<br>Hi Ricky,<div>Where is your Kamailio config?? is this your full ngrep capture??</div><div>Rgds,</div><div>Luzango.<br><div class="gmail_quote"><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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Message: 2<br>
Date: Tue, 28 Oct 2008 23:10:12 -0700 (PDT)<br>
From: Ricky Gutierrez <<a href="mailto:xserverlinux@yahoo.com">xserverlinux@yahoo.com</a>><br>
Subject: [Kamailio-Users] I don't have asterisk audio to openser -<br>
mediaproxy<br>
To: <a href="mailto:users@lists.kamailio.org">users@lists.kamailio.org</a><br>
Message-ID: <<a href="mailto:654712.75990.qm@web59903.mail.ac4.yahoo.com">654712.75990.qm@web59903.mail.ac4.yahoo.com</a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
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>
<br>
From: "Ventas" <<a href="mailto:sip%3A112@192.168.10.1">sip:112@192.168.10.1</a>>;tag=69451218021829df<br>
To: <<a href="mailto:sip%3A2685249@192.168.10.1">sip:2685249@192.168.10.1</a>>;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2<br>
Contact: <sip:112@192.168.10.30:5060;transport=udp><br>
Supported: path<br>
Call-ID: <a href="mailto:fb5f5dac83056f72@192.168.10.30">fb5f5dac83056f72@192.168.10.30</a><br>
CSeq: 7492 ACK<br>
User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a><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 <a href="http://192.168.10.30:5060" target="_blank">192.168.10.30:5060</a> -> <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a><br>
INVITE <a href="mailto:sip%3A2685249@192.168.10.1">sip:2685249@192.168.10.1</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03<br>
From: "Ventas" <<a href="mailto:sip%3A112@192.168.10.1">sip:112@192.168.10.1</a>>;tag=69451218021829df<br>
To: <<a href="mailto:sip%3A2685249@192.168.10.1">sip:2685249@192.168.10.1</a>><br>
Contact: <sip:112@192.168.10.30:5060;transport=udp><br>
Supported: replaces, timer, path<br>
Proxy-Authorization: Digest username="112", realm="<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>", algorithm=MD5, uri="<a href="mailto:sip%3A2685249@192.168.10.1">sip:2685249@192.168.10.1</a>", nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", response="c40a9387fdf5de29115c1edadc7f79db"<br>
Call-ID: <a href="mailto:fb5f5dac83056f72@192.168.10.30">fb5f5dac83056f72@192.168.10.30</a><br>
CSeq: 7493 INVITE<br>
User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a><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 <a href="http://192.168.10.30" target="_blank">192.168.10.30</a><br>
s=SIP Call<br>
c=IN IP4 <a href="http://192.168.10.30" target="_blank">192.168.10.30</a><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 <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a> -> <a href="http://192.168.10.30:5060" target="_blank">192.168.10.30:5060</a><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" <<a href="mailto:sip%3A112@192.168.10.1">sip:112@192.168.10.1</a>>;tag=69451218021829df<br>
To: <<a href="mailto:sip%3A2685249@192.168.10.1">sip:2685249@192.168.10.1</a>><br>
Call-ID: <a href="mailto:fb5f5dac83056f72@192.168.10.30">fb5f5dac83056f72@192.168.10.30</a><br>
CSeq: 7493 INVITE<br>
Server: OpenSER (1.3.2-notls (i386/linux))<br>
Content-Length: 0<br>
<br>
<br>
#<br>
U +0.000115 <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a> -> <a href="http://192.168.10.1:5070" target="_blank">192.168.10.1:5070</a><br>
INVITE <a href="http://sip:2685249@192.168.10.1:5070" target="_blank">sip:2685249@192.168.10.1:5070</a> SIP/2.0<br>
Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=69451218021829df;nat=yes><br>
Via: SIP/2.0/UDP <a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;branch=z9hG4bK400f.b93e5c35.0<br>
Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03<br>
From: "Ventas" <<a href="mailto:sip%3A112@192.168.10.1">sip:112@192.168.10.1</a>>;tag=69451218021829df<br>
To: <<a href="mailto:sip%3A2685249@192.168.10.1">sip:2685249@192.168.10.1</a>><br>
Contact: <sip:112@192.168.10.30:5060;transport=udp><br>
Supported: replaces, timer, path<br>
Call-ID: <a href="mailto:fb5f5dac83056f72@192.168.10.30">fb5f5dac83056f72@192.168.10.30</a><br>
CSeq: 7493 INVITE<br>
User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a><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 <a href="http://192.168.10.30" target="_blank">192.168.10.30</a><br>
s=SIP Call<br>
c=IN IP4 <a href="http://192.168.1.64" target="_blank">192.168.1.64</a><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 <a href="http://192.168.10.1:5070" target="_blank">192.168.10.1:5070</a> -> <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a><br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP <a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;branch=z9hG4bK400f.b93e5c35.0;received=<a href="http://192.168.10.1" target="_blank">192.168.10.1</a><br>
Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03<br>
Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=69451218021829df;nat=yes><br>
From: "Ventas" <<a href="mailto:sip%3A112@192.168.10.1">sip:112@192.168.10.1</a>>;tag=69451218021829df<br>
To: <<a href="mailto:sip%3A2685249@192.168.10.1">sip:2685249@192.168.10.1</a>><br>
Call-ID: <a href="mailto:fb5f5dac83056f72@192.168.10.30">fb5f5dac83056f72@192.168.10.30</a><br>
CSeq: 7493 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>
Contact: <<a href="http://sip:2685249@192.168.10.1:5070" target="_blank">sip:2685249@192.168.10.1:5070</a>><br>
Content-Length: 0<br>
<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 <-><br>
eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan (<a href="http://192.168.11.1" target="_blank">192.168.11.1</a>) <-> 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>
<br>
<br>
<br></blockquote></div><br clear="all"><br>-- <br>Luzango Mfupe<br>TUUNE MOBILE<br>Tel:0128440528/0123825710<br>Tshwane-RSA<br><br>"...Ships are safe in harbor, but they were never meant to stay there......."<br>
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