endpoint.
Alyed
Return-Path: <[EMAIL PROTECTED]> Tue Jan 03 12:47:02 2006
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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"
>> To:
>> Call-ID: [EMAIL PROTECTED]
>> 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:[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"
>> Max-Forwards: 70
>> To:
>> User-Agent: SJphone/1.60.299a/L (SJ Labs)
>>
>>
>> 9 headers, 0 lines
>>
>>
>>
>> any ideas?
>>
>>
>>
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>> Mon, 2 Jan 2006 23:32:08 -0600 (CST)
>> Message-ID: <[EMAIL PROTECTED]>
>> Date: Mon, 02 Jan 2006 23:30:25 -0600
>> From: "Eric \"ManxPower\" Wieling"
>> User-Agent: Thunderbird 1.5 (Windows/20051201)
>> MIME-Version: 1.0
>> To: [EMAIL PROTECTED],
>> Asterisk Users Mailing List - Non-Commercial Discussion
>>
>> Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
>> References:
>> In-Reply-To:
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
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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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