1) your asterisk box talks to OpenSIPS 2) in that case OpenSIPS should traverse NAT 3) you should not do nat=yes for that device since Asterisk talks to OpenSIPS (but then it might not matter)
Either take OpenSIPS out of the way or configure NAT traversal w/media and it should work Martin On Sun, Apr 12, 2009 at 9:22 PM, troxlinux <[email protected]> wrote: > uff , no me fije que envié un mensaje en español a la lista de ingles ... > > I send sip log > > > --- > Retransmitting #2 (NAT) to 192.168.10.3:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3 > Via: SIP/2.0/UDP > 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d > Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e> > From: "Lucy" <sip:[email protected]>;tag=a363e3a864940c0e > To: <sip:*[email protected]>;tag=as3a76126d > Call-ID: [email protected] > CSeq: 30032 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:*[email protected]:5070> > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=root 3005 3005 IN IP4 192.168.10.3 > s=session > c=IN IP4 192.168.10.3 > t=0 0 > m=audio 13584 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > --- > Retransmitting #3 (NAT) to 192.168.10.3:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3 > Via: SIP/2.0/UDP > 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d > Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e> > From: "Lucy" <sip:[email protected]>;tag=a363e3a864940c0e > To: <sip:*[email protected]>;tag=as3a76126d > Call-ID: [email protected] > CSeq: 30032 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:*[email protected]:5070> > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=root 3005 3005 IN IP4 192.168.10.3 > s=session > c=IN IP4 192.168.10.3 > t=0 0 > m=audio 13584 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > --- > -- <SIP/111-08d20da8> Playing 'vm-password' (language 'es') > Retransmitting #4 (NAT) to 192.168.10.3:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3 > Via: SIP/2.0/UDP > 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d > Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e> > From: "Lucy" <sip:[email protected]>;tag=a363e3a864940c0e > To: <sip:*[email protected]>;tag=as3a76126d > Call-ID: [email protected] > CSeq: 30032 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:*[email protected]:5070> > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=root 3005 3005 IN IP4 192.168.10.3 > s=session > c=IN IP4 192.168.10.3 > t=0 0 > m=audio 13584 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > --- > -- <SIP/111-08d20da8> Playing 'vm-youhave' (language 'es') > Reliably Transmitting (NAT) to 192.168.10.3:5060: > OPTIONS sip:192.168.10.3 SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK63ca3652;rport > From: "asterisk" <sip:[email protected]:5070>;tag=as690b573d > To: <sip:192.168.10.3> > Contact: <sip:[email protected]:5070> > Call-ID: [email protected] > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 13 Apr 2009 02:20:27 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > --- > twoxserver*CLI> > <--- SIP read from 192.168.10.3:5060 ---> > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK63ca3652;rport=5070 > From: "asterisk" <sip:[email protected]:5070>;tag=as690b573d > To: <sip:192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.478d > Call-ID: [email protected] > CSeq: 102 OPTIONS > Server: OpenSIPS (1.5.0-notls (i386/linux)) > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > Really destroying SIP dialog > '[email protected]' Method: OPTIONS > -- <SIP/111-08d20da8> Playing 'digits/6' (language 'es') > -- <SIP/111-08d20da8> Playing 'vm-messages' (language 'es') > -- <SIP/111-08d20da8> Playing 'vm-first' (language 'es') > -- <SIP/111-08d20da8> Playing 'vm-message' (language 'es') > == Parsing '/var/spool/asterisk/voicemail/default/111/INBOX/msg0000.txt': > Found > Retransmitting #5 (NAT) to 192.168.10.3:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3 > Via: SIP/2.0/UDP > 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d > Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e> > From: "Lucy" <sip:[email protected]>;tag=a363e3a864940c0e > To: <sip:*[email protected]>;tag=as3a76126d > Call-ID: [email protected] > CSeq: 30032 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:*[email protected]:5070> > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=root 3005 3005 IN IP4 192.168.10.3 > s=session > c=IN IP4 192.168.10.3 > t=0 0 > m=audio 13584 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > --- > -- <SIP/111-08d20da8> Playing 'vm-received' (language 'es') > -- <SIP/111-08d20da8> Playing 'digits/yesterday' (language 'es') > -- <SIP/111-08d20da8> Playing 'digits/at' (language 'es') > -- <SIP/111-08d20da8> Playing 'digits/8' (language 'es') > -- <SIP/111-08d20da8> Playing 'digits/30' (language 'es') > Retransmitting #6 (NAT) to 192.168.10.3:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3 > Via: SIP/2.0/UDP > 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d > Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e> > From: "Lucy" <sip:[email protected]>;tag=a363e3a864940c0e > To: <sip:*[email protected]>;tag=as3a76126d > Call-ID: [email protected] > CSeq: 30032 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:*[email protected]:5070> > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=root 3005 3005 IN IP4 192.168.10.3 > s=session > c=IN IP4 192.168.10.3 > t=0 0 > m=audio 13584 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > --- > -- <SIP/111-08d20da8> Playing 'digits/and' (language 'es') > -- <SIP/111-08d20da8> Playing 'digits/9' (language 'es') > -- <SIP/111-08d20da8> Playing 'digits/p-m' (language 'es') > -- <SIP/111-08d20da8> Playing > '/var/spool/asterisk/voicemail/default/111/INBOX/msg0000' (language > 'es') > [Apr 12 20:20:36] WARNING[3528]: app_voicemail.c:5619 play_message: > Playback of message > /var/spool/asterisk/voicemail/default/111/INBOX/msg0000 failed > -- <SIP/111-08d20da8> Playing 'vm-advopts' (language 'es') > [Apr 12 20:20:38] WARNING[3140]: chan_sip.c:1976 retrans_pkt: Maximum > retries exceeded on transmission [email protected] for > seqno 30032 (Critical Response) -- See doc/sip-retransmit.txt. > [Apr 12 20:20:38] WARNING[3140]: chan_sip.c:1998 retrans_pkt: Hanging > up call [email protected] - no reply to our critical > packet (see doc/sip-retransmit.txt). > == Spawn extension (netsoluciones, *981, 2) exited non-zero on > 'SIP/111-08d20da8' > Really destroying SIP dialog '[email protected]' Method: INVITE > > > 2009/4/12 Alex Balashov <[email protected]>: >> Mejor que obtengamos un packet capture para investigarlo mas. > > sera un bug o algo por el estilo > > saludoss > > -- > rickygm > > http://gnuforever.homelinux.com > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
