[Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

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 

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From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
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To: alyed.tzompa at simitel.com
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Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
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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 at 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 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";tag=2438130825771721203
> Max-Forwards: 70
> To: ;tag=as7222f729
> User-Agent: SJphone/1.60.299a/L (SJ Labs)
> 
> 
> 9 headers, 0 lines
> 
> 
> 
> any ideas?
> 
> 
> 
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> From: "Eric \"ManxPower\" Wieling" 
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> To: alyed.tzompa at simitel.com,
> Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> 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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