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:[EMAIL PROTECTED]:5060>;tag=2438130825771721203
To: <sip:[EMAIL PROTECTED]:5060>;tag=as7222f729
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
90.0.0.10;rport;branch=z9hG4bK5a00000a000000c043bab4f944b4f6f3000002f2
Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
From: "unknown"<sip:[EMAIL PROTECTED]:5060>;tag=2438130825771721203
Max-Forwards: 70
To: <sip:[EMAIL PROTECTED]: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
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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.