<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><div class="gmail_quote">On Wed, Oct 29, 2008 at 6:39 PM, Ricky Gutierrez <span dir="ltr"><<a href="mailto:xserverlinux@yahoo.com">xserverlinux@yahoo.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div style="font-family:tahoma,new york,times,serif;font-size:10pt"><div>Hi luzano thank you for your help and time, this it is my full ngrep.<br>
I have asterisk, mediaproxy and openser together in the same pc<br></div><div style="font-family:tahoma,new york,times,serif;font-size:10pt"><br>I have the doubt that when a incoming call from the PSTN by asterisk,
sends it to an extension of openser, to which I do not request
authentication to him within invites, I believe that there it is where
I have problems with mediaproxy<br><br>that it leaves you want to see of the openser.cfg, or everything?<br><br>regards ..<br><pre>interface: any<br>filter: (ip) and ( port 5060 )<br>#<br>U +0.063740 <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>
INVITE <a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a> SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport <br>From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>
To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>> <br>Contact: <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>> <br>Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>
CSeq: 102 INVITE <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <div class="Ih2E3d"><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>
</div>Content-Type: application/sdp <br>Content-Length: 238 <br> <br>v=0 <br>o=root 9850 9850 IN IP4 <a href="http://192.168.10.1" target="_blank">192.168.10.1</a> <br>s=session <br>c=IN IP4 <a href="http://192.168.10.1" target="_blank">192.168.10.1</a> <br>
t=0 0 <br>m=audio 14750 RTP/AVP 0
101 <div class="Ih2E3d"><br>a=rtpmap:0 PCMU/8000 <br></div><div class="Ih2E3d">a=rtpmap:101 telephone-event/8000 <br></div>a=fmtp:101 0-16 <br>a=silenceSupp:off - - - - <br>a=ptime:20 <br>a=sendrecv <br><br>#<br>U +0.004203 <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><div class="Ih2E3d">
<br>SIP/2.0 100 Giving a try <br></div>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>
To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>> <br>Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>
CSeq: 102 INVITE <div class="Ih2E3d"><br>Server: OpenSER (1.3.2-notls (i386/linux)) <br>Content-Length: 0 <br> <br><br>#<br></div>U +0.000275 <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a> -> <a href="http://192.168.10.30:5062" target="_blank">192.168.10.30:5062</a><br>
INVITE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 <br>Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=as30de9085;nat=yes> <br>Via: SIP/2.0/UDP <a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;branch=z9hG4bK7ce3.dd78e5e3.0 <br>
Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>From: "asterisk"
<<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>> <br>
Contact: <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>> <br>Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>
CSeq: 102 INVITE <br>User-Agent: Asterisk PBX <br>Max-Forwards: 69 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <div class="Ih2E3d"><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>
</div>Content-Type: application/sdp <br>Content-Length: 238 <div class="Ih2E3d"><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></div>
o=root 9850 9850 IN IP4 <a href="http://192.168.10.1" target="_blank">192.168.10.1</a> <br>s=session <div class="Ih2E3d"><br>c=IN IP4 <a href="http://192.168.1.64" target="_blank">192.168.1.64</a> <br>t=0 0 <br></div>m=audio 35058 RTP/AVP 0 101 <div class="Ih2E3d">
<br>a=rtpmap:0 PCMU/8000 <br></div><div class="Ih2E3d">a=rtpmap:101 telephone-event/8000 <br></div>a=fmtp:101 0-16 <br>a=silenceSupp:off - - - - <br>a=ptime:20 <br>a=sendrecv <br><br>#<br>U +0.026845 <a href="http://192.168.10.30:5062" target="_blank">192.168.10.30:5062</a> -> <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a><div class="Ih2E3d">
<br>SIP/2.0 100 Trying <br></div>Via:
SIP/2.0/UDP <a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;branch=z9hG4bK7ce3.dd78e5e3.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>
To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>> <br>Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>
CSeq: 102 INVITE <div class="Ih2E3d"><br>User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a> <br></div>Content-Length: 0 <br> <br><br>#<br>U +0.009886 <a href="http://192.168.10.30:5062" target="_blank">192.168.10.30:5062</a> -> <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a><br>
SIP/2.0 180 Ringing <br>Via: SIP/2.0/UDP <a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;branch=z9hG4bK7ce3.dd78e5e3.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=as30de9085;nat=yes> <br>
From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>>;tag=21c220c2e075d838 <br>
Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>CSeq: 102 INVITE <div class="Ih2E3d"><br>User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a> <br>
</div>Contact:
<sip:113@192.168.10.30:5062;transport=udp> <div class="Ih2E3d"><br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Length: 0 <br> <br><br>#<br></div>U +0.000169 <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>
SIP/2.0 180 Ringing <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=as30de9085;nat=yes> <br>From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>
To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>>;tag=21c220c2e075d838 <br>Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>
CSeq: 102 INVITE <div class="Ih2E3d"><br>User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a> <br></div>Contact: <sip:113@192.168.10.30:5062;transport=udp> <div class="Ih2E3d"><br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>
Content-Length: 0 <br></div>P-hint: Onreply-route - fixcontact <br> <br><br>#<br>U +0.000140 <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a> -> <a href="http://192.168.1.64:5064" target="_blank">192.168.1.64:5064</a><br>
OPTIONS
<a href="http://sip:130@192.168.1.64:5064" target="_blank">sip:130@192.168.1.64:5064</a> SIP/2.0 <br>Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport <br>From: "asterisk" <<a href="mailto:sip%3Aasterisk@190.184.35.4" target="_blank">sip:asterisk@190.184.35.4</a>>;tag=as35db6300 <br>
To: <<a href="http://sip:130@192.168.1.64:5064" target="_blank">sip:130@192.168.1.64:5064</a>> <br>Contact: <<a href="mailto:sip%3Aasterisk@190.184.35.4" target="_blank">sip:asterisk@190.184.35.4</a>> <br>Call-ID: <a href="mailto:16e287fc2b24afb97eb1759952eff3d3@190.184.35.4" target="_blank">16e287fc2b24afb97eb1759952eff3d3@190.184.35.4</a> <br>
CSeq: 102 OPTIONS <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <div class="Ih2E3d"><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>
</div>Content-Length: 0 <br> <br><br>#<br>U +0.000017 <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a> -> <a href="http://192.168.10.19:5064" target="_blank">192.168.10.19:5064</a><br>OPTIONS <a href="http://sip:130@192.168.1.64:5064" target="_blank">sip:130@192.168.1.64:5064</a> SIP/2.0 <br>
Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport <br>From: "asterisk" <<a href="mailto:sip%3Aasterisk@190.184.35.4" target="_blank">sip:asterisk@190.184.35.4</a>>;tag=as35db6300 <br>To: <<a href="http://sip:130@192.168.1.64:5064" target="_blank">sip:130@192.168.1.64:5064</a>> <br>
Contact: <<a href="mailto:sip%3Aasterisk@190.184.35.4" target="_blank">sip:asterisk@190.184.35.4</a>> <br>Call-ID: <a href="mailto:16e287fc2b24afb97eb1759952eff3d3@190.184.35.4" target="_blank">16e287fc2b24afb97eb1759952eff3d3@190.184.35.4</a> <br>
CSeq: 102
OPTIONS <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <div class="Ih2E3d"><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br></div>
Content-Length: 0 <br> <br><br>#<br>U +0.008208 <a href="http://192.168.10.19:5064" target="_blank">192.168.10.19:5064</a> -> <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a><br>SIP/2.0 200 OK <br>
To: <<a href="http://sip:130@192.168.1.64:5064" target="_blank">sip:130@192.168.1.64:5064</a>>;tag=9611a90af8ab9321i1 <br>From: "asterisk" <<a href="mailto:sip%3Aasterisk@190.184.35.4" target="_blank">sip:asterisk@190.184.35.4</a>>;tag=as35db6300 <br>
Call-ID: <a href="mailto:16e287fc2b24afb97eb1759952eff3d3@190.184.35.4" target="_blank">16e287fc2b24afb97eb1759952eff3d3@190.184.35.4</a> <br>CSeq: 102 OPTIONS <br>Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 <br>
Server: Linksys/SPA942-5.2.8 <br>Content-Length: 0 <br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER <br>Supported: replaces <br> <br><br>#<br>U +0.000014 <a href="http://192.168.1.64:5064" target="_blank">192.168.1.64:5064</a> -> <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a><br>
SIP/2.0 200 OK <br>To: <<a href="http://sip:130@192.168.1.64:5064" target="_blank">sip:130@192.168.1.64:5064</a>>;tag=9611a90af8ab9321i1 <br>From: "asterisk" <<a href="mailto:sip%3Aasterisk@190.184.35.4" target="_blank">sip:asterisk@190.184.35.4</a>>;tag=as35db6300 <br>
Call-ID:
<a href="mailto:16e287fc2b24afb97eb1759952eff3d3@190.184.35.4" target="_blank">16e287fc2b24afb97eb1759952eff3d3@190.184.35.4</a> <br>CSeq: 102 OPTIONS <br>Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 <br>Server: Linksys/SPA942-5.2.8 <br>
Content-Length: 0 <br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER <br>Supported: replaces <br> <br><br>#<br>U +0.956449 <a href="http://192.168.10.19:5064" target="_blank">192.168.10.19:5064</a> -> <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a><br>
NOTIFY sip:<a href="http://190.184.35.4" target="_blank">190.184.35.4</a> SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 <br>From: <<a href="mailto:sip%3A130@190.184.35.4" target="_blank">sip:130@190.184.35.4</a>>;tag=7c3557f6145bd125o1 <br>
To: <sip:<a href="http://190.184.35.4" target="_blank">190.184.35.4</a>> <br>Call-ID: <a href="mailto:ce8ea9b2-baee4a99@192.168.10.19" target="_blank">ce8ea9b2-baee4a99@192.168.10.19</a> <br>CSeq: 12 NOTIFY <br>Max-Forwards: 70 <br>
Event: keep-alive <br>User-Agent: Linksys/SPA942-5.2.8 <br>Content-Length: 0 <br> <br><br>#<br>U +0.000027 <a href="http://192.168.1.64:5064" target="_blank">192.168.1.64:5064</a> -> <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a><br>
NOTIFY sip:<a href="http://190.184.35.4" target="_blank">190.184.35.4</a> SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 <br>From: <<a href="mailto:sip%3A130@190.184.35.4" target="_blank">sip:130@190.184.35.4</a>>;tag=7c3557f6145bd125o1 <br>
To:
<sip:<a href="http://190.184.35.4" target="_blank">190.184.35.4</a>> <br>Call-ID: <a href="mailto:ce8ea9b2-baee4a99@192.168.10.19" target="_blank">ce8ea9b2-baee4a99@192.168.10.19</a> <br>CSeq: 12 NOTIFY <br>Max-Forwards: 70 <br>
Event: keep-alive <br>User-Agent: Linksys/SPA942-5.2.8 <br>Content-Length: 0 <br> <br><br>#<br>U +0.156145 <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a> -> <a href="http://192.168.1.64:5064" target="_blank">192.168.1.64:5064</a><br>
SIP/2.0 489 Bad event <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=<a href="http://192.168.1.64" target="_blank">192.168.1.64</a> <br>From: <<a href="mailto:sip%3A130@190.184.35.4" target="_blank">sip:130@190.184.35.4</a>>;tag=7c3557f6145bd125o1 <br>
To: <sip:<a href="http://190.184.35.4" target="_blank">190.184.35.4</a>>;tag=as2002c003 <br>Call-ID: <a href="mailto:ce8ea9b2-baee4a99@192.168.10.19" target="_blank">ce8ea9b2-baee4a99@192.168.10.19</a> <br>CSeq: 12 NOTIFY <div class="Ih2E3d">
<br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br></div>Content-Length: 0 <br> <br><br>#<br>U +0.000017 <a href="http://190.184.35.4:5060" target="_blank">190.184.35.4:5060</a> -> <a href="http://192.168.10.19:5064" target="_blank">192.168.10.19:5064</a><br>
SIP/2.0 489 Bad event <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=<a href="http://192.168.1.64" target="_blank">192.168.1.64</a> <br>From: <<a href="mailto:sip%3A130@190.184.35.4" target="_blank">sip:130@190.184.35.4</a>>;tag=7c3557f6145bd125o1 <br>
To:
<sip:<a href="http://190.184.35.4" target="_blank">190.184.35.4</a>>;tag=as2002c003 <br>Call-ID: <a href="mailto:ce8ea9b2-baee4a99@192.168.10.19" target="_blank">ce8ea9b2-baee4a99@192.168.10.19</a> <br>CSeq: 12 NOTIFY <div class="Ih2E3d">
<br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br></div>Content-Length: 0 <br> <br><br>#<br>U +5.895003 <a href="http://192.168.10.30:5062" target="_blank">192.168.10.30:5062</a> -> <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a><br>
SIP/2.0 200 OK <br>Via: SIP/2.0/UDP <a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;branch=z9hG4bK7ce3.dd78e5e3.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=as30de9085;nat=yes> <br>
From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>>;tag=21c220c2e075d838 <br>
Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>CSeq: 102 INVITE <div class="Ih2E3d"><br>User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a> <br>
</div>Contact: <sip:113@192.168.10.30:5062;transport=udp> <div class="Ih2E3d"><br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
<br>Content-Type: application/sdp <br></div>Supported: replaces, timer <br>Content-Length: 212 <br> <br>v=0 <br>o=113 8000 8000 IN IP4 <a href="http://192.168.10.30" target="_blank">192.168.10.30</a> <div class="Ih2E3d">
<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></div>m=audio 5004 RTP/AVP 0 101 <div class="Ih2E3d"><br>a=sendrecv <br>a=rtpmap:0 PCMU/8000 <br></div><div class="Ih2E3d">
a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11 <br><br>#<br></div>U +0.001400 <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>
SIP/2.0 200 OK <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=as30de9085;nat=yes> <br>From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>
To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>>;tag=21c220c2e075d838 <br>Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>
CSeq: 102 INVITE <div class="Ih2E3d"><br>User-Agent: Grandstream GXP2020 <a href="http://1.1.6.16" target="_blank">1.1.6.16</a> <br></div>Contact: <sip:113@192.168.10.30:5062;transport=udp> <div class="Ih2E3d"><br>Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE <br>Content-Type: application/sdp <br></div>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 <a href="http://192.168.10.30" target="_blank">192.168.10.30</a> <div class="Ih2E3d"><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>
</div>m=audio 35058 RTP/AVP 0 101 <div class="Ih2E3d"><br>a=sendrecv <br>a=rtpmap:0 PCMU/8000 <br></div><div class="Ih2E3d">a=ptime:20 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-11 <br><br>#<br></div>U +0.000557 <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>
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=z9hG4bK3f155ae6;rport <br>Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=as30de9085;nat=yes> <br>
From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>>;tag=21c220c2e075d838 <br>
Contact: <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>> <br>Call-ID:
<a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>CSeq: 102 ACK <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Content-Length: 0 <br>
<br><br>#<br>U +0.000215 <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a> -> <a href="http://192.168.10.30:5062" target="_blank">192.168.10.30:5062</a><br>ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 <br>
Record-Route: <sip:<a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;lr=on;ftag=as30de9085;nat=yes> <br>Via: SIP/2.0/UDP <a href="http://192.168.10.1" target="_blank">192.168.10.1</a>;branch=z9hG4bK7ce3.dd78e5e3.2 <br>
Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070 <br>From: "asterisk" <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>>;tag=as30de9085 <br>
To: <<a href="mailto:sip%3A113@192.168.10.1" target="_blank">sip:113@192.168.10.1</a>>;tag=21c220c2e075d838 <br>Contact: <<a href="http://sip:asterisk@192.168.10.1:5070" target="_blank">sip:asterisk@192.168.10.1:5070</a>> <br>
Call-ID: <a href="mailto:373456b8787e65d9764158381c9da273@192.168.10.1" target="_blank">373456b8787e65d9764158381c9da273@192.168.10.1</a> <br>CSeq: 102 ACK <br>User-Agent: Asterisk PBX <br>Max-Forwards: 69 <br>Content-Length: 0 <br>
P-hint: LR|fixcontact,setflag6 <br> <br><br>#<br>U +0.886545 <a href="http://192.168.10.28:5060" target="_blank">192.168.10.28:5060</a> -> <a href="http://192.168.10.1:5060" target="_blank">192.168.10.1:5060</a><br></pre>
<br><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 <<a href="mailto:luzango.mfupe@gmail.com" target="_blank">luzango.mfupe@gmail.com</a>><div class="Ih2E3d">
<br><b><span style="font-weight:bold">To:</span></b> <a href="mailto:users@lists.kamailio.org" target="_blank">users@lists.kamailio.org</a><br></div><b><span style="font-weight:bold">Sent:</span></b> Wednesday, October 29, 2008 5:12:29 AM<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy<br></font><div class="Ih2E3d"><br>
<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></div><br>
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</div></blockquote></div><br><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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