This is bug. I solve this problem by upgrading to 1.5.0 ----------------- Vladimir Romanov Yota Lab | http://www.yota.ru CTO +7 (960) 239-0853
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of troxlinux Sent: Saturday, April 11, 2009 9:17 AM To: [email protected] Subject: [OpenSIPS-Users] 488 Not acceptable here Hi list , I am making some tests with a server opensips and adds him the rtpproxy for the nat, the problem is that when adding the nat and to call to an extension that don't answer it doesn't jump me to the asterisk voicemail and it shows me an error 488 I explain that in the same server opensips I have installed asterisk , in the asterisk cli when the call is not answered he throws me this error: WARNING[3178]: chan_sip.c:5201 process_sdp: Unable to lookup host in c= line, 'IN IP4 192.168.10.3192.168.10.3' the sdp writes it twice , as I can avoid this? ## log sip## # U +0.019539 192.168.10.30:5064 -> 192.168.10.3:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.3:5060;branch=0 From: sip:[email protected];tag=cd0baa81 To: sip:192.168.10.30:5064;tag=a8c59398c8984470 Call-ID: [email protected] CSeq: 1 OPTIONS User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:[email protected]:5064;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 # U +2.000872 192.168.10.3:5060 -> 192.168.10.3:5070 INVITE sip:[email protected]:5070 SIP/2.0 Record-Route: <sip:192.168.10.3;lr=on;ftag=42d5a8fbdbb60640o0> Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1 Via: SIP/2.0/UDP 192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed From: <sip:[email protected]>;tag=42d5a8fbdbb60640o0 To: "Opensips-14x" <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE Max-Forwards: 69 Contact: <sip:[email protected]:5060;nat=yes;nat=yes> Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 263 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp P-hint: inbound->inbound P-hint: Route[20]: Rtpproxy P-hint: Route[20]: Rtpproxy v=0 o=- 811136 811136 IN IP4 192.168.10.19 s=- c=IN IP4 192.168.10.3192.168.10.3 t=0 0 m=audio 3500435006 RTP/AVP 18 101 a=rtpmap:18 G729a/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes # U +0.000123 192.168.10.3:5060 -> 192.168.10.30:5064 CANCEL sip:[email protected]:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.0 From: <sip:[email protected]>;tag=42d5a8fbdbb60640o0 Call-ID: [email protected] To: "Opensips-14x" <sip:[email protected]> CSeq: 102 CANCEL Max-Forwards: 70 User-Agent: OpenSIPS (1.4.5-notls (i386/linux)) Content-Length: 0 # U +0.001572 192.168.10.3:5070 -> 192.168.10.3:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1;received=192.168.10.3 Via: SIP/2.0/UDP 192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed From: <sip:[email protected]>;tag=42d5a8fbdbb60640o0 To: "Opensips-14x" <sip:[email protected]>;tag=as50300fb2 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000244 192.168.10.3:5060 -> 192.168.10.3:5070 ACK sip:[email protected]:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1 From: <sip:[email protected]>;tag=42d5a8fbdbb60640o0 Call-ID: [email protected] To: "Opensips-14x" <sip:[email protected]>;tag=as50300fb2 CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSIPS (1.4.5-notls (i386/linux)) Content-Length: 0 regardss -- rickygm http://gnuforever.homelinux.com _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
