[Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

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  

----------------------------------------
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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 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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>> Date: Mon, 02 Jan 2006 23:30:25 -0600
>> From: "Eric \"ManxPower\" Wieling" 
>> User-Agent: Thunderbird 1.5 (Windows/20051201)
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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
>> References: 
>> In-Reply-To: 
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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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