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
