<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 list, I am giving him returned script from openser trying to
find the problem to this message " 488 acceptable Not here" , I have my openser integrated recently with asterisk and the voicemail recently adds rtpproxy to solve problems of nat and was almost a success, some details to improve, but after adding the rtpproxy when I call to UAC and it does not answer the call on the telephone shows a 488 message to me No Aceptable, it does not jump to the voicemai..<br><br>I have 2 networks cards in my server, and looking at the sip log, I see twice in the sdp the interfaz it public<br><br>any idea as solving this problem?<br><br>regards <br><br>rickygm <br><br><pre>#<br>U +1.675624 192.168.10.1:5060 -> 192.168.10.1:5070<br>INVITE sip:u120@192.168.10.1:5070 SIP/2.0 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 <br>Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 <br>From:
<sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1> <br>Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes> <br>Supported: replaces, timer, path <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 INVITE <br>User-Agent: Grandstream GXV3000 1.1.3.14 <br>Max-Forwards: 69 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK <br>Content-Type: application/sdp <br>Content-Length: 604 <br>P-hint: inbound->inbound <br>P-hint: Route[20]: Rtpproxy <br>P-hint: Route[20]: Rtpproxy <br> <br>v=0 <br>o=119 8000 8001 IN IP4 192.168.10.28 <br>s=SIP Call <br><span style="font-weight: bold;">c=IN IP4 192.168.1.64192.168.1.64 </span><br>t=0 0 <br>m=audio 3504835048 RTP/AVP 18 4 3 2 0 101 <br>a=sendrecv <br>a=rtpmap:18 G729/8000 <br>a=rtpmap:4 G723/8000 <br>a=rtpmap:3 GSM/8000 <br>a=rtpmap:2 G726-32/8000 <br>a=rtpmap:0 PCMU/8000 <br>a=ptime:20 <br>a=rtpmap:101 telephone-event/8000
<br>a=fmtp:101 0-11 <br>m=video 3505035050 RTP/AVP 99 34 <br>a=sendrecv <br>a=rtpmap:99 H264/90000 <br>a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4CXID= <br>a=rtpmap:34 H263/90000 <br>a=fmtp:34 CIF=2 MaxBR=1280 <br>a=framerate:20 <br>a=nortpproxy:yes <br>a=nortpproxy:yes <br><br>#<br>U +0.000052 192.168.10.1:5060 -> 192.168.10.27:5060<br>CANCEL sip:120@192.168.10.27:5060 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>To: <sip:120@192.168.10.1> <br>CSeq: 12216 CANCEL <br>Max-Forwards: 70 <br>User-Agent: OpenSER (1.3.2-notls (i386/linux)) <br>Content-Length: 0 <br> <br><br>#<br>U +0.000411 192.168.10.1:5070 -> 192.168.10.1:5060<br>SIP/2.0 488 Not acceptable here <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1;received=192.168.10.1 <br>Via:
SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1>;tag=as4f1165d5 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 INVITE <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 +0.000120 192.168.10.1:5060 -> 192.168.10.1:5070<br>ACK sip:u120@192.168.10.1:5070 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>To: <sip:120@192.168.10.1>;tag=as4f1165d5 <br>CSeq: 12216 ACK <br>Max-Forwards: 70 <br>User-Agent: OpenSER (1.3.2-notls (i386/linux)) <br>Content-Length: 0 <br> <br><br>#<br>U +0.000144 192.168.10.1:5060 -> 192.168.10.28:5060<br>SIP/2.0 488 Not acceptable here <br>Via:
SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1>;tag=as4f1165d5 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br>P-hint: Onreply-route - fixcontact <br> <br><br>#<br>U +0.000787 192.168.10.27:5060 -> 192.168.10.1:5060<br>SIP/2.0 200 OK <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 CANCEL <br>User-Agent: Grandstream GXV3000 1.1.3.14 <br>Supported: replaces, timer, 100rel, path <br>Content-Length: 0 <br> <br><br>#<br>U +0.000544 192.168.10.27:5060 ->
192.168.10.1:5060<br>SIP/2.0 487 Request Cancelled <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 <br>Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 <br>Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 INVITE <br>User-Agent: Grandstream GXV3000 1.1.3.14 <br>Content-Length: 0 <br> <br><br>#<br>U +0.000067 192.168.10.1:5060 -> 192.168.10.27:5060<br>ACK sip:120@192.168.10.27:5060 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 <br>CSeq: 12216 ACK <br>Max-Forwards: 70 <br>User-Agent: OpenSER (1.3.2-notls (i386/linux))
<br>Content-Length: 0 <br> <br><br>#<br>U +0.001390 192.168.10.28:5060 -> 192.168.10.1:5060<br>ACK sip:120@192.168.10.1 SIP/2.0 <br>Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK286620c52595f4b3 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1>;tag=as4f1165d5 <br>Contact: <sip:119@192.168.10.28:5060> <br>Proxy-Authorization: Digest username="119", realm="192.168.10.1", algorithm=MD5, uri="sip:120@192.168.10.1", nonce="4919edaf0a873106bfd0b18f347709360f00f7eb", response="a5a26c257b491718def2b43488a39853" <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 ACK <br>User-Agent: Grandstream GXV3000 1.1.3.14 <br>Max-Forwards: 70 <br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK <br>Content-Length: 0 <br> <br><br>#<br>U +0.497023 192.168.10.1:5060 -> 192.168.10.28:5060<br>SIP/2.0 488 Not acceptable here <br>Via: SIP/2.0/UDP
192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1>;tag=as4f1165d5 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br>P-hint: Onreply-route - fixcontact <br> <br><br>#<br>U +0.999813 192.168.10.1:5060 -> 192.168.10.28:5060<br>SIP/2.0 488 Not acceptable here <br>Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 <br>From: <sip:119@192.168.10.1>;tag=69372fe200a41663 <br>To: <sip:120@192.168.10.1>;tag=as4f1165d5 <br>Call-ID: 7c55e4667d473f76@192.168.10.28 <br>CSeq: 12216 INVITE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Supported: replaces <br>Content-Length: 0 <br>P-hint: Onreply-route
- fixcontact <br> <br><br>exit<br>67 received, 0 dropped<br></pre><br></div></div><br>
</body></html>