<font face="arial" size="2">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<br /> this sould be a RTP addressing stuff<br /> <br /><br />Alyed <br /> <br /> </font><font face="Tahoma, Arial, Sans-Serif" size="2"><hr align="center" size="2" width="100%" />Return-Path: <eric@fnords.org> Tue Jan 03 11:46:59 2006<br />Received: from bourbon.fnords.org [209.16.72.158] by mail11.webcontrolcenter.com with SMTP;<br /> Tue, 3 Jan 2006 11:46:59 -0700<br />Received: from [172.16.13.73] (unknown [172.16.13.73])<br /> (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits))<br /> (No client certificate requested)<br /> by bourbon.fnords.org (Postfix) with ESMTP id 91F6386;<br /> Tue, 3 Jan 2006 12:46:58 -0600 (CST)<br />Message-ID: <43BAC63B.20608@fnords.org><br />Date: Tue, 03 Jan 2006 12:45:15 -0600<br />From: "Eric \"ManxPower\" Wieling" <eric@fnords.org><br />User-Agent: Thunderbird 1.5 (Windows/20051201)<br />MIME-Version: 1.0<br />To: alyed.tzompa@simitel.com<br />Cc: asterisk-users@lists.digium.com<br />Subject: Re: [Asterisk-Users] SIP through freeBSD NAT<br />References: <905b384491eb4829ab5dbd04fbc5fc24@simitel.com><br />In-Reply-To: <905b384491eb4829ab5dbd04fbc5fc24@simitel.com><br />Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br />Content-Transfer-Encoding: 7bit<br />X-SmarterMail-Spam: SPF_None</font><br /><br />Use a codec your phone supports like ulaw.<br /><br />Alyed Tzompa wrote:<br />> made the changes in sip.conf so now it reads:<br />> <br />> disallow=all<br />> allow ilbc<br />> <br />> now I when the call is placed it is not hanged up, but I cannot hear <br />> anything. I think it's becasue Asterisk is sending the RTP's to a wrong <br />> address (my<br />> internal IP).<br />> Looked at the sip debug and got the following:<br />> <br />> -- Executing BackGround("SIP/alyed-5a8d", <br />> "/var/lib/asterisk/sounds/testt") in new stack<br />> We're at 200.78.243.12 port 13458<br />> Answering with preferred capability 0x400(ILBC)<br />> Answering with non-codec capability 0x1(G723)<br />> Reliably Transmitting (NAT):<br />> SIP/2.0 200 OK<br />> Via: SIP/2.0/UDP <br />> 90.0.0.10;branch=z9hG4bK5a00000a000000c043bab4f9390f1bef000002ef;received=201.127.53.246;rport=5060<br />> From: "unknown"<sip:alyed @www.myip.net:5060="">;tag=2438130825771721203<br />> To: <sip:400 @www.myip.net:5060="">;tag=as7222f729<br />> Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10<br />> CSeq: 2 INVITE<br />> User-Agent: Asterisk PBX<br />> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER<br />> Contact: <sip:400 @200.78.243.12=""><br />> Content-Type: application/sdp<br />> Content-Length: 220<br />> <br />> v=0<br />> o=root 17028 17028 IN IP4 200.78.243.12<br />> s=session<br />> c=IN IP4 200.78.243.12<br />> t=0 0<br />> m=audio 13458 RTP/AVP 97 101<br />> a=rtpmap:97 iLBC/8000<br />> a=rtpmap:101 telephone-event/8000<br />> a=fmtp:101 0-16<br />> a=silenceSupp:off - - - -<br />> <br />> to 201.127.53.246:5060<br />> -- Playing '/var/lib/asterisk/sounds/test' (language 'en')<br />> Integra2*CLI><br />> <br />> Sip read:<br />> ACK sip:400@200.78.243.12 SIP/2.0<br />> Via: SIP/2.0/UDP <br />> 90.0.0.10;rport;branch=z9hG4bK5a00000a000000c043bab4f944b4f6f3000002f2<br />> Content-Length: 0<br />> Call-ID: CAB8D822-1DD1-11B2-B69A-FEE14D7A103A@90.0.0.10<br />> CSeq: 2 ACK<br />> From: "unknown"<sip:alyed @www.myip.net:5060="">;tag=2438130825771721203<br />> Max-Forwards: 70<br />> To: <sip:400 @www.myipl.net:5060="">;tag=as7222f729<br />> User-Agent: SJphone/1.60.299a/L (SJ Labs)<br />> <br />> <br />> 9 headers, 0 lines<br />> <br />> <br />> <br />> any ideas?<br />> <br />> <br />> <br />> ------------------------------------------------------------------------<br />> Return-Path: <eric> Mon Jan 02 22:32:10 2006<br />> Received: from bourbon.fnords.org [209.16.72.158] by <br />> mail11.webcontrolcenter.com with SMTP;<br />> Mon, 2 Jan 2006 22:32:10 -0700<br />> Received: from [172.18.3.242] (24-179-48-91.static.slid.la.charter.com <br />> [24.179.48.91])<br />> (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits))<br />> (No client certificate requested)<br />> by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88;<br />> Mon, 2 Jan 2006 23:32:08 -0600 (CST)<br />> Message-ID: <43BA0BF1.3070104@fnords.org><br />> Date: Mon, 02 Jan 2006 23:30:25 -0600<br />> From: "Eric \"ManxPower\" Wieling" <eric><br />> User-Agent: Thunderbird 1.5 (Windows/20051201)<br />> MIME-Version: 1.0<br />> To: alyed.tzompa@simitel.com,<br />> Asterisk Users Mailing List - Non-Commercial Discussion <br />> <asterisk-users><br />> Subject: Re: [Asterisk-Users] SIP through freeBSD NAT<br />> References: <da06c3886a364e62ba1f92b767917836><br />> In-Reply-To: <da06c3886a364e62ba1f92b767917836><br />> Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br />> Content-Transfer-Encoding: 7bit<br />> X-SmarterMail-Spam: SPF_None<br />> <br />> Alyed Tzompa wrote:<br />> > sip.conf<br />> > [general]<br />> > port=5060<br />> > externip = www.theip.net<br />> > localnet = 192.168.1.0<br />> > localmask = 255.255.255.0<br />> > allow=all<br />> <br />> Don't use allow=all. Use disallow=all and then allow= line for the<br />> specific codec you want to use.<br />> <br /><br /><br /></da06c3886a364e62ba1f92b767917836></da06c3886a364e62ba1f92b767917836></asterisk-users></eric></eric></sip:400></sip:alyed></sip:400></sip:400></sip:alyed>