Hi Ricky 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=yesRgds,

On Wed, Oct 29, 2008 at 6:39 PM, Ricky Gutierrez xserverlinux@yahoo.comwrote:

Hi luzano thank you for your help and time, this it is my full ngrep. I have asterisk, mediaproxy and openser together in the same pc

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

that it leaves you want to see of the openser.cfg, or everything?

regards ..

interface: any filter: (ip) and ( port 5060 ) # U +0.063740 192.168.10.1:5070 -> 192.168.10.1:5060 INVITE sip:113@192.168.10.1 sip%3A113@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1> Contact: sip:asterisk@192.168.10.1:5070 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238

v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 14750 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

# U +0.004203 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0

# U +0.000275 192.168.10.1:5060 -> 192.168.10.30:5062 INVITE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1> Contact: sip:asterisk@192.168.10.1:5070 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy

v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

# U +0.026845 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Content-Length: 0

# U +0.009886 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: sip:113@192.168.10.30:5062;transport=udp Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0

# U +0.000169 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: sip:113@192.168.10.30:5062;transport=udp Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact

# U +0.000140 190.184.35.4:5060 -> 192.168.1.64:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4 sip%3Aasterisk@190.184.35.4>;tag=as35db6300 To: sip:130@192.168.1.64:5064 Contact: <sip:asterisk@190.184.35.4 sip%3Aasterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

# U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4 sip%3Aasterisk@190.184.35.4>;tag=as35db6300 To: sip:130@192.168.1.64:5064 Contact: <sip:asterisk@190.184.35.4 sip%3Aasterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

# U +0.008208 192.168.10.19:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: sip:130@192.168.1.64:5064;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4 sip%3Aasterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces

# U +0.000014 192.168.1.64:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: sip:130@192.168.1.64:5064;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4 sip%3Aasterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3@190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces

# U +0.956449 192.168.10.19:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4 sip%3A130@190.184.35.4>;tag=7c3557f6145bd125o1 To: sip:190.184.35.4 Call-ID: ce8ea9b2-baee4a99@192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0

# U +0.000027 192.168.1.64:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4 sip%3A130@190.184.35.4>;tag=7c3557f6145bd125o1 To: sip:190.184.35.4 Call-ID: ce8ea9b2-baee4a99@192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0

# U +0.156145 190.184.35.4:5060 -> 192.168.1.64:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4 sip%3A130@190.184.35.4>;tag=7c3557f6145bd125o1 To: sip:190.184.35.4;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99@192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

# U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4 sip%3A130@190.184.35.4>;tag=7c3557f6145bd125o1 To: sip:190.184.35.4;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99@192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

# U +5.895003 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: sip:113@192.168.10.30:5062;transport=udp Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE

Content-Type: application/sdp Supported: replaces, timer Content-Length: 212

v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11

# U +0.001400 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: sip:113@192.168.10.30:5062;transport=udp Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy

v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11

# U +0.000557 192.168.10.1:5070 -> 192.168.10.1:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport Route: sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1>;tag=21c220c2e075d838 Contact: sip:asterisk@192.168.10.1:5070 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0

# U +0.000215 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.2 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070 From: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as30de9085 To: <sip:113@192.168.10.1 sip%3A113@192.168.10.1>;tag=21c220c2e075d838 Contact: sip:asterisk@192.168.10.1:5070 Call-ID: 373456b8787e65d9764158381c9da273@192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 P-hint: LR|fixcontact,setflag6

# U +0.886545 192.168.10.28:5060 -> 192.168.10.1:5060

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*From:* luzango mfupe luzango.mfupe@gmail.com *To:* users@lists.kamailio.org *Sent:* Wednesday, October 29, 2008 5:12:29 AM *Subject:* Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

Hi Ricky,Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango.