[Asterisk-Users] SIP through freeBSD NAT

[Julio Arruda]

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"<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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>> Message-ID: <43BA0BF1.3070104 at fnords.org>
>> Date: Mon, 02 Jan 2006 23:30:25 -0600
>> From: "Eric \"ManxPower\" Wieling" <eric at fnords.org>
>> User-Agent: Thunderbird 1.5 (Windows/20051201)
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>> To: alyed.tzompa at simitel.com,
>> Asterisk Users Mailing List - Non-Commercial Discussion 
>> <asterisk-users at lists.digium.com>
>> Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
>> References: <da06c3886a364e62ba1f92b767917836 at simitel.com>
>> In-Reply-To: <da06c3886a364e62ba1f92b767917836 at simitel.com>
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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.