[Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

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 at www.myip.net:5060>;tag=2438130825771721203
 To: <sip:400 at www.myip.net:5060>;tag=as7222f729
 Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A at 90.0.0.10
 CSeq: 2 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact: <sip:400 at 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 at 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 at 90.0.0.10
 CSeq: 2 ACK
 From: "unknown"<sip:alyed at www.myip.net:5060>;tag=2438130825771721203
 Max-Forwards: 70
 To: <sip:400 at www.myipl.net:5060>;tag=as7222f729
 User-Agent: SJphone/1.60.299a/L (SJ Labs)

 9 headers, 0 lines

 any ideas? 

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 Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
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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.


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