246/attachment.htm

Alyed Tzompa

2006-Jan-03 12:07 UTC

head link

[Asterisk-Users] SIP through freeBSD NAT

SJphone supports ilbc, anyway tryed it with ulaw, alaw and gsm (all of them
supported by SJphone), but the behaviour is the same. That's why I thought
 this sould be a RTP addressing stuff

Alyed 

----------------------------------------
Return-Path: <eric@fnords.org> Tue Jan 03 11:46:59 2006
Received: from bourbon.fnords.org [209.16.72.158] by mail11.webcontrolcenter.com
with SMTP;
 Tue, 3 Jan 2006 11:46:59 -0700
Received: from [172.16.13.73] (unknown [172.16.13.73])
 (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits))
 (No client certificate requested)
 by bourbon.fnords.org (Postfix) with ESMTP id 91F6386;
 Tue, 3 Jan 2006 12:46:58 -0600 (CST)
Message-ID: <43BAC63B.20608@fnords.org>
Date: Tue, 03 Jan 2006 12:45:15 -0600
From: "Eric \"ManxPower\" Wieling" <eric@fnords.org>
User-Agent: Thunderbird 1.5 (Windows/20051201)
MIME-Version: 1.0
To: alyed.tzompa@simitel.com
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
References: <905b384491eb4829ab5dbd04fbc5fc24@simitel.com>
In-Reply-To: <905b384491eb4829ab5dbd04fbc5fc24@simitel.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Content-Transfer-Encoding: 7bit
X-SmarterMail-Spam: SPF_None 

Use a codec your phone supports like ulaw.

Alyed Tzompa wrote:> made the changes in sip.conf so now it reads:
> 
> disallow=all
> allow ilbc
> 
> now I when the call is placed it is not hanged up, but I cannot hear 
> anything. I think it's becasue Asterisk is sending the RTP's to a
wrong
> address (my
> internal IP).
> Looked at the sip debug and got the following:
> 
> -- Executing BackGround("SIP/alyed-5a8d", 
> "/var/lib/asterisk/sounds/testt") in new stack
> We're at 200.78.243.12 port 13458
> Answering with preferred capability 0x400(ILBC)
> Answering with non-codec capability 0x1(G723)
> Reliably Transmitting (NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 
>
90.0.0.10;branch=z9hG4bK5a00000a000000c043bab4f9390f1bef000002ef;received=201.127.53.246;rport=5060
> From: "unknown";tag=2438130825771721203
> To: ;tag=as7222f729
> Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: 
> Content-Type: application/sdp
> Content-Length: 220
> 
> v=0
> o=root 17028 17028 IN IP4 200.78.243.12
> s=session
> c=IN IP4 200.78.243.12
> t=0 0
> m=audio 13458 RTP/AVP 97 101
> a=rtpmap:97 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> 
> to 201.127.53.246:5060
> -- Playing '/var/lib/asterisk/sounds/test' (language 'en')
> Integra2*CLI>
> 
> Sip read:
> ACK sip:400@200.78.243.12 SIP/2.0
> Via: SIP/2.0/UDP 
> 90.0.0.10;rport;branch=z9hG4bK5a00000a000000c043bab4f944b4f6f3000002f2
> Content-Length: 0
> Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
> CSeq: 2 ACK
> From: "unknown";tag=2438130825771721203
> Max-Forwards: 70
> To: ;tag=as7222f729
> User-Agent: SJphone/1.60.299a/L (SJ Labs)
> 
> 
> 9 headers, 0 lines
> 
> 
> 
> any ideas?
> 
> 
> 
> ------------------------------------------------------------------------
> Return-Path:  Mon Jan 02 22:32:10 2006
> Received: from bourbon.fnords.org [209.16.72.158] by 
> mail11.webcontrolcenter.com with SMTP;
> Mon, 2 Jan 2006 22:32:10 -0700
> Received: from [172.18.3.242] (24-179-48-91.static.slid.la.charter.com 
> [24.179.48.91])
> (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits))
> (No client certificate requested)
> by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88;
> Mon, 2 Jan 2006 23:32:08 -0600 (CST)
> Message-ID: <43BA0BF1.3070104@fnords.org>
> Date: Mon, 02 Jan 2006 23:30:25 -0600
> From: "Eric \"ManxPower\" Wieling" 
> User-Agent: Thunderbird 1.5 (Windows/20051201)
> MIME-Version: 1.0
> To: alyed.tzompa@simitel.com,
> Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
> References: 
> In-Reply-To: 
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> Content-Transfer-Encoding: 7bit
> X-SmarterMail-Spam: SPF_None
> 
> Alyed Tzompa wrote:
> > sip.conf
> > [general]
> > port=5060
> > externip = www.theip.net
> > localnet = 192.168.1.0
> > localmask = 255.255.255.0
> > allow=all
> 
> Don't use allow=all. Use disallow=all and then allow= line for the
> specific codec you want to use.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060103/634b5ea1/attachment.htm

Alyed Tzompa

2006-Jan-03 12:18 UTC

head link

[Asterisk-Users] SIP through freeBSD NAT

Tryed what Eric suggested in the other thread (changing  in sip.conf:  allow=all
for     disallow=all allow=somecodec)
 so now the call is not being hanged up, but cannot hear anything. Tryied it
with ilbc,alaw, ulaw and gsm

 I still think it sould be a matter of RTP addressing since I get the following
after a sip debug :

     -- Executing BackGround("SIP/alyed-5a8d",
"/var/lib/asterisk/sounds/test") in new stack
 We're at 200.78.243.12 port 13458
 Answering with preferred capability 0x400(ILBC)
 Answering with non-codec capability 0x1(G723)
 Reliably Transmitting (NAT):
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP
90.0.0.10;branch=z9hG4bK5a00000a000000c043bab4f9390f1bef000002ef;received=201.127.53.246;rport=5060
 From:
"unknown"<sip:alyed@www.myip.net:5060>;tag=2438130825771721203
 To: <sip:400@www.myip.net:5060>;tag=as7222f729
 Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
 CSeq: 2 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact: <sip:400@200.78.243.12>
 Content-Type: application/sdp
 Content-Length: 220

 v=0
 o=root 17028 17028 IN IP4 200.78.243.12
 s=session
 c=IN IP4 200.78.243.12
 t=0 0
 m=audio 13458 RTP/AVP 97 101
 a=rtpmap:97 iLBC/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -

  to 201.127.53.246:5060
     -- Playing '/var/lib/asterisk/sounds/test' (language 'en')
 Integra2*CLI>

 Sip read:
 ACK sip:400@200.78.243.12 SIP/2.0
 Via: SIP/2.0/UDP
90.0.0.10;rport;branch=z9hG4bK5a00000a000000c043bab4f944b4f6f3000002f2
 Content-Length: 0
 Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
 CSeq: 2 ACK
 From:
"unknown"<sip:alyed@www.myip.net:5060>;tag=2438130825771721203
 Max-Forwards: 70
 To: <sip:400@www.myip.net:5060>;tag=as7222f729
 User-Agent: SJphone/1.60.299a/L (SJ Labs)

 9 headers, 0 lines

 think it is addressing the rtp to my internal IP, but don't know who can I
address it in the right way. I'm using the default STUN config in the
SJphone :
  STUN server address --> stun.softjoys.com :3478, refresh time out
-->1200000  conclusive response timeout-->0 retrunsmissions number -->
13
 and nat= yes in the sip.conf

 But still no sound in my endpoint

Alyed 

Alyed Tzompa Sosa

Simitel
VoIP developer
+52 (55) 24 52 43 90 Ext. (107)
alyed.tzompa@simitel.com 

----------------------------------------
Return-Path: <tony@jago.net> Mon Jan 02 13:29:11 2006
Received: from mail08.syd.optusnet.com.au [211.29.132.189] by
mail11.webcontrolcenter.com with SMTP;
 Mon, 2 Jan 2006 13:29:11 -0700
Received: from router.vpn.marvy.net (c220-237-85-239.kelvn1.qld.optusnet.com.au
[220.237.85.239])
 by mail08.syd.optusnet.com.au (8.12.11/8.12.11) with ESMTP id k02KT9sT019578
 (version=TLSv1/SSLv3 cipher=EDH-RSA-DES-CBC3-SHA bits=168 verify=NO)
 for <alyed.tzompa@simitel.com>; Tue, 3 Jan 2006 07:29:09 +1100
Received: from alcan (alcan.vpn.cc.uq.edu.au [192.168.136.2])
 by router.vpn.marvy.net (8.13.3/8.13.3) with SMTP id k02KT88l022322
 for <alyed.tzompa@simitel.com>; Tue, 3 Jan 2006 06:29:08 +1000 (EST)
 (envelope-from tony@jago.net)
Message-ID: <001901c60fdb$3a781300$0288a8c0@alcan>
From: "Tony Jago" <tony@jago.net>
To: <alyed.tzompa@simitel.com>
References: <da06c3886a364e62ba1f92b767917836@simitel.com>
Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
Date: Tue, 3 Jan 2006 06:29:27 +1000
MIME-Version: 1.0
Content-Type: multipart/alternative;
 boundary="----=_NextPart_000_0016_01C6102F.0BCE3AD0"
X-Priority: 3
X-MSMail-Priority: Normal
X-Mailer: Microsoft Outlook Express 6.00.2900.2670
X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2670
X-SmarterMail-Spam: SPF_None 

 "Unable to agree on media streams" means that the two devices
can't pick a common codec. This shouldn't have anything to do with the
firewall.   ----- Original Message -----  From: Alyed Tzompa  To:
asterisk-users@lists.digium.com  Sent: Tuesday, January 03, 2006 5:32 AM
Subject: [Asterisk-Users] SIP through freeBSD NAT
Hi everyone

My problem is the following:
I'm trying to make a call from a sip phone (SJphone) behind a Restricted
Cone NAT towards and Asterisk behind another NAT
(a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the
remote Asterisk.

My soft phone registers without problems to the Asterisk but once the record
starts to play I get a hangup. SJphone outputs
"End reason:  Unable to agree on media streams". 

I'm forwarding SIP and IAX ports from the remote NAT towards the Asterisk
box (i've tryied it with IAX with no problems) using
the following config in the remote NAT:

/etc/pf.conf
.....
# outgoing UDP port 5060 connections use source port 5060 on firewall
nat on $ext_if inet proto udp from any port = 5060 to any -> ($ext_if) port
5060

# Redirect all trafic from NAT:asterisk_port to 192.168.1.5:asterisk_port
rdr on $ext_if inet proto { tcp, udp } from any to any port 4569 ->
192.168.1.5 port 4569
rdr on $ext_if inet proto { tcp, udp } from any to $ext_if port 5060 ->
192.168.1.5 port 5060
rdr on $ext_if inet proto { tcp , udp} from any to any port 10000:20000 ->
192.168.1.5 port 10000:20000

# Let the Internet see our services
  pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port
4569 keep state
pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port
5060 keep state
.....
------------------------------------------------------------------

I think the problem might relay in this "pass in log-all" since once I
commented the last line and the SJphone was unable to
register, but I haven't found a way to set up a range using this
"pass" command (it complains saying that the " : " is valid
only
with the "rdr " command) but I haven't found info explaining why I
should (or shouldn't) use this "pass" command.

My Asterisk config is:

sip.conf
[general]
port=5060
externip = www.theip.net
localnet = 192.168.1.0
localmask = 255.255.255.0
allow=all

[user]
....
nat=yes
canreinvite=no
host=dynamic
--------------------------------------------
extensions.conf

exten => 400,1,Background(/var/lib/asterisk/sounds/myrecord)
exten => 400,2,Hangup
exten => 400,102,Hangup
---------------------------------------------

Thanx a lot!
ww6 

----------------------------------------

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060103/040668b9/attachment.htm

Alyed Tzompa

2006-Jan-03 13:11 UTC

head link

[Asterisk-Users] SIP through freeBSD NAT

it does support ilbc, alaw, ulaw and gsm. I've tryied all but get the same
results with all of them the phone doesn't hangs up, but cannot hear
anything in my
 endpoint. 

Alyed  

----------------------------------------
Return-Path: <asterisk-users-bounces@lists.digium.com> Tue Jan 03 12:47:02
2006
Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
mail11.webcontrolcenter.com with SMTP;
 Tue, 3 Jan 2006 12:47:02 -0700
Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])

Eric "ManxPower" Wieling wrote:> Use a codec your phone supports like ulaw.
> 
Assuming he is using SJphone, that I understand, would support iLBC even 
in the free version ?
> Alyed Tzompa wrote:
> 
>> made the changes in sip.conf so now it reads:
>>
>> disallow=all
>> allow ilbc
>>
>> now I when the call is placed it is not hanged up, but I cannot hear 
>> anything. I think it's becasue Asterisk is sending the RTP's to
a
>> wrong address (my
>> internal IP).
>> Looked at the sip debug and got the following:
>>
>> -- Executing BackGround("SIP/alyed-5a8d", 
>> "/var/lib/asterisk/sounds/testt") in new stack
>> We're at 200.78.243.12 port 13458
>> Answering with preferred capability 0x400(ILBC)
>> Answering with non-codec capability 0x1(G723)
>> Reliably Transmitting (NAT):
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 
>>
90.0.0.10;branch=z9hG4bK5a00000a000000c043bab4f9390f1bef000002ef;received=201.127.53.246;rport=5060
>>
>> From: "unknown";tag=2438130825771721203
>> To: ;tag=as7222f729
>> Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
>> CSeq: 2 INVITE
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: 
>> Content-Type: application/sdp
>> Content-Length: 220
>>
>> v=0
>> o=root 17028 17028 IN IP4 200.78.243.12
>> s=session
>> c=IN IP4 200.78.243.12
>> t=0 0
>> m=audio 13458 RTP/AVP 97 101
>> a=rtpmap:97 iLBC/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>>
>> to 201.127.53.246:5060
>> -- Playing '/var/lib/asterisk/sounds/test' (language
'en')
>> Integra2*CLI>
>>
>> Sip read:
>> ACK sip:400@200.78.243.12 SIP/2.0
>> Via: SIP/2.0/UDP 
>> 90.0.0.10;rport;branch=z9hG4bK5a00000a000000c043bab4f944b4f6f3000002f2
>> Content-Length: 0
>> Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10
>> CSeq: 2 ACK
>> From: "unknown";tag=2438130825771721203
>> Max-Forwards: 70
>> To: ;tag=as7222f729
>> User-Agent: SJphone/1.60.299a/L (SJ Labs)
>>
>>
>> 9 headers, 0 lines
>>
>>
>>
>> any ideas?
>>
>>
>>
>>
------------------------------------------------------------------------
>> Return-Path:  Mon Jan 02 22:32:10 2006
>> Received: from bourbon.fnords.org [209.16.72.158] by 
>> mail11.webcontrolcenter.com with SMTP;
>> Mon, 2 Jan 2006 22:32:10 -0700
>> Received: from [172.18.3.242] (24-179-48-91.static.slid.la.charter.com 
>> [24.179.48.91])
>> (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits))
>> (No client certificate requested)
>> by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88;
>> Mon, 2 Jan 2006 23:32:08 -0600 (CST)
>> Message-ID: <43BA0BF1.3070104@fnords.org>
>> Date: Mon, 02 Jan 2006 23:30:25 -0600
>> From: "Eric \"ManxPower\" Wieling" 
>> User-Agent: Thunderbird 1.5 (Windows/20051201)
>> MIME-Version: 1.0
>> To: alyed.tzompa@simitel.com,
>> Asterisk Users Mailing List - Non-Commercial Discussion 
>> 
>> Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
>> References: 
>> In-Reply-To: 
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>> Content-Transfer-Encoding: 7bit
>> X-SmarterMail-Spam: SPF_None
>>
>> Alyed Tzompa wrote:
>> > sip.conf
>> > [general]
>> > port=5060
>> > externip = www.theip.net
>> > localnet = 192.168.1.0
>> > localmask = 255.255.255.0
>> > allow=all
>>
>> Don't use allow=all. Use disallow=all and then allow= line for the
>> specific codec you want to use._______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users


-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060103/9858f2e9/attachment.htm

asterisk users - Jan 2006 - SIP through freeBSD NAT

[Asterisk-Users] SIP through freeBSD NAT

[Asterisk-Users] SIP through freeBSD NAT

[Asterisk-Users] SIP through freeBSD NAT

[Asterisk-Users] SIP through freeBSD NAT

[Asterisk-Users] SIP through freeBSD NAT

[Asterisk-Users] SIP through freeBSD NAT