<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>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 192.168.10.1:5070 -> 192.168.10.1:5060<br>INVITE sip:113@192.168.10.1 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1> <br>Contact: <sip:asterisk@192.168.10.1:5070> <br>Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 <br>CSeq: 102 INVITE <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Type: application/sdp <br>Content-Length: 238 <br> <br>v=0 <br>o=root 9850 9850 IN IP4 192.168.10.1 <br>s=session <br>c=IN IP4 192.168.10.1 <br>t=0 0 <br>m=audio 14750 RTP/AVP 0
101 <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-16 <br>a=silenceSupp:off - - - - <br>a=ptime:20 <br>a=sendrecv <br><br>#<br>U +0.004203 192.168.10.1:5060 -> 192.168.10.1:5070<br>SIP/2.0 100 Giving a try <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1> <br>Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 <br>CSeq: 102 INVITE <br>Server: OpenSER (1.3.2-notls (i386/linux)) <br>Content-Length: 0 <br> <br><br>#<br>U +0.000275 192.168.10.1:5060 -> 192.168.10.30:5062<br>INVITE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>From: "asterisk"
<sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1> <br>Contact: <sip:asterisk@192.168.10.1:5070> <br>Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 <br>CSeq: 102 INVITE <br>User-Agent: Asterisk PBX <br>Max-Forwards: 69 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Type: application/sdp <br>Content-Length: 238 <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=root 9850 9850 IN IP4 192.168.10.1 <br>s=session <br>c=IN IP4 192.168.1.64 <br>t=0 0 <br>m=audio 35058 RTP/AVP 0 101 <br>a=rtpmap:0 PCMU/8000 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-16 <br>a=silenceSupp:off - - - - <br>a=ptime:20 <br>a=sendrecv <br><br>#<br>U +0.026845 192.168.10.30:5062 -> 192.168.10.1:5060<br>SIP/2.0 100 Trying <br>Via:
SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1> <br>Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 <br>CSeq: 102 INVITE <br>User-Agent: Grandstream GXP2020 1.1.6.16 <br>Content-Length: 0 <br> <br><br>#<br>U +0.009886 192.168.10.30:5062 -> 192.168.10.1:5060<br>SIP/2.0 180 Ringing <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 <br>Call-ID: 373456b8787e65d9764158381c9da273@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> <br><br>#<br>U +0.000169 192.168.10.1:5060 -> 192.168.10.1:5070<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:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 <br>Call-ID: 373456b8787e65d9764158381c9da273@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>U +0.000140 190.184.35.4:5060 -> 192.168.1.64:5064<br>OPTIONS
sip:130@192.168.1.64:5064 SIP/2.0 <br>Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport <br>From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 <br>To: <sip:130@192.168.1.64:5064> <br>Contact: <sip:asterisk@190.184.35.4> <br>Call-ID: 16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 <br>CSeq: 102 OPTIONS <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br> <br><br>#<br>U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064<br>OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 <br>Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport <br>From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 <br>To: <sip:130@192.168.1.64:5064> <br>Contact: <sip:asterisk@190.184.35.4> <br>Call-ID: 16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 <br>CSeq: 102
OPTIONS <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Date: Wed, 29 Oct 2008 16:26:18 GMT <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br> <br><br>#<br>U +0.008208 192.168.10.19:5064 -> 190.184.35.4:5060<br>SIP/2.0 200 OK <br>To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 <br>From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 <br>Call-ID: 16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 <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 192.168.1.64:5064 -> 190.184.35.4:5060<br>SIP/2.0 200 OK <br>To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 <br>From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 <br>Call-ID:
16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 <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 192.168.10.19:5064 -> 190.184.35.4:5060<br>NOTIFY sip:190.184.35.4 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 <br>From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 <br>To: <sip:190.184.35.4> <br>Call-ID: ce8ea9b2-baee4a99@192.168.10.19 <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 192.168.1.64:5064 -> 190.184.35.4:5060<br>NOTIFY sip:190.184.35.4 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 <br>From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 <br>To:
<sip:190.184.35.4> <br>Call-ID: ce8ea9b2-baee4a99@192.168.10.19 <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 190.184.35.4:5060 -> 192.168.1.64:5064<br>SIP/2.0 489 Bad event <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 <br>From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 <br>To: <sip:190.184.35.4>;tag=as2002c003 <br>Call-ID: ce8ea9b2-baee4a99@192.168.10.19 <br>CSeq: 12 NOTIFY <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br> <br><br>#<br>U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064<br>SIP/2.0 489 Bad event <br>Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 <br>From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 <br>To:
<sip:190.184.35.4>;tag=as2002c003 <br>Call-ID: ce8ea9b2-baee4a99@192.168.10.19 <br>CSeq: 12 NOTIFY <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br> <br><br>#<br>U +5.895003 192.168.10.30:5062 -> 192.168.10.1:5060<br>SIP/2.0 200 OK <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 <br>Call-ID: 373456b8787e65d9764158381c9da273@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> <br>v=0 <br>o=113 8000 8000 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 0 101 <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>U +0.001400 192.168.10.1:5060 -> 192.168.10.1:5070<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:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 <br>Call-ID: 373456b8787e65d9764158381c9da273@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 35058 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>U +0.000557 192.168.10.1:5070 -> 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=z9hG4bK3f155ae6;rport <br>Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 <br>Contact: <sip:asterisk@192.168.10.1:5070> <br>Call-ID:
373456b8787e65d9764158381c9da273@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>U +0.000215 192.168.10.1:5060 -> 192.168.10.30:5062<br>ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.2 <br>Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070 <br>From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 <br>To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 <br>Contact: <sip:asterisk@192.168.10.1:5070> <br>Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 <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 192.168.10.28:5060 -> 192.168.10.1:5060<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 <luzango.mfupe@gmail.com><br><b><span style="font-weight: bold;">To:</span></b> users@lists.kamailio.org<br><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><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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