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<div style="font-family: tahoma,new york,times,serif; font-size: 10pt;"><div>Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, I have gotten myself an ip it public with my supplier, I have two network cards in the pc that I am using for openser and mediaproxy together with asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the pstn through asterisk I don't have audio, if I call to the pstn they listen to me well .<br><br><pre>From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df <br>To: <sip:2685249@192.168.10.1>;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 <br>Contact: <sip:112@192.168.10.30:5060;transport=udp> <br>Supported: path <br>Call-ID: fb5f5dac83056f72@192.168.10.30 <br>CSeq: 7492 ACK <br>User-Agent: Grandstream GXP2020 1.1.6.16 <br>Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Length: 0 <br> <br><br>#<br>U
+0.022110 192.168.10.30:5060 -> 192.168.10.1:5060<br>INVITE sip:2685249@192.168.10.1 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 <br>From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df <br>To: <sip:2685249@192.168.10.1> <br>Contact: <sip:112@192.168.10.30:5060;transport=udp> <br>Supported: replaces, timer, path <br>Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:2685249@192.168.10.1", nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", response="c40a9387fdf5de29115c1edadc7f79db" <br>Call-ID: fb5f5dac83056f72@192.168.10.30 <br>CSeq: 7493 INVITE <br>User-Agent: Grandstream GXP2020 1.1.6.16 <br>Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Type: application/sdp <br>Content-Length: 358 <br> <br>v=0 <br>o=112 8000 8001 IN IP4 192.168.10.30 <br>s=SIP Call <br>c=IN IP4 192.168.10.30
<br>t=0 0 <br>m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 <br>a=sendrecv <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:3 GSM/8000 <br>a=rtpmap:97 iLBC/8000 <br>a=fmtp:97 mode=20 <br>a=rtpmap:2 G726-32/8000 <br>a=rtpmap:9 G722/16000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11 <br><br>#<br>U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060<br>SIP/2.0 100 Giving a try <br>Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03;rport=5060 <br>From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df <br>To: <sip:2685249@192.168.10.1> <br>Call-ID: fb5f5dac83056f72@192.168.10.30 <br>CSeq: 7493 INVITE <br>Server: OpenSER (1.3.2-notls (i386/linux)) <br>Content-Length: 0 <br> <br><br>#<br>U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070<br>INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> <br>Via: SIP/2.0/UDP
192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 <br>Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 <br>From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df <br>To: <sip:2685249@192.168.10.1> <br>Contact: <sip:112@192.168.10.30:5060;transport=udp> <br>Supported: replaces, timer, path <br>Call-ID: fb5f5dac83056f72@192.168.10.30 <br>CSeq: 7493 INVITE <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-Type: application/sdp <br>Content-Length: 358 <br>P-hint: route(3)|setflag7,forcerport,fix_contact <br>P-hint: inbound->inbound <br>P-hint: Route[6]: mediaproxy <br> <br>v=0 <br>o=112 8000 8001 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 35040 RTP/AVP 0 18 3 97 2 9 101 <br>a=sendrecv <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:3
GSM/8000 <br>a=rtpmap:97 iLBC/8000 <br>a=fmtp:97 mode=20 <br>a=rtpmap:2 G726-32/8000 <br>a=rtpmap:9 G722/16000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11 <br><br>#<br>U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060<br>SIP/2.0 100 Trying <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received=192.168.10.1 <br>Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> <br>From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df <br>To: <sip:2685249@192.168.10.1> <br>Call-ID: fb5f5dac83056f72@192.168.10.30 <br>CSeq: 7493 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Contact: <sip:2685249@192.168.10.1:5070> <br>Content-Length: 0 <br></pre><br><br>I don't have a lot of experience with mediaproxy, and I
have some doubts that such you see they can help me to clarify, inside the file mediaproxy.ini some options appear which I have configured them but I am not sure if it is the best way.<br><br>my scenario is the following one:<br><br><-> UAC<-> NAT <-> ADSL <-> Internet <->
eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan (192.168.11.1) <-> UAC <br><br><br>[MediaProxy]<br><br>start = yes<br>socket = /var/run/mediaproxy.sock<br>group = openser<br>listen = None<br>allow = None<br>proxyIP = x.x.x.x (public ip)<br>;portRange = 60000:65000<br>portRange = 35000:65000<br>TOS = 0xb8<br>idleTimeout = 60<br>holdTimeout = 3600<br>forceClose = 0<br><br>[Accounting]<br>; one of none, radius or database<br>accounting = none<br><br>[Database]<br>user = dbuser<br>password = dbpass<br>host = dbhost<br>database = radius<br>table = radacct<br><br>[Radius]<br>secret = secret<br>server = localhost<br>authport = 1812<br>acctport = 1813<br>dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary<br>retries = 2<br>timeout = 3<br><br><br><br><br>this couple of you line inside the openser, I don't still understand them according to the guide of ser getting
started they are for asymmetric clients, but I don't find an example<br><br>modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients")<br>modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients")<br><br>somebody that can give me a good help...<br><br>regards <br><br>rickygm<br></div></div></div><br>
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