<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:tahoma,new york,times,serif;font-size:10pt"><div>I don't have problems when I make calls to the pstn I listen well and people listen to me well, the problem is when I receive a call from the pstn I don't listen anything and they don't listen to me, inside the sip.conf already has configured the values nat, externip localnet .<br><br><br>I believe that the problem is that openser detects as nat the ip of my asterisk, eye > "I have the openser and the mediaproxy with asterisk in the same pc"<br><br><br>### Sip Log Asterisk #### <br><br><--- SIP read from 192.168.10.1:5060 ---><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070<br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes><br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f<br>To:
<sip:113@192.168.10.1>;tag=a72df908ec08f63d<br>Call-ID: 1661965d0c28650d517426333e4a6ae5@192.168.10.1<br>CSeq: 102 INVITE<br>User-Agent: Grandstream GXP2020 1.1.6.16<br>Contact: <sip:113@192.168.10.30:5062;transport=udp><br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE<br>Content-Length: 0<br>P-hint: Onreply-route - fixcontact<br><br><br><-------------><br>--- (12 headers 0 lines) ---<br> -- SIP/openser-08c0ea58 is ringing<br>xserver*CLI><br><--- SIP read from 192.168.10.1:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070<br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes><br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f<br>To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d<br>Call-ID: 1661965d0c28650d517426333e4a6ae5@192.168.10.1<br>CSeq: 102 INVITE<br>User-Agent:
Grandstream GXP2020 1.1.6.16<br>Contact: <sip:113@192.168.10.30:5062;transport=udp><br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE<br>Content-Type: application/sdp<br>Supported: replaces, timer<br>Content-Length: 212<br>P-hint: Onreply-route - fixcontact<br>P-hint: onreply_route|usemediaproxy<br><br>v=0<br>o=113 8000 8000 IN IP4 192.168.10.30<br>s=SIP Call<br>c=IN IP4 192.168.1.64<br>t=0 0<br>m=audio 35004 RTP/AVP 0 101<br>a=sendrecv<br>a=rtpmap:0 PCMU/8000<br>a=ptime:20<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-11<br><br><-------------><br>--- (15 headers 11 lines) ---<br>Found RTP audio format 0<br>Found RTP audio format 101<br>Peer audio RTP is at port 192.168.1.64:35004<br>Found audio description format PCMU for ID 0<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4
(ulaw)<br>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>Peer audio RTP is at port 192.168.1.64:35004<br>list_route: hop: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes><br>set_destination: Parsing <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> for address/port to send to<br>set_destination: set destination to 192.168.10.1, port 5060<br>Transmitting (NAT) to 192.168.10.1:5060:<br>ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0<br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK32b6019c;rport<br>Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes><br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f<br>To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d<br>Contact: <sip:asterisk@192.168.10.1:5070><br>Call-ID: 1661965d0c28650d517426333e4a6ae5@192.168.10.1<br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX<br>Max-Forwards:
70<br>Content-Length: 0<br><br><br>---<br> -- SIP/openser-08c0ea58 answered Zap/4-1<br>xserver*CLI><br><--- SIP read from 192.168.10.1:5060 ---><br>BYE sip:asterisk@192.168.10.1:5070 SIP/2.0<br>Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d><br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0<br>Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0<br>From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d<br>To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f<br>Supported: path<br>Call-ID: 1661965d0c28650d517426333e4a6ae5@192.168.10.1<br>CSeq: 9793 BYE<br>User-Agent: Grandstream GXP2020 1.1.6.16<br>Max-Forwards: 69<br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE<br>Content-Length: 0<br>P-hint: LR|fixcontact,setflag6<br><br><br><-------------><br>--- (14 headers 0 lines) ---<br>Sending to 192.168.10.1 : 5060
(NAT)<br><br><--- Transmitting (NAT) to 192.168.10.1:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0;received=192.168.10.1<br>Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0<br>Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d><br>From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d<br>To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f<br>Call-ID: 1661965d0c28650d517426333e4a6ae5@192.168.10.1<br>CSeq: 9793 BYE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:asterisk@192.168.10.1:5070><br>Content-Length: 0<br><br><br></div><div style="font-family: tahoma,new york,times,serif; font-size: 10pt;"><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight:
bold;">From:</span></b> luzango mfupe <luzango.mfupe@gmail.com><br><b><span style="font-weight: bold;"></span></b></font><br>
<div>Hi Ricky</div>I should have seen how you handle NAT in kamaiilo.conf but you can also edit sip.conf in Asterisk and try to put Nat=yes<div>Rgds,<br><br><br>
</div>
</div></div></div><br>
</body></html>