- sr-users - kamailio.org

[Users] Call dropped after 6 ACKs. Branch=0 problem?
by kjcsb 02 Apr '07

02 Apr '07

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, the call is established but often (but not always) eventually dropped. In the following dialogue, it can be seen that when the ACK is passed from the gateway on to Asterisk, OpenSER appends a Via with branch=0. After a number of these ACKs are passed on, the call is eventually dropped. It has been suggested by someone on the Asterisk list that "While "branch=0" is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of the transaction." i.e. the implication is that it's an OpenSER problem. Does that seem right? If so how can it be corrected? If not, could anyone advise what the problem might be or suggest any resolutions? U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 INVITE sip:6499777777@domain.co.nz SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 147.202.nnn.nnn:5060 "Noisy feedback tells: pid=29716 req_src_ip=202.180.nnn.nnn req_src_port=5060 in_uri=sip:6499777777@domain.co.nz out_uri=sip:6499777777@203.89.nnn.nnn via_cnt==1" U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16301 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16301 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK013cd54d;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK013cd54d;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK54b95a19;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK54b95a19;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK46a352b9;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK46a352b9;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK2da46c51;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK2da46c51;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK23a5accc;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK23a5accc;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK153c7273;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK153c7273;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK01e676a5;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK01e676a5;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0a0ddae8;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0a0ddae8;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 BYE sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 BYE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKe5ab.311dd295.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKe5ab.311dd295.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Thanks Cameron

1 0

[Serusers] Unreacheable destination problems
by Amanda Mattiuz 01 Apr '07

01 Apr '07

Hello! I'm using SER + rtpproxy, but altough the NAT problem seems to be solved, I get no audio ... analysing the traffic with ethereal, I realized that the traffic going from the proxy to the UAs never arrives (ICMP unreacheable destination)... I assume it is because the users have their ports blocked, is that correct? I changed the rtpproxy port range (PORT_MIN and PORT_MAX ) to lower ports, but still I got the same messages... Does anybody knows anything that can be done by me in terms of configurations of the rtpproxy or ser.cfg or whatever, given the fact that I don't have power to open the ports??..... Thanks, Amanda -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

1 0

[Users] Re: Digest Authentication Problem with header parsing problem
by Abdurrahman Sahin 01 Apr '07

01 Apr '07

hi all; i solved my problem; i was setting my server's ip address as the realm, and i guess providing an ip address as the realm causes problem. so i changed the realm name to a domain name , then set a sip proxy address for the client i also changed SIP_DOMAIN to a domain name. ----- Original Message ----- From: Abdurrahman Sahin To: users(a)openser.org Sent: Sunday, April 01, 2007 3:00 PM Subject: Digest Authentication Problem with header parsing problem hi all; i'm trying to activate digest authentication with mysql, but i see a strange parsing error in the logs. here is my logser log ================== Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: New request - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=200 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG: get_hdr_body : content_length=0 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: found end of header Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: find_first_route: No Route headers found Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: loose_route: There is no Route HF Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=4000 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: pre_auth(): Credentials with given realm not found ] in status 6: <<"apo"<sip:apo@ua_public_ip_address;tag=5d692f7d>> . R: parse_to : unexpected char [ Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: ERROR:parse_from_header: bad from header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: set_err_info: ec: 1, el: 3, ei: 'error parsing From' Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: xl_get_from_attr: ERROR cannot parse FROM header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: Register authentication failed - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: build_auth_hf(): 'WWW-Authenticate: Digest realm="server_public_ip_address", nonce ' 60f8cf5a7a5eb40c26f914280272b51f58caaa7" Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: receive_msg: cleaning up and sip msg received at the server side ============================== U ua_public_ip_address:28443 -> 192.168.200.2:5060 REGISTER sip:server_public_ip_address SIP/2.0. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:apo@ua_public_ip_address:28443;rinstance=1ff3024a97431241>. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4f01. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="server_public_ip_address", nonce="460f8cf5a7a5eb40c26f914280272b51f58caaa7". Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. openser.cfg file register section ======================================= # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("server_public_ip_address", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("server_public_ip_address", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; }

1 0

[Users] Digest Authentication Problem with header parsing problem
by Abdurrahman Sahin 01 Apr '07

01 Apr '07

hi all; i'm trying to activate digest authentication with mysql, but i see a strange parsing error in the logs. here is my logser log ================== Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: New request - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=200 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: DEBUG: get_hdr_body : content_length=0 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: found end of header Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: find_first_route: No Route headers found Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: loose_route: There is no Route HF Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=4000 Apr 1 13:39:04 testsrv200 /usr/local/sbin/openser[577]: pre_auth(): Credentials with given realm not found ] in status 6: <<"apo"<sip:apo@ua_public_ip_address;tag=5d692f7d>> . R: parse_to : unexpected char [ Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: ERROR:parse_from_header: bad from header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: set_err_info: ec: 1, el: 3, ei: 'error parsing From' Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: xl_get_from_attr: ERROR cannot parse FROM header Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: Register authentication failed - M=REGISTER RURI=sip:server_public_ip_address F=<null> T=sip:apo@server_public_ip_address IP=ua_public_ip_address ID=NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY. Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: build_auth_hf(): 'WWW-Authenticate: Digest realm="server_public_ip_address", nonce ' 60f8cf5a7a5eb40c26f914280272b51f58caaa7" Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: parse_headers: flags=ffffffffffffffff Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: check_via_address(ua_public_ip_address, ua_public_ip_address, 0) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 13:39:05 testsrv200 /usr/local/sbin/openser[577]: receive_msg: cleaning up and sip msg received at the server side ============================== U ua_public_ip_address:28443 -> 192.168.200.2:5060 REGISTER sip:server_public_ip_address SIP/2.0. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:apo@ua_public_ip_address:28443;rinstance=1ff3024a97431241>. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. . # U 192.168.200.2:5060 -> ua_public_ip_address:28443 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP ua_public_ip_address:28443;branch=z9hG4bK-d87543-08523511e5151c18-1--d87543-;rport=28443. To: "apo"<sip:apo@server_public_ip_address>;tag=329cfeaa6ded039da25ff8cbb8668bd2.4f01. From: "apo"<sip:apo@ua_public_ip_address;tag=5d692f7d. Call-ID: NWYyZGNjMWRjYjFlNTk5YjlhMjE5ODA4MWQ0ZGFiNTY.. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="server_public_ip_address", nonce="460f8cf5a7a5eb40c26f914280272b51f58caaa7". Server: OpenSER (1.2.0-notls (x86_64/linux)). Content-Length: 0. openser.cfg file register section ======================================= # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("server_public_ip_address", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("server_public_ip_address", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; }

1 0

[Users] UA simulator
by Michel Bensoussan 01 Apr '07

01 Apr '07

Hello This question is for openser developers. How do you test your code? Is there a UA simulator to generate many SIP messages and tests different scenarios? Thanks and regards, Michel.

2 1

[Users] how to handle REFER 302-moved temporarily
by zm23 31 Mar '07

31 Mar '07

Hello All, I am trying to find out the proper/best way to handle REFERs coming from phone. My setup is as follows: Cisco gw openser phone gw 2 DID 1234 -----> Invite -----------------> --------------------> <----- 302 refer <--------------------- <--------------------302 moved.... new #5678 sent back After receiving new invite, openser sends it out to another gateway 5678 ----> invite ----------------------> ---------------------------------------------------> yet i am still seeing the 302 messages sent back from proxy to cisco gw. <------------------------------- -- 302 moved Call does connect to 5678 but is dropped after few seconds with only one way audio. What needs to be done so that when second invite hits proxy, there are no more 302s sent back? Should I process the 302 in proxy itself? What is the best way to make the 2nd call to 5678 so that it can be identified as a separate call? Thank you in advance for your help in this regards. -- Zahid

2 3

[Users] bad user name/uid number: -u openser
by Abdurrahman Sahin 31 Mar '07

31 Mar '07

hi all; i just received "bad user name/uid number: -u openser" error msg when i try to start openser server, after installing openser database on mysql and using openser.cfg generator scpript at sipwise.com /rtpproxy section any help about this msg is appreciated.

2 1

Re: [Users] Re: openser behind nat UAs behind NAT
by Abdurrahman Sahin 31 Mar '07

31 Mar '07

hi; when i use a stun server i can register my UAs , but i didnt like this method, i think i should be able to do it without a stun server at first, i see nathelper Module in the documentatios. is there somebody to show how to use nathelper module? ----- Original Message ----- From: raviprakash sunkara To: asahin Sent: Saturday, March 31, 2007 7:25 AM Subject: Re: [Users] Re: openser behind nat UAs behind NAT Hi Asahin, When U with OpenSER Behind NAT, I need to know that is OpenSER is Behind the NAT, which having the Firewall router/NAT. Is the OpenSER is private IP or Public IP (ISP) if the OpenSER is Having the Private IP, of that IP is should forwards the SIP port and RTP ports, and in openserclt, SIP Domain is the public ip. alias should be the Sip domain name In Register Section if!www_authorize("alias name or realm", " subscriber" ) On 3/31/07, asahin <abdsahin(a)gmail.com> wrote: hi all; i am now able to register my client with the openserver server. i used STUN server option of x-lite and stun.voipuser.org as the stun server i also set/exported SIP_DOMAIN variable to my public openser server ip address. i wonder do i have to use a stun server ? ----- Original Message ----- From: asahin To: users(a)openser.org Sent: Friday, March 30, 2007 10:26 PM Subject: openser behind nat UAs behind NAT hi; i installed and tested openser on the internal network, it was working. i tried to test it behind NAT with x-lite sip client, but it failed. i defined a port mapping on the adsl modem for udp/tcp 5060 ports to forward the packets to the openser installed machine. when i try to register to openser i received a 408 request timeout message. i guess its due to external ip of the openser server. i think i should define external ip of the modem to the openser as if it's its own ip, but dont where to define it. here is the ngrep dump at the server. U external_ip_of_ua:23975 -> 192.168.200.2:5060 REGISTER sip:external_ip_of_openser SIP/2.0..Via: SIP/2.0/UDP external_ip_of_ua:23975;branch=z9hG4bK-d87543-307c62021a71bf6d-1--d87543-;rport..Max-Forwards: 70..Contact: <sip:apo@external_ip_of_ua:23975;rinstance=6a2c0ccf1a30b6bf>..To: "apo"<sip:apo@external_ip_of_openser>..From: "apo"<sip:apo@external_ip_of_ua;tag=da6a3851..Call-ID: ODJmOTVkNmIxNjE0NWM3MWNiNTQ4ZjFlMmVkZDZmYjQ...CSeq: 1 REGISTER.. Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1006e stamp 34025..Content-Length: 0....#U 192.168.200.2:5060 -> external_ip_of_openser:5060 REGISTER sip:external_ip_of_openser SIP/2.0..Via: SIP/2.0/UDP 192.168.200.2;branch=z9hG4bKd734.53940821.0..Via: SIP/2.0/UDP external_ip_of_ua:23975;branch=z9hG4bK-d87543-307c62021a71bf6d-1--d87543-;rport=23975..Max-Forwards: 69..Contact: <sip:apo@external_ip_of_ua:23975;rinstance=6a2c0ccf1a30b6bf>..To: "apo"<sip:apo@external_ip_of_openser>..From: "apo"<sip:apo@external_ip_of_ua;tag=da6a3851..Call- ID: ODJmOTVkNmIxNjE0NWM3MWNiNTQ4ZjFlMmVkZDZmYjQ...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, B YE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1006e stamp 34025..Content-Length: 0..P-hint: out bound.... my openser.cfg file is the initial openser openser.cfg file i didnt change it. ------------- # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib64/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

2 1

[Users] Re: openser behind nat UAs behind NAT
by asahin 30 Mar '07

30 Mar '07

hi all; i am now able to register my client with the openserver server. i used STUN server option of x-lite and stun.voipuser.org as the stun server i also set/exported SIP_DOMAIN variable to my public openser server ip address. i wonder do i have to use a stun server ? ----- Original Message ----- From: asahin To: users(a)openser.org Sent: Friday, March 30, 2007 10:26 PM Subject: openser behind nat UAs behind NAT hi; i installed and tested openser on the internal network, it was working. i tried to test it behind NAT with x-lite sip client, but it failed. i defined a port mapping on the adsl modem for udp/tcp 5060 ports to forward the packets to the openser installed machine. when i try to register to openser i received a 408 request timeout message. i guess its due to external ip of the openser server. i think i should define external ip of the modem to the openser as if it's its own ip, but dont where to define it. here is the ngrep dump at the server. U external_ip_of_ua:23975 -> 192.168.200.2:5060 REGISTER sip:external_ip_of_openser SIP/2.0..Via: SIP/2.0/UDP external_ip_of_ua:23975;branch=z9hG4bK-d87543-307c62021a71bf6d-1--d87543-;rport..Max-Forwards: 70..Contact: <sip:apo@external_ip_of_ua:23975;rinstance=6a2c0ccf1a30b6bf>..To: "apo"<sip:apo@external_ip_of_openser>..From: "apo"<sip:apo@external_ip_of_ua;tag=da6a3851..Call-ID: ODJmOTVkNmIxNjE0NWM3MWNiNTQ4ZjFlMmVkZDZmYjQ...CSeq: 1 REGISTER.. Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1006e stamp 34025..Content-Length: 0....#U 192.168.200.2:5060 -> external_ip_of_openser:5060 REGISTER sip:external_ip_of_openser SIP/2.0..Via: SIP/2.0/UDP 192.168.200.2;branch=z9hG4bKd734.53940821.0..Via: SIP/2.0/UDP external_ip_of_ua:23975;branch=z9hG4bK-d87543-307c62021a71bf6d-1--d87543-;rport=23975..Max-Forwards: 69..Contact: <sip:apo@external_ip_of_ua:23975;rinstance=6a2c0ccf1a30b6bf>..To: "apo"<sip:apo@external_ip_of_openser>..From: "apo"<sip:apo@external_ip_of_ua;tag=da6a3851..Call- ID: ODJmOTVkNmIxNjE0NWM3MWNiNTQ4ZjFlMmVkZDZmYjQ...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, B YE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1006e stamp 34025..Content-Length: 0..P-hint: out bound.... my openser.cfg file is the initial openser openser.cfg file i didnt change it. ------------- # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib64/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

1 0

[Users] openser behind nat UAs behind NAT
by asahin 30 Mar '07

30 Mar '07

hi; i installed and tested openser on the internal network, it was working. i tried to test it behind NAT with x-lite sip client, but it failed. i defined a port mapping on the adsl modem for udp/tcp 5060 ports to forward the packets to the openser installed machine. when i try to register to openser i received a 408 request timeout message. i guess its due to external ip of the openser server. i think i should define external ip of the modem to the openser as if it's its own ip, but dont where to define it. here is the ngrep dump at the server. U external_ip_of_ua:23975 -> 192.168.200.2:5060 REGISTER sip:external_ip_of_openser SIP/2.0..Via: SIP/2.0/UDP external_ip_of_ua:23975;branch=z9hG4bK-d87543-307c62021a71bf6d-1--d87543-;rport..Max-Forwards: 70..Contact: <sip:apo@external_ip_of_ua:23975;rinstance=6a2c0ccf1a30b6bf>..To: "apo"<sip:apo@external_ip_of_openser>..From: "apo"<sip:apo@external_ip_of_ua;tag=da6a3851..Call-ID: ODJmOTVkNmIxNjE0NWM3MWNiNTQ4ZjFlMmVkZDZmYjQ...CSeq: 1 REGISTER.. Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1006e stamp 34025..Content-Length: 0....#U 192.168.200.2:5060 -> external_ip_of_openser:5060 REGISTER sip:external_ip_of_openser SIP/2.0..Via: SIP/2.0/UDP 192.168.200.2;branch=z9hG4bKd734.53940821.0..Via: SIP/2.0/UDP external_ip_of_ua:23975;branch=z9hG4bK-d87543-307c62021a71bf6d-1--d87543-;rport=23975..Max-Forwards: 69..Contact: <sip:apo@external_ip_of_ua:23975;rinstance=6a2c0ccf1a30b6bf>..To: "apo"<sip:apo@external_ip_of_openser>..From: "apo"<sip:apo@external_ip_of_ua;tag=da6a3851..Call- ID: ODJmOTVkNmIxNjE0NWM3MWNiNTQ4ZjFlMmVkZDZmYjQ...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANCEL, OPTIONS, B YE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 1006e stamp 34025..Content-Length: 0..P-hint: out bound.... my openser.cfg file is the initial openser openser.cfg file i didnt change it. ------------- # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib64/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

1 0

Re: [Users] 1.1.1 to 1.2 : radius failed
by Bogdan-Andrei Iancu 30 Mar '07

30 Mar '07

Bodin, looks like the error has its roots in the radiusclient library: rc_check_reply: received invalid reply digest from RADIUS server I'm not a RADIUS expert, so I cannot say more than this.....:( ...maybe try to use wireshark to see the packages between server and client radius. regards, bogdan Bodin Bruno wrote: > Bogdan-Andrei Iancu a écrit : >> Hi Bodin, >> >> Have you try to see if your radius cfg is ok? see: >> >> http://www.openser.org/docs/openser-radius-1.0.x.html#testing_radius_commun… >> >> >> regards, >> bogdan >> >> >> Bodin Bruno wrote: >>> Bogdan-Andrei Iancu a écrit : >>>> Hi Bodin, >>>> >>>> look into the system log (/var/log/{syslog|messages}) to get the >>>> error generated by the radius library. >>>> >>>> regards, >>>> bogdan >>>> >>>> Bodin Bruno wrote: >>>>> Same radius client config, same radius server config, just 1.2 >>>>> from 1.1.1 but an error : >>>>> >>>>> 5(23229) ERROR:auth_radius:radius_authorize_sterman: rc_auth failed >>>>> >>>>> Radius server get a good authenticate but openser failed this >>>>> after :( >>>> >>> On radius side, no error, in /var/log/messages there is only thise >>> line : >>> Mar 28 14:26:21 zebulon openser: rc_check_reply: received invalid >>> reply digest from RADIUS server >>> >>> >> > Hi, > Yes for sure. When a use 1.1.1 it work , and when I use 1.2 that > doesnt work, I thin k there is a difference between the two ... > > thank >

4 6

[Users] 0-terminated packet
by Tobias Lindgren 30 Mar '07

30 Mar '07

Hi all, Can anyone explain what this error mean? Mar 30 16:50:05 openser /sbin/openser[22632]: WARNING: udp_rcv_loop: upstream bug - 0-terminated packet from 192.168.0.1 50195 I'm getting this after an "Accepted" of a NOTIFY-message. Br, /Tobias

2 1

[Users] users authentication
by Joao Pereira 30 Mar '07

30 Mar '07

Hello I already have an OpenSER runnig nicely. Now I installed a second OpenSER (with other a newdomain) with the same database configuration and the same ser.cfg (except the IP). Everyting works fine... except the authentication... OpenSER always answers 401 Unauthorized. It only accepts clients when I comment this part: if (!www_authorize("newdomain.com", "subscriber")) { www_challenge("newdomain.com", "0"); break; }; The users exist in the database and I know the username / password. Whats happening? Thanks regards Joao Pereira

3 5

[Users] Unauthorized Calls - PLEASE HELP!
by Daryl Sanders 30 Mar '07

30 Mar '07

Hi Everyone, I aparently have something in my openser.cfg that is allowing unauthorized calls to go through to our PSTN gateways. I have included my config below for review. I would appreciate any help understanding how this might be happening. I am currently reviewing the CDRs from my PSTN gateways for clues as well. This is a pretty basic configuration with no NAT involved. Regards, Daryl route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; if (method=="INVITE" || method=="ACK" || method=="BYE") { setflag(1); }; if (method=="INVITE") { if (is_user_in("From","inactive")) { if (uri =~ "^sip:911@") { xlog("L_NOTICE", "[$Tf] R1: $ci -- Allowing 911 Emergency Call on Inactive User\n" ); } else { sl_send_reply("403", "Forbidden"); xlog("L_NOTICE", "[$Tf] R1: $ci -- User Inactive\n" ); return; }; }; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog( "L_NOTICE", "[$Tf] RR: $ci -- Loose Route $rm ($rd).\n" ); if (!t_relay()) { sl_reply_error(); }; return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="REGISTER") { route(2); return; } else if (method=="INVITE") { route(3); return; } else if (method=="BYE" || method=="CANCEL") { t_relay(); exit; } lookup("aliases"); if (uri!=myself) { route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- # Trusted Provider IPs if (!src_ip==x.x.x.x)&&(!src_ip==x.x.x.x)&&(!src_ip==x.x.x.x) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(1); return; }; if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){ sl_send_reply("479", "We do not forward to private IP addresses"); }; if ((uri =~ "^sip:0@")|| /* Operator Assistance */ (uri =~ "^sip:911@")|| /* 911 Emergency */ (uri =~ "^sip:411@")|| /* Directory Assistance */ (uri =~ "^sip:1[0-9]{10}@")) { /* Domestic PSTN */ route(4); return; }; if (uri=~"^sip:0111[0-9]*@") { # Kill calls to 011+1... (invalid dialing) sl_send_reply("406", "Not Acceptable"); return; } if (uri=~"^sip:011[0-9]*@") { # International PSTN if(!is_user_in("From","gateway1")) { strip(3); # Remove 011 for Gateway2 } route(4); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- prefix("+"); # add "+" to Request URI append_hf("P-Asserted-Identity: \"User\"<sip:+1$avp(s:rpid)@x.x.x.x>\r\n"); uac_replace_from("$fn","sip:+$fU@$fd:5060"); if(is_user_in("From","gateway1")) { force_send_socket(x.x.x.x:5060); xlog("L_NOTICE", "[$Tf] Message sent via IP-1\n" ); } else { force_send_socket(x.x.x.x:5060); xlog("L_NOTICE", "[$Tf] Message sent via IP-2\n" ); }; ds_select_domain("1","0"); route(1); } onreply_route[1] { # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_INFO", "[$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_INFO", "[$Tf] ORR: $ci -- $rs $rr\n" ); } } failure_route[2] { # 408 -- timeout -- typically the end party has not answered # Since we cancel t_on_failure() on a provisional response we should not be # getting a 408 timeout from a gateway at this stage.. it will just "fall through" # If fr_timer expires t_check_status("408") is true, although $rs is <null> if( t_check_status("408") ){ xlog( "L_NOTICE", "[$Tf] FR: $ci -- TIMEOUT for Gateway $rd\n" ); } else { xlog( "L_NOTICE", "[$Tf] FR: $ci -- $rs reason $rr\n" ); } # 403 -- Not a valid number, or possibly no permission to use the gateway if( t_check_status("403") ){ xlog("L_NOTICE", "[$Tf] FR: $ci -- SIP-$rs Forbidden\n" ); return; } # 486 -- User Busy if( t_check_status("486") ){ xlog("L_NOTICE", "[$Tf] FR: $ci -- SIP-$rs Destination Busy\n" ); return; } # 487 -- Request Cancelled (usually in response to a CANCEL transaction) if( t_check_status("487") ){ xlog("L_NOTICE", "[$Tf] FR: $ci -- SIP-$rs Request Cancelled\n" ); return; } # At this stage we try the next gateway, if no next gateway we bail. if( ds_next_domain() ){ t_on_reply("1"); t_on_failure("2"); xlog( "L_NOTICE", "[$Tf] FR: $ci Next gateway $fU -> $tU via $rd\n" ); if( !t_relay() ){ xlog( "L_WARN", "[$Tf] FR: $ci -- ERROR - Can not t_relay()\n" ); return; } return; } else { xlog( "L_WARN", "[$Tf] FR: $ci No more gateways -> 503.\n" ); t_reply("503", "Service unavailable -- no more gateways" ); return; } }

3 3

[Users] t_relay() with variables(proto:ip:port)
by Benko 30 Mar '07

30 Mar '07

Hello! I would like to relay messages to a location("proto:ip:port") i retrieve from a db-table with avp_db_query, unfortunately t_relay doesn't allow using avps or pseudo-variables currently(iirc). Do you know a workaround without using t_relay or is there a way to use avps/pseudo-variables with t_relay() since 1.2? Thx Christian

3 7

[Users] A Problem with Serweb server monitoring
by Jarkko Nevala 30 Mar '07

30 Mar '07

Hi. I'm having problems with Serweb server monitoring script. As Instructed Serweb install file I have cron job, that executes "read_ser_moni.php" once in a minute. Every time the script is executed I get this error on serweb log: "serweb [error] server monitoring - 500 command 't_stats' not available" It also happens, when I'm executing the script from command line or executing manually fifo command with openserctl ("openserctl fifo t_stat"). It seems, that openser receives the command, but doesen't know what to do with it, because I get this error on syslog from openser: ERROR: fifo_server: command t_stats is not available **** done consume INFO: fifo_server: command empty Has the t_stat fifo command depricated or am I just missing some parameter from my cfg. thanks. JN

2 1

Re: [Users] Radius and OpenSer
by raviprakash sunkara 30 Mar '07

30 Mar '07

Sorry , U can use Alepo radius server , this Alepo radius server should communicate with Radiusclient of the OpenSER radiusclient dictionary should mached the Alepo radius server. I think U have Alepo radius server On 3/30/07, Jidntt <jidntt(a)yahoo.com.ar> wrote: > > Hi list, > > > > I've a question… I've a billing system named Alepo. > > I need to know if OpenSer support radius for connect to Alepo and how to > configure this. > > > > Thanks > > > > Pablo > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

1 0

[Users] authentication with other DB
by Joao Pereira 30 Mar '07

30 Mar '07

Hello I just installed OpenSER and configured digest authentication with a local database. The content of the database was copied from other server. The ser.cfg and the database are the same... but OpenSER dont allow the users to register.... :( Whats the problem of doing a postgresql dump and then recover the DB in other PC? OpenSER doesnt allow it? Thanks regards Joao pereira

2 1

[Users] Radius and OpenSer
by Jidntt 30 Mar '07

30 Mar '07

Hi list, I've a question. I've a billing system named Alepo. I need to know if OpenSer support radius for connect to Alepo and how to configure this. Thanks Pablo

1 0

[Users] 1.1.1 to 1.2 : radius failed
by Bodin Bruno 30 Mar '07

30 Mar '07

Same radius client config, same radius server config, just 1.2 from 1.1.1 but an error : 5(23229) ERROR:auth_radius:radius_authorize_sterman: rc_auth failed Radius server get a good authenticate but openser failed this after :(

3 4

RE: [Users] out of memory
by Papadopoulos Georgios 30 Mar '07

30 Mar '07

Hi Bogdan, I found the following thread in the list http://www.openser.org/pipermail/users/2006-June/005339.html I suspect that I have the same issue. If I assume that our DNS is strange, then my 1.1.1-notls version will be affected, right? Is there a patch to fix this? Eventually I want to go to 1.2 but I have to do a lot of testing before I can put it in production. So for the moment I need a fix for 1.1.1. Also back in December I sent this to the list http://www.openser.org/pipermail/users/2006-December/008205.html >From the start, we had the issue of openser stopping to process any incoming requests. On December I thought it was DNS related. I am still in the dark about this. Lately I added logging and I discovered the "out of memory" error. I am guessing that the error was there from the start, I just did not know about it. Do all of these make any sense to you? And is there a way to test our DNS to determine whether it is buggy? In the meantime I will follow the instructions on the wiki to get the memory dump. thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, March 28, 2007 11:54 AM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory > > Hi Papadopoulos, > > the error you get complains about the lack of private memory (per > process) - the statistics show info about the shared memory. See: > http://www.openser.org/dokuwiki/doku.php/troubleshooting:memory > > but look for pkg instead of shm..... > > regards, > bogdan > > Papadopoulos Georgios wrote: > > Hello, > > > > Lately our OpenSER 1.1.1 is running out of memory and I get the > > following message > > 4(25190) get_hdr_field: out of memory The result is that it stops > > processing any incoming request. This is a production system and of > > course it causes a great deal of frustration. > > > > The last time this happened I got the following statistics before > > restarting > > > ./openserctl fifo get_statistics > > Total statistics = 37 > > Total modules = 5 > > Module name = core; statistics=8 > > core:rcv_requests = 3259334 > > core:rcv_replies = 8816 > > core:fwd_requests = 736 > > core:fwd_replies = 137 > > core:drop_requests = 743 > > core:drop_replies = 0 > > core:err_requests = 0 > > core:err_replies = 43 > > Module name = shmem; statistics=6 > > shmem:total_size = 268435456 > > shmem:used_size = 1675864 > > shmem:real_used_size = 1835104 > > shmem:max_used_size = 4399048 > > shmem:free_size = 266600352 > > shmem:fragments = 17710 > > Module name = sl; statistics=9 > > sl:1xx_replies = 0 > > sl:2xx_replies = 1690410 > > sl:3xx_replies = 43 > > sl:4xx_replies = 334827 > > sl:5xx_replies = 1185629 > > sl:6xx_replies = 0 > > sl:sent_replies = 3210909 > > sl:sent_err_replies = 0 > > sl:received_ACKs = 743 > > Module name = tm; statistics=11 > > tm:received_replies = 8636 > > tm:relayed_replies = 7096 > > tm:local_replies = 1718 > > tm:UAS_transactions = 7003 > > tm:UAC_transactions = 94 > > tm:2xx_transactions = 1528 > > tm:3xx_transactions = 3 > > tm:4xx_transactions = 5129 > > tm:5xx_transactions = 468 > > tm:6xx_transactions = 0 > > tm:inuse_transactions = 0 > > Module name = usrloc; statistics=3 > > usrloc:location-users = 0 > > usrloc:location-contacts = 0 > > usrloc:location-expires = 7282 > > > > I am not sure how to read these numbers but I see a big "free_size" > > and still OpenSER is out of memory. Any suggestions? > > > > thank you > > > > George > > > > > > > > > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, > > please notify the sender immediately. Unless you are the intended > > recipient or his/her representative you are not authorized to, and > > must not, read, copy, distribute, use or retain this message or any > > part of it. E-mail transmission cannot be guaranteed to be > secure or > > error-free as information could be intercepted, corrupted, lost, > > destroyed, arrive late or incomplete, or contain viruses. > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

2 1

RE: [Users] out of memory
by Papadopoulos Georgios 30 Mar '07

30 Mar '07

Hi Bogdan I followed the instructions to get the memory dump. Now openser will not start. Mar 28 13:44:56 ser1 openser[1567]: convert_row: No memory left Mar 28 13:44:56 ser1 openser[1567]: convert_rows: Error while converting row #2536 Mar 28 13:44:56 ser1 openser[1567]: convert_result: Error while converting rows Mar 28 13:44:56 ser1 openser[1567]: store_result: Error while converting result Mar 28 13:44:56 ser1 openser[1567]: preload_udomain(): Error while doing db_query Mar 28 13:44:56 ser1 openser[1567]: register_udomain(): Error while preloading domain 'location' Mar 28 13:44:56 ser1 openser[1567]: domain_fixup(): Error while registering domain Mar 28 13:44:56 ser1 openser[1567]: ERROR: fix_actions: fixing failed (code=-1) at cfg line 282 Mar 28 13:44:56 ser1 openser[1567]: ERROR: fix_expr : fix_actions error In MySQL location table we have around 2600 rows. The default version of openser has no problem loading them at startup. The version compiled with DBG_QM_MALLOC and without F_MALLOC cannot load the location table. The same happens if I increase the shared memory with the -m option. Does this mean that ul is stored in private memory? Does this mean I should increase PKG_MEM_POOL_SIZE? thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, March 28, 2007 11:54 AM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory > > Hi Papadopoulos, > > the error you get complains about the lack of private memory (per > process) - the statistics show info about the shared memory. See: > http://www.openser.org/dokuwiki/doku.php/troubleshooting:memory > > but look for pkg instead of shm..... > > regards, > bogdan > > Papadopoulos Georgios wrote: > > Hello, > > > > Lately our OpenSER 1.1.1 is running out of memory and I get the > > following message > > 4(25190) get_hdr_field: out of memory The result is that it stops > > processing any incoming request. This is a production system and of > > course it causes a great deal of frustration. > > > > The last time this happened I got the following statistics before > > restarting > > > ./openserctl fifo get_statistics > > Total statistics = 37 > > Total modules = 5 > > Module name = core; statistics=8 > > core:rcv_requests = 3259334 > > core:rcv_replies = 8816 > > core:fwd_requests = 736 > > core:fwd_replies = 137 > > core:drop_requests = 743 > > core:drop_replies = 0 > > core:err_requests = 0 > > core:err_replies = 43 > > Module name = shmem; statistics=6 > > shmem:total_size = 268435456 > > shmem:used_size = 1675864 > > shmem:real_used_size = 1835104 > > shmem:max_used_size = 4399048 > > shmem:free_size = 266600352 > > shmem:fragments = 17710 > > Module name = sl; statistics=9 > > sl:1xx_replies = 0 > > sl:2xx_replies = 1690410 > > sl:3xx_replies = 43 > > sl:4xx_replies = 334827 > > sl:5xx_replies = 1185629 > > sl:6xx_replies = 0 > > sl:sent_replies = 3210909 > > sl:sent_err_replies = 0 > > sl:received_ACKs = 743 > > Module name = tm; statistics=11 > > tm:received_replies = 8636 > > tm:relayed_replies = 7096 > > tm:local_replies = 1718 > > tm:UAS_transactions = 7003 > > tm:UAC_transactions = 94 > > tm:2xx_transactions = 1528 > > tm:3xx_transactions = 3 > > tm:4xx_transactions = 5129 > > tm:5xx_transactions = 468 > > tm:6xx_transactions = 0 > > tm:inuse_transactions = 0 > > Module name = usrloc; statistics=3 > > usrloc:location-users = 0 > > usrloc:location-contacts = 0 > > usrloc:location-expires = 7282 > > > > I am not sure how to read these numbers but I see a big "free_size" > > and still OpenSER is out of memory. Any suggestions? > > > > thank you > > > > George > > > > > > > > > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, > > please notify the sender immediately. Unless you are the intended > > recipient or his/her representative you are not authorized to, and > > must not, read, copy, distribute, use or retain this message or any > > part of it. E-mail transmission cannot be guaranteed to be > secure or > > error-free as information could be intercepted, corrupted, lost, > > destroyed, arrive late or incomplete, or contain viruses. > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

2 1

[Serusers] authentication with other DB
by Joao Pereira 30 Mar '07

30 Mar '07

Hello I just installed SER and configured digest authentication with a local database. The content of the database was copied from other server. The ser.cfg and the database are the same... but SER dont allow the users to register.... :( Whats the problem of doing a postgresql dump and then recover the DB in other PC? SER doenst allow it? Thanks regards Joao pereira

1 0

[Serusers] MySql Module Error!!
by KUMAR 30 Mar '07

30 Mar '07

I was using SER 0.9.6 earlier. Recently, I downloaded the SER 2.0 source code and installed it as follows. make group_include="standard standard-dep stable experimental presence" all It went without any problems. But there seems to be some kind of problem with loading of mysql module. I have also removed the references for fifo from my old configuration file. When i check the configuration file with ser -c i get the following error message: 0(13302) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: fm_malloc 0(13302) parse error (34,13-14): failed to load module Can anyone please help me as soon as possible. Am i missing something?? Thank you in advance kumar

1 0

Re: Fwd: [Users] Openser fails when compiled on Solaris 64bit
by Sergio Gutierrez 30 Mar '07

30 Mar '07

Hi Daniel. In the meantime, I have begun to test using more standard conditions; I am not using the optimized compiler and I am running openser compiled on 32 bits. I have discovered an even nastier issue. When Openser is configured in forking mode, and with or without log_stderror, it presents the same symptom I reported when compiled on 64 bits (it exhausts the shared memory reservation, and crashes with core dumping). This is the backtrace: (gdb) bt #0 0x00000000 in ?? () #1 0xfe989b3c in getRow (ipType=1, ipAddress=0x141888) at openserSIPPortTable.c:125 #2 0xfe989c5c in createRowsFromIPList (theList=0x141a50, listSize=1, protocol=0, snmpIndex=0xffbff870) at openserSIPPortTable.c:201 #3 0xfe98a07c in init_openserSIPPortTable () at openserSIPPortTable.c:241 #4 0xfe98d708 in spawn_agentx_child () at sub_agent.c:74 #5 0xfe98cf04 in mod_child_init (rank=0) at snmpstats.c:271 #6 0x00059ca8 in init_mod_child (m=0x133840, rank=0, type=0xc3c08 "PROC_MAIN") at sr_module.c:406 #7 0x00059c24 in init_mod_child (m=0x1338e0, rank=0, type=0xc3c08 "PROC_MAIN") at sr_module.c:395 #8 0x000336c4 in main_loop () at main.c:952 #9 0x00035be4 in main (argc=9, argv=0xe7800) at main.c:1399 When I configure fork=no, openser does not crash, and works right. I apologize if I should better create a new thread. If you suggest so I will do it. Thanks in advance for your attention. Kind regards Sergio Gutierrez. EPM Telecomunicaciones Medellín, Colombia Tel: 574 - 3950022 > ---------- Forwarded message ---------- > > > From: *Sergio Gutierrez* <saguti(a)gmail.com <mailto: saguti(a)gmail.com>> > > > Date: Mar 27, 2007 9:52 AM > > > Subject: Re: [Users] Openser fails when compiled on Solaris 64bit > > > To: daniel(a)voice-system.ro <mailto: daniel(a)voice-system.ro> > > > Cc: users(a)openser.org <mailto:users@openser.org> > > > > > > Hi Daniel. > > > > > > I am using mysql version 5.0.37, compiled on 64 bits too. > > > > > > These are the printing of the symbols you asked: > > > > > > (gdb) print row > > > $1 = (db_row_t *) 0x10026bf68 > > > (gdb) print ROW_VALUES(row) > > > No symbol "ROW_VALUES" in current context. > > > (gdb) print VAL_NULL(ROW_VALUES(row) > > > No symbol "VAL_NULL" in current context. > > > (gdb) > > > > > > > > > Thanks. > > > > > > > > > On 3/27/07, *Daniel-Constantin Mierla* < daniel(a)voice-system.ro > > > <mailto:daniel@voice-system.ro >> wrote: > > > > > > Hello Sergio, > > > > > > seems to be some data corrupted from what database returned. What > > > is the > > > database type you use, mysql? > > > > > > Could you print the the content of row and ROW_VALUES(row) (via > > print > > > command in gdb)? > > > > > > Thanks, > > > Daniel > > > > > > > > > On 03/26/07 23:46, Sergio Gutierrez wrote: > > > > Hi again Daniel. > > > > > > > > Testing other things trying to solve the problem I found that > > the > > > > segmentation fault occurs when Openser is restarted and there > > are > > > > records within location database table. > > > > If the table is empty initially, openser starts OK, and users > > can > > > > register without problem. > > > > > > > > Below is the backtrace of a error when the location table is not > > > empty: > > > > > > > > #0 0xffffffff7d90e2a0 in preload_udomain (_c=0x10026ad90, > > > > _d=0xffffffff5028e9f0) at udomain.c:404 > > > > 404 if (VAL_NULL(ROW_VALUES(row)) > > > > (gdb) bt > > > > #0 0xffffffff7d90e2a0 in preload_udomain (_c=0x10026ad90, > > > > _d=0xffffffff5028e9f0) at udomain.c:404 > > > > #1 0xffffffff7d915844 in child_init (_rank=1) at ul_mod.c:344 > > > > #2 0x0000000100082500 in init_mod_child (m=0x1, rank=1, > > > > type=0x1000f41b0 "CHILD") at > > > > /home/operador/openser-1.2.0-notls//sr_module.c:400 > > > > #3 0x0000000100082460 in init_mod_child (m=0x1, rank=1, > > > > type=0x1000f41b0 "CHILD") at > > > > /home/operador/openser-1.2.0-notls//sr_module.c:394 > > > > #4 0x0000000100082460 in init_mod_child (m=0x1, rank=1, > > > > type=0x1000f41b0 "CHILD") at > > > > /home/operador/openser-1.2.0-notls//sr_module.c:394 > > > > #5 0x0000000100082770 in init_child (rank=1) at > > > > /home/operador/openser-1.2.0-notls//sr_module.c:394 > > > > #6 0x0000000100032414 in main_loop () at > > > > /home/operador/openser-1.2.0-notls//main.c:724 > > > > #7 0x00000001000345a4 in main (argc=2, argv=0xffffff7eaeeec4ff) > > at > > > > /home/operador/openser-1.2.0-notls//main.c:1399 > > > > > > > > > > > > Thanks. > > > > > > > > > > > > > > > > > > > > > > > > On 3/26/07, *Sergio Gutierrez* < saguti(a)gmail.com > > > <mailto:saguti@gmail.com> > > > > <mailto: saguti(a)gmail.com <mailto:saguti@gmail.com >>> wrote: > > > > > > > > Hi Daniel. > > > > > > > > This is the backtrace of the core. > > > > Thanks > > > > > > > > #0 0xffffffff7d90ee9c in preload_udomain (_c=0x10026b7d800, > > > > > > _d=0x0) at > > > > /home/operador/openser- 1.2.0-notls > > /modules/usrloc//udomain.c:404 > > > > #1 0xffffffff7d9131c4 in child_init (_rank=1344858528) at > > > > /home/operador/openser- 1.2.0-notls > > /modules/usrloc//ul_mod.c:344 > > > > #2 0x0000000100082500 in init_mod_child (m=0x1, rank=1, > > > > type=0x1000f41b0 "CHILD") at > > > > /home/operador/openser- 1.2.0-notls//sr_module.c:400 > > > > #3 0x0000000100082460 in init_mod_child (m=0x1, rank=1, > > > > type=0x1000f41b0 "CHILD") at > > > > /home/operador/openser-1.2.0-notls //sr_module.c:394 > > > > #4 0x0000000100082460 in init_mod_child (m=0x1, rank=1, > > > > type=0x1000f41b0 "CHILD") at > > > > /home/operador/openser- 1.2.0-notls //sr_module.c:394 > > > > #5 0x0000000100082770 in init_child (rank=1) at > > > > /home/operador/openser-1.2.0-notls//sr_module.c:394 > > > > #6 0x0000000100032414 in main_loop () at > > > > /home/operador/openser- 1.2.0-notls//main.c:724 > > > > #7 0x00000001000345a4 in main (argc=2, > > argv=0xffffff7e7007a000) > > > > at /home/operador/openser-1.2.0-notls//main.c:1399 > > > > > > > > > > > > > > > > > > > > On 3/26/07, *Daniel-Constantin Mierla* < > > > daniel(a)voice-system.ro <mailto: daniel(a)voice-system.ro> > > > > <mailto: daniel(a)voice-system.ro > > > <mailto:daniel@voice-system.ro>>> wrote: > > > > > > > > Hello, > > > > > > > > can you send a gdb backtrace :-D -- I cannot read adb > > > backtrace. > > > > > > > > Cheers, > > > > Daniel > > > > > > > > > > > > On 03/26/07 19:42, Sergio Gutierrez wrote: > > > > > Hi Daniel. > > > > > > > > > > This is the backtrace, obtained with adb: > > > > > > > > > > usrloc.so`preload_udomain+0x4cc (1, ffffffff7228e9f0, > > > 100215a00, > > > > > ffffffff7d919e58, ffffffffffefe3f8, 0) > > > > > 0xffffffff7d9131bc(2, ffffffff7da1c9b0, > > ffffffff7d91a170, > > > > > ffffffff7228e9a0, ffffffff7da1b0a8, ffffffffffeff0c8) > > > > > init_mod_child+0xd8(100269930, 1, 1000f41b0, > > 1000f40f0, 1, > > > > 100269870) > > > > > init_mod_child+0x38(100269bb0, 1, 1000f41b0, > > 1000f40f0, 1, > > > > 1002699f0) > > > > > init_mod_child+0x38(100269d30, 1, 1000f41b0, > > 1000f40f0, 1, > > > > 100269c70) > > > > > init_child+0xa8(1, 100269df0, 1, 1000f4168, 10020d000, > > > 100269eb0) > > > > > main_loop+0xf34(0, 8c, 1000eefd8, ffffffff720083c4, > > > 1002694b0, 0) > > > > > main+0x1e10(100215, 9, 0, 10020d000, 1000ef5b8, > > 10020d000) > > > > > _start+0x7c(0, 0, 0, 0, 0, 0) > > > > > > > > > > On 3/26/07, *Sergio Gutierrez* <saguti(a)gmail.com > > > <mailto: saguti(a)gmail.com> > > > > <mailto: saguti(a)gmail.com <mailto: saguti(a)gmail.com>> > > > > > <mailto: saguti(a)gmail.com <mailto: saguti(a)gmail.com> > > > <mailto:saguti@gmail.com <mailto: saguti(a)gmail.com>>>> wrote: > > > > > > > > > > Hi Daniel. > > > > > > > > > > I used 512 for -m parameter. > > > > > > > > > > Would you like the backtrace of the core file? > > > > > > > > > > Thanks. > > > > > > > > > > > > > > > On 3/26/07, *Daniel-Constantin Mierla* < > > > > daniel(a)voice-system.ro <mailto: daniel(a)voice-system.ro> > > > <mailto: daniel(a)voice-system.ro <mailto:daniel@voice-system.ro>> > > > > > <mailto: daniel(a)voice-system.ro > > > <mailto: daniel(a)voice-system.ro> > > > > <mailto: daniel(a)voice-system.ro > > > <mailto:daniel@voice-system.ro>>>> wrote: > > > > > > > > > > Hello, > > > > > > > > > > could you send a backtrace? What was the value > > for > > > > parameter -m? > > > > > > > > > > Cheers, > > > > > Daniel > > > > > > > > > > > > > > > On 03/26/07 19:29, Sergio Gutierrez wrote: > > > > > > Hi > > > > > > > > > > > > I am trying to run Openser compiled on 64 > > > bits on a > > > > SPARC > > > > > Machine > > > > > > running Solaris 10. > > > > > > > > > > > > When I create a simple configuration file > > for > > > > testing radius > > > > > > integration, Openser starts to consume the > > whole > > > > memory > > > > > reservation > > > > > > (-m parameter) and fails with segmentation > > > fault error. > > > > > > > > > > > > after several tests, I have found that the > > > error is > > > > caused by > > > > > save() > > > > > > function (registrar module). > > > > > > > > > > > > This is the main route my configuration > > file: > > > > > > > > > > > > route { > > > > > > if(method=="REGISTER") > > > > > > { > > > > > > if(!radius_www_authorize("")) > > > > > > { > > > > > > www_challenge("", > > "0"); > > > > > > return; > > > > > > }; > > > > > > > > > > > > if(!save("location")) > > > > > > { > > > > > > sl_reply_error(); > > > > > > }; > > > > > > return; > > > > > > > > > > > > } > > > > > > else > > > > > > { > > > > > > } > > > > > > } > > > > > > > > > > > > Thanks in advance for your help. > > > > > > > > > > > > Sergio G. > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Users mailing list > > > > > > Users(a)openser.org <mailto: Users(a)openser.org > > > > > > <mailto: Users(a)openser.org <mailto: Users(a)openser.org>> > > > > <mailto: Users(a)openser.org <mailto: Users(a)openser.org> > > > <mailto: Users(a)openser.org <mailto: Users(a)openser.org>>> > > > > > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > < http://openser.org/cgi-bin/mailman/listinfo/users > > > <http://openser.org/cgi-bin/mailman/listinfo/users>> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >

3 12

[Users] rtpproxy redundancy
by Daniel Corbe 29 Mar '07

29 Mar '07

For various reasons we've elected to go with Porta/Maxim Sobalov's rtpproxy as opposed to the MediaProxy code; however, we would like to know if its possible to do any of the following with rtpproxy or if we should consider going to MediaProxy instead 1) Multiple Load Balanced rtpproxy servers 2) Voice path fail over, so that if we lose 1 RTP proxy the call continues on another. I was thinking 2 could be implemented using an OpenBSD/CARP FreeBSD/CARP type of scenario, where 2 rtpproxy servers share 1 IP address; however, OpenSER would need to set up an rtpproxy session on BOTH servers to accomplish this and the OS can do the rest. Any opinions or comments are welcome.

2 1

Fwd: [Users] SNMPstats module not returning data
by Sergio Gutierrez 29 Mar '07

29 Mar '07

Hi Jefrey Thanks for your answer. I will reply every point: # /opt/net-snmp/bin/snmpwalk -v 2c -c community localhost .1.3.6.1.4.1.27483 OPENSER-REG-MIB::openser = No Such Object available on this agent at this OID # ps -ef | grep snmpd root 1649 579 0 17:50:07 pts/2 0:17 /opt/net-snmp/sbin/snmpd -d -f -Dagentx -x tcp:localhost:705 root 2935 2778 0 09:59:21 pts/3 0:00 grep snmpd (I am running debug mode; the syntax of ps command is a little bti different because I am using Solaris) # grep -i agentx /var/log/snmpd.log agentx_build: packet built okay agentx/master: handle pdu (req=0x3ac55fdb,trans=0x0,sess=0xf) agentx/master: send response, stat 0 (req=0x3ac55fdb,trans=0x0,sess=0xf) agentx_build: packet built okay agentx/master: handle pdu (req=0x3ac55fdc,trans=0x0,sess=0xf) agentx/master: send response, stat 0 (req=0x3ac55fdc,trans=0x0,sess=0xf) agentx_build: packet built okay agentx/master: handle pdu (req=0x3ac55fdd,trans=0x0,sess=0xf) agentx/master: send response, stat 0 (req=0x3ac55fdd,trans=0x0,sess=0xf) agentx_build: packet built okay ... # cat /opt/net-snmp/share/snmp/snmpstats.conf agentXsocket tcp:localhost:705 # cat /opt/net-snmp/share/snmp/snmpd.conf syslocation XXXX syscontact XXXX agentaddress udp:161 rocommunity community localhost master agentx As you can see, both of the files are located at the same directory. Finally, I do have tried the check. I can see Openser starts OK and the master agent receives the connection of the sub agent. Thank you very much for your help. Kind regards. Sergio G. On 3/29/07, jmagder <jmagder(a)somanetworks.com > wrote: > > Hi Sergio, > > First off, could you try running "/opt/net-snmp/bin/snmpwalk -v 2c -c > community sipserver .1.3.6.1.4.1.27483"? If that isn't succeeding, > then it would greatly help me debug your problem if you could answer the > following questions: > > 1) What is the output of "ps aux | grep snmpd"? > > 2) What is the output of "cat /var/log/messages | grep AgentX" > > 3) Can you show me the contents of snmpstats.conf ? > > 4) Does snmpd.conf contain "master agentx"? > > 5) Is snmpd.conf located in the same directory as snmpstats.conf? > > 6) Have you tried the steps in "Testing for a proper Configuration" in > section 1.6.4? > > Thanks Sergio! > > - Jeffrey Magder > > Sergio Gutierrez wrote: > > Hi list. > > > > I have installed Openser 1.2.0, with Net-snmp version 5.4 to support > > the SNMPstats module. I have configured the subagent according to the > > instructions on the documentation, but I am not being able to obtain > > any data. > > > > I am running both products on Solaris 10 Sparc. > > > > # /opt/net-snmp/bin/snmpwalk -v 2c -c community sipserver > > .1.3.6.1.4.1.27483.3.1.1.1.1.1.1 > > SNMPv2-SMI::enterprises.27483.3.1.1.1.1.1 = No Such Object available > > on this agent at this OID > > > > # /opt/net-snmp/bin/snmpwalk -v 2c -c community sipserver > > openserSIPProtocolVersion > > SNMPv2-SMI::enterprises.27483.3.1.1.1.1.1 = No Such Object available > > on this agent at this OID > > > > Is there any other thing I should do after compiling and configuring? > > > > Thanks in advance for your attention > > > > Kind Regards. > > > > Sergio Gutiérrez > > > > > > > > ------------------------------------------------------------------------ > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

2 2

[Users] SNMPstats module not returning data
by Sergio Gutierrez 29 Mar '07

29 Mar '07

Hi list. I have installed Openser 1.2.0, with Net-snmp version 5.4 to support the SNMPstats module. I have configured the subagent according to the instructions on the documentation, but I am not being able to obtain any data. I am running both products on Solaris 10 Sparc. # /opt/net-snmp/bin/snmpwalk -v 2c -c community sipserver .1.3.6.1.4.1.27483.3.1.1.1.1.1.1 SNMPv2-SMI::enterprises.27483.3.1.1.1.1.1 = No Such Object available on this agent at this OID # /opt/net-snmp/bin/snmpwalk -v 2c -c community sipserver openserSIPProtocolVersion SNMPv2-SMI::enterprises.27483.3.1.1.1.1.1 = No Such Object available on this agent at this OID Is there any other thing I should do after compiling and configuring? Thanks in advance for your attention Kind Regards. Sergio Gutiérrez

3 2

[Users] how to check openser server is functional after initial setup
by Abdurrahman Sahin 29 Mar '07

29 Mar '07

hi; i just compiled openser on a machine with amd64 cpu and Fedora Core 5 as the OS. just set export CPU=athlon64. then make all is there any tool to test openser functionalities(seeing a process is running doesnt mean anything for me).? after successfull compilation i can see openser is listening tcp/udp 5060 ports. there are 3 executables in /usr/local/sbin directory -rwxr-xr-x 1 root root 2.9M Mar 29 09:37 openser -rwxr-xr-x 1 root root 41K Mar 29 09:37 openserctl -rwxr-xr-x 1 root root 15K Mar 29 09:37 openserunix and some configuration files in /usr/local/etc/openser -rw-r--r-- 1 root root 1.8K Mar 29 09:37 dictionary.radius -rw-r--r-- 1 root root 4.1K Mar 29 09:37 openser.cfg -rw-r--r-- 1 root root 1.2K Mar 29 09:37 openserctlrc netstat status tcp 0 0 192.168.200.2:5060 0.0.0.0:* LISTEN 31533/openser off (0.00/0/0) tcp 0 0 127.0.0.1:5060 0.0.0.0:* LISTEN 31533/openser off (0.00/0/0) udp 0 0 192.168.200.2:5060 0.0.0.0:* 31533/openser off (0.00/0/0) udp 0 0 127.0.0.1:5060 0.0.0.0:* 31533/openser off (0.00/0/0)

2 1

RE: [Users] Simple xcap-emulator for OpenSER
by Toni Barata 29 Mar '07

29 Mar '07

Hi Sukerry, Klaus, It was really a missing ";" after the "require_once('db_mysql.php')". Thanks, Toni -----Original Message----- From: sukerry [mailto:sukerry@126.com] Sent: quinta-feira, 29 de Março de 2007 0:41 To: Toni Barata; Klaus Darilion Cc: users openser.org Subject: [Spam?] Re: RE: [Users] Simple xcap-emulator for OpenSER Toni Barata, in Config.php, a ";" is missed before the term "class" ======= 2007-03-28 23:47:03 ======= >Hi Klaus, > > >I have been using xcap-0.2.1 without problems until yesterday. Today I have upgrade to the xcap-0.5 but the following error occurs: > >PHP Parse error: parse error, unexpected T_CLASS in /var/www/xcap-root/config.php on line 19 > >I have php4 on my system and the db_mysql.php file is php3 version. Could this be the problem? Should I downgrade php4 to php3? > >Thanks, >Toni > >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Klaus Darilion >Sent: quarta-feira, 28 de Mar?o de 2007 10:54 >To: sukerry >Cc: Klaus Darilion; Adrian Georgescu; users openser.org >Subject: Re: [Users] Simple xcap-emulator for OpenSER > >there is nothing to compile. Read the Readme. you have to install PEAR >and the modules mentioned in the READ. > >e.g. >pear install Auth_HTTP >pear install Log >pear install DB > >regards >klaus > > >sukerry wrote: >> Adrian Georgescu, >> >> >> Thanks for your great job! >> >> However , when I compile php, an error says that Log.php can not be found! >> >> Thanks >> >> sukerry >> >> >> >> ======= 2007-03-28 00:56:33======= >> >>> Hello, >>> >>> We wrote a simple xcap emulator to be able to manage the XCAP >>> documents used by OpenSER presence agent, by using PHP scripts in >>> combination with Apache web server. >>> >>> I have tested it against OpenSER with MySQL and Eyebeam for Apple. >>> Klaus Darilion has tested it against Postgress and Eyebeam for Windows. >>> >>> The software can be downloaded from: >>> >>> http://download.dns-hosting.info/XCAP/xcap-0.5.tar.gz >>> >>> Regards, >>> Adrian >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >> >> 　　　　　　　　sukerry >> 　　　　　　　　sukerry(a)126.com >> 　　　　　　　　　　2007-03-28 >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　sukerry 　　　　　　　　sukerry(a)126.com 　　　　　　　　　　2007-03-29

1 0

Re: [Users] ReInvite Issue
by Klaus Darilion 29 Mar '07

29 Mar '07

I think the problem is the gateway of bandwidth.com The 183 session progress have all the same To-tag and the same branch value in the topmost via header, although the 3 INVITEs sent to bandwidth.com have different branch value in the Via header. Thus, I suspect the dialog matching of the gateways is broken and it thinks the second and third INVITE is a retransmission of the first INVITE? Although I'm not sure as the gateway sends the 183 reply 3 times in a row. Strange. What happens in the PSTN? Which of the 3 phones ring? +17144828306 ? regards klaus Brandon Armstead wrote: > Hello, here is the sipcap: > > interface: eth0 (74.52.58.48/255.255.255.240) > filter: ip and ( port 5060 ) > match: 9492666205 > ################### > U +0.000000 ATA:5062 -> OpenSER:5060

1 0

[Users] OpenSER @ Von @ San Jose
by Bogdan-Andrei Iancu 29 Mar '07

29 Mar '07

Hi, OpenSER made a nice presence at Von Spring in San Jose. There were several presentation on the OpenSER topic, both in the expo area -in the Von Theatre- and in the conference sessions. For more, see: http://www.openser.org/events/2007-03-20-VoN-Spring/ There are other related presentations, but I'm still struggling to get them all. OpenSER drew and got a lot of attention - many people were aware of OpenSER and its potential, there were also quite some solutions (from expo area) using OpenSER and I personally met people using it for a large set of scenarios. And most important, I find more and more people are attracted by OpenSER as a viable solution for VoIP systems. As usual, VoN was an interesting experience. There is a changed of focus since last year - based on both the presented solution (expos) and on the conference topic, the interest shifted from IMS to "mobile VoIP" (fix mobile convergence). At least that was my personal impression and I find it most interesting as I think it reflects the real current needs (or requirements) of the VoIP solutions. Regards, Bogdan

1 0

[Serusers] SER + Prepaid with Radius AAA.
by Shafraz Thawfeek 29 Mar '07

29 Mar '07

Hello All, Its really feels nice to have joined the list. I'm in the process of deploying a SER based solution for prepaid users. We're using a FreeRadius based billing solution from a provider. If I'm not mistaken, SER is not aware of the call state, what this means to me is, in case the user account on the radius runs out of credit, SER will not know about it and will not be able to disconnect the call. Am I correct on this? If I am correct, I would like to know what would be the workaround for this? I'm I am wrong, then I would like to know on how we could get the call disconnection working? I have a Nextone which would be sitting in front of the SER and acting as a Mirror Proxy for SER. The purpose of this is to overcome NAT traversal issues and free SER from that. Nextone doesn't understand or talk radius. SER will be the registrar and handle AAA. The user call will be sent back into the Nextone and then terminated from there. SER will not be handling media. If to get my disconnect upon credit exhaust scenario working, what changes should I introduce into my existing network model? Thank you. Shaf.

3 9

[Users] G729 Codec question
by Andreas M. 29 Mar '07

29 Mar '07

Hello, i have a question regarding codec G729. We want to implement this in our client software, but there exists many flavours. Are all G.729 codec flavours compatible to each other? For example: G729 to G729A or B ? or G729B to A and vice versa ? We are currently using Cisco AS5300 Gateways, that supports G729 and G729B. But we want to change our gateways, then only G729A is available. i know this question has nothing to do with openser, but i think there are many readers with experience. best regards, Andreas M.

2 1

Re: [Users] ReInvite Issue
by Brandon Armstead 29 Mar '07

29 Mar '07

Hello, here is the sipcap below: match: 9492666205 ################### U +0.538693 GATEWAY_1:5060 -> OpenSER:5060 INVITE sip:9492666205@OpenSER:5060 SIP/2.0. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Contact: <sip:7135178240@64.24.35.73>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. User-Agent: US LEC VOIP. Max-Forwards: 69. Date: Thu, 29 Mar 2007 09:46:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 156. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 64.24.35.73. t=0 0. m=audio 31806 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U +0.002135 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK49bd.5c8e4a8.0;rport=5060. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Server: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. Warning: 392 OpenSER:5060 "Noisy feedback tells: pid=15173 req_src_ip=GATEWAY_1 req_src_port=5060 in_uri=sip:9492666205@OpenSER:5060 out_uri=sip:9492666205@OpenSER:5060 via_cnt==2". . # U +0.000088 OpenSER:5060 -> GATEWAY_2:5060 INVITE sip:+17144828306@GATEWAY_2:5060 SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Contact: <sip:7135178240@64.24.35.73>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Max-Forwards: 68. Date: Thu, 29 Mar 2007 09:46:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 174. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 OpenSER. t=0 0. m=audio 37158 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=nortpproxy:yes. # U +0.000015 OpenSER:5060 -> GATEWAY_2:5060 INVITE sip:+17608263100@GATEWAY_2:5060 SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Contact: <sip:7135178240@64.24.35.73>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Max-Forwards: 68. Date: Thu, 29 Mar 2007 09:46:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 174. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 OpenSER. t=0 0. m=audio 37158 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=nortpproxy:yes. # U +0.000019 OpenSER:5060 -> GATEWAY_2:5060 INVITE sip:+19496164708@GATEWAY_2:5060 SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Contact: <sip:7135178240@64.24.35.73>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Max-Forwards: 68. Date: Thu, 29 Mar 2007 09:46:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 174. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 OpenSER. t=0 0. m=audio 37158 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=nortpproxy:yes. # U +0.022478 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 100 Trying bandwidth.com. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32282 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+17144828306@GATEWAY_2:5060 out_uri=sip:+17144828306@4.68.250.148:5060 via_cnt==3". . # U +0.000376 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 100 Trying bandwidth.com. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32282 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+17608263100@GATEWAY_2:5060 out_uri=sip:+17608263100@4.68.250.148:5060 via_cnt==3". . # U +0.000124 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 100 Trying bandwidth.com. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32282 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+19496164708@GATEWAY_2:5060 out_uri=sip:+19496164708@4.68.250.148:5060 via_cnt==3". . ####### U +0.110379 OpenSER:5060 -> ATA:5062 INVITE sip:9492666205@192.168.1.126:5062 SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "Cell Phone TX" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>. Contact: <sip:7135178240@OpenSER:5060>. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Max-Forwards: 67. Date: Thu, 29 Mar 2007 09:46:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 174. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 OpenSER. t=0 0. m=audio 37158 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=nortpproxy:yes. # U +0.035038 ATA:5062 -> OpenSER:5060 SIP/2.0 100 Trying. To: <sip:9492666205@GATEWAY_1>. From: "Cell Phone TX" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Server: Sipura/SPA841-3.1.4(a). Content-Length: 0. . # U +0.035106 ATA:5062 -> OpenSER:5060 SIP/2.0 180 Ringing. To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. From: "Cell Phone TX" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Server: Sipura/SPA841-3.1.4(a). Content-Length: 0. . # U +0.000287 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 180 Ringing. To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. From: "Cell Phone TX" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Server: Sipura/SPA841-3.1.4(a). Content-Length: 0. . ######################################## U +6.689904 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.000286 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. # U +0.139883 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.000251 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.003394 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. # U +0.000090 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. ####### U +4.012603 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.000126 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.007244 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. # U +0.000041 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. ## U +0.575203 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.000125 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.003655 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. # U +0.000041 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. ################### U +4.011642 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.000377 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.000192 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. # U +0.011882 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. ############### U +4.003638 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 209.244.42.237. t=0 0. m=audio 64520 RTP/AVP 0. a=ptime:20. # U +0.008482 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175161585 1175161586 IN IP4 209.244.42.237. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 37160 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. ######################################################### U +2.220013 OpenSER:5060 -> GATEWAY_2:5060 CANCEL sip:+17608263100@GATEWAY_2:5060 SIP/2.0. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>. CSeq: 102 CANCEL. User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. . # U +0.000015 OpenSER:5060 -> GATEWAY_2:5060 CANCEL sip:+19496164708@GATEWAY_2:5060 SIP/2.0. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>. CSeq: 102 CANCEL. User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. . # U +0.008073 OpenSER:5060 -> ATA:5062 CANCEL sip:9492666205@192.168.1.126:5062 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>. CSeq: 102 CANCEL. Content-Length: 0. . # U +0.023630 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-91e7. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32286 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+17608263100@GATEWAY_2:5060 out_uri=sip:+17608263100@4.68.250.148:5060 via_cnt==1". . # U +0.000096 OpenSER:5060 -> GATEWAY_2:5060 ACK sip:+17608263100@GATEWAY_2:5060 SIP/2.0. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-91e7. CSeq: 102 ACK. User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. . # U +0.005775 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-09f6. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32277 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+19496164708@GATEWAY_2:5060 out_uri=sip:+19496164708@4.68.250.148:5060 via_cnt==1". . # U +0.000089 OpenSER:5060 -> GATEWAY_2:5060 ACK sip:+19496164708@GATEWAY_2:5060 SIP/2.0. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-09f6. CSeq: 102 ACK. User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. . # U +0.012529 ATA:5062 -> OpenSER:5060 SIP/2.0 487 Request Terminated. To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. From: "Cell Phone TX" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 INVITE. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. Server: Sipura/SPA841-3.1.4(a). Content-Length: 0. . # U +0.000074 OpenSER:5060 -> ATA:5062 ACK sip:9492666205@192.168.1.126:5062 SIP/2.0. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. CSeq: 102 ACK. User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. . # U +0.003301 ATA:5062 -> OpenSER:5060 SIP/2.0 200 OK. To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. CSeq: 102 CANCEL. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. Server: Sipura/SPA841-3.1.4(a). Content-Length: 0. . # U +0.002123 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 200 ok -- no more pending branches. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-91e7. CSeq: 102 CANCEL. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32286 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+17608263100@GATEWAY_2:5060 out_uri=sip:+17608263100@GATEWAY_2:5060 via_cnt==1". . # U +0.005996 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 200 ok -- no more pending branches. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-09f6. CSeq: 102 CANCEL. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32277 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+19496164708@GATEWAY_2:5060 out_uri=sip:+19496164708@GATEWAY_2:5060 via_cnt==1". . ########exit 450 received, 0 dropped On 3/29/07, Brandon Armstead <brandon.armstead(a)gmail.com> wrote: > > Hello, here is the sipcap: > > interface: eth0 (74.52.58.48/255.255.255.240) > filter: ip and ( port 5060 ) > match: 9492666205 > ################### > U +0.000000 ATA:5062 -> OpenSER:5060 > NOTIFY sip:demo.demo.com SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.126:5062;branch=z9hG4bK-73d5713f. > From: "Brandon Armstead" < sip:9492666205@demo.demo.com > >;tag=cb22d7a7f754d0o0. > To: <sip:demo.demo.com>. > Call-ID: 69b08a3a-8a5e9f10(a)192.168.1.126 . > CSeq: 176 NOTIFY. > Max-Forwards: 70. > Event: keep-alive. > User-Agent: Sipura/SPA841-3.1.4(a). > Content-Length: 0. > . > > # > U +0.000125 OpenSER:5060 -> ATA:5062 > SIP/2.0 200 ok. > Via: SIP/2.0/UDP 192.168.1.126:5062 > ;branch=z9hG4bK-73d5713f;rport=5062;received=ATA. > From: "Brandon Armstead" <sip:9492666205@demo.demo.com > >;tag=cb22d7a7f754d0o0. > To: <sip:demo.demo.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.a9e4. > Call-ID: 69b08a3a-8a5e9f10(a)192.168.1.126 . > CSeq: 176 NOTIFY. > Server: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > Warning: 392 OpenSER:5060 "Noisy feedback tells: pid=15173 req_src_ip=ATA > req_src_port=5062 in_uri=sip: demo.demo.com out_uri=sip:demo.demo.comvia_cnt==1". > . > > ### > U +0.538693 GATEWAY_1:5060 -> OpenSER:5060 > INVITE sip:9492666205@OpenSER:5060 SIP/2.0. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Contact: <sip:7135178240@64.24.35.73>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > User-Agent: US LEC VOIP. > Max-Forwards: 69. > Date: Thu, 29 Mar 2007 09:46:26 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Content-Type: application/sdp. > Content-Length: 156. > . > v=0. > o=root 7184 7184 IN IP4 64.24.35.73. > s=session. > c=IN IP4 64.24.35.73. > t=0 0. > m=audio 31806 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=silenceSupp:off - - - -. > > # > U +0.002135 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP GATEWAY_1;branch= z9hG4bK49bd.5c8e4a8.0;rport=5060. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Server: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > Warning: 392 OpenSER:5060 "Noisy feedback tells: pid=15173 > req_src_ip=GATEWAY_1 req_src_port=5060 in_uri=sip:9492666205@OpenSER:5060 > out_uri= sip:9492666205@OpenSER:5060 via_cnt==2". > . > > # > U +0.000088 OpenSER:5060 -> GATEWAY_2:5060 > INVITE sip:+17144828306@GATEWAY_2:5060 SIP/2.0. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Contact: <sip:7135178240@64.24.35.73>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Max-Forwards: 68. > Date: Thu, 29 Mar 2007 09:46:26 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Content-Type: application/sdp. > Content-Length: 174. > . > v=0. > o=root 7184 7184 IN IP4 64.24.35.73. > s=session. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37158 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=silenceSupp:off - - - -. > a=nortpproxy:yes. > > # > U +0.000015 OpenSER:5060 -> GATEWAY_2:5060 > INVITE sip:+17608263100@GATEWAY_2:5060 SIP/2.0. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Contact: < sip:7135178240@64.24.35.73>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Max-Forwards: 68. > Date: Thu, 29 Mar 2007 09:46:26 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Content-Type: application/sdp. > Content-Length: 174. > . > v=0. > o=root 7184 7184 IN IP4 64.24.35.73. > s=session. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37158 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=silenceSupp:off - - - -. > a=nortpproxy:yes. > > # > U +0.000019 OpenSER:5060 -> GATEWAY_2:5060 > INVITE sip:+19496164708@GATEWAY _2:5060 SIP/2.0. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch= z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" <sip:7135178240@64.24.35.73 >;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Contact: <sip:7135178240@64.24.35.73>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Max-Forwards: 68. > Date: Thu, 29 Mar 2007 09:46:26 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Content-Type: application/sdp. > Content-Length: 174. > . > v=0. > o=root 7184 7184 IN IP4 64.24.35.73. > s=session. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37158 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=silenceSupp:off - - - -. > a=nortpproxy:yes. > > # > U +0.022478 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 100 Trying bandwidth.com. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch= z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" <sip:7135178240@64.24.35.73 >;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Server: Bandwidth.com SDE (v1.0). > Content-Length: 0. > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32282 > req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+17144828306@GATEWAY_2:5060 out_uri= > sip:+17144828306@4.68.250.148:5060 via_cnt==3". > . > > # > U +0.000376 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 100 Trying bandwidth.com. > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK49bd.98a53e2.2. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Server: Bandwidth.com SDE (v1.0). > Content-Length: 0. > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32282 > req_src_ip=OpenSER req_src_port=5060 in_uri= sip:+17608263100@GATEWAY_2:5060 > out_uri=sip:+17608263100@4.68.250.148:5060 via_cnt==3". > . > > # > U +0.000124 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 100 Trying bandwidth.com . > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Server: Bandwidth.com SDE (v1.0). > Content-Length: 0. > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32282 > req_src_ip=OpenSER req_src_port=5060 in_uri= sip:+19496164708@GATEWAY_2:5060 > out_uri=sip:+19496164708@4.68.250.148:5060 via_cnt==3". > . > > ####### > U +0.110379 OpenSER:5060 -> ATA:5062 > INVITE sip:9492666205@192.168.1.126:5062 SIP/2.0. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. > Via: SIP/2.0/UDP OpenSER;rport=5060;branch= z9hG4bK49bd.98a53e2.0. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "Cell Phone TX" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>. > Contact: <sip:7135178240@OpenSER:5060>. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Max-Forwards: 67. > Date: Thu, 29 Mar 2007 09:46:26 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Content-Type: application/sdp. > Content-Length: 174. > . > v=0. > o=root 7184 7184 IN IP4 64.24.35.73. > s=session. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37158 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=silenceSupp:off - - - -. > a=nortpproxy:yes. > > # > U +0.035038 ATA:5062 -> OpenSER:5060 > SIP/2.0 100 Trying. > To: <sip:9492666205@GATEWAY_1>. > From: "Cell Phone TX" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK49bd.a8a53e2.0. > Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060 ;branch=z9hG4bK529c9661;rport=5060. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Server: Sipura/SPA841-3.1.4(a). > Content-Length: 0. > . > > # > U +0.035106 ATA:5062 -> OpenSER:5060 > SIP/2.0 180 Ringing. > To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. > From: "Cell Phone TX" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. > Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Server: Sipura/SPA841-3.1.4(a). > Content-Length: 0. > . > > # > U +0.000287 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 180 Ringing. > To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. > From: "Cell Phone TX" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Server: Sipura/SPA841-3.1.4(a). > Content-Length: 0. > . > > ######################################## > U +6.689904 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.000286 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > # > U +0.139883 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.000251 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.003394 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > # > U +0.000090 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > ####### > U +4.012603 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.000126 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.007244 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > # > U +0.000041 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > ########## > U +3.433005 ATA:5062 -> OpenSER:5060 > NOTIFY sip: demo.demo.com SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.126:5062;branch=z9hG4bK-689dad84. > From: "Brandon Armstead" < sip:9492666205@demo.demo.com > >;tag=cb22d7a7f754d0o0. > To: <sip:demo.demo.com>. > Call-ID: 69b08a3a-8a5e9f10(a)192.168.1.126 . > CSeq: 177 NOTIFY. > Max-Forwards: 70. > Event: keep-alive. > User-Agent: Sipura/SPA841-3.1.4(a). > Content-Length: 0. > . > > # > U +0.000095 OpenSER:5060 -> ATA:5062 > SIP/2.0 200 ok. > Via: SIP/2.0/UDP 192.168.1.126:5062 > ;branch=z9hG4bK-689dad84;rport=5062;received=ATA. > From: "Brandon Armstead" <sip:9492666205@demo.demo.com > >;tag=cb22d7a7f754d0o0. > To: <sip:demo.demo.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.2d30. > Call-ID: 69b08a3a-8a5e9f10(a)192.168.1.126 . > CSeq: 177 NOTIFY. > Server: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > Warning: 392 OpenSER:5060 "Noisy feedback tells: pid=15175 req_src_ip=ATA > req_src_port=5062 in_uri=sip: demo.demo.com out_uri=sip:demo.demo.comvia_cnt==1". > . > > ## > U +0.575203 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.000125 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.003655 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > # > U +0.000041 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > ################### > U +4.011642 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.000377 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.000192 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > # > U +0.011882 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > ############### > U +4.003638 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.1,SIP/2.0/UDP > GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 128. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 209.244.42.237. > t=0 0. > m=audio 64520 RTP/AVP 0. > a=ptime:20. > > # > U +0.008482 OpenSER:5060 -> GATEWAY_1:5060 > SIP/2.0 183 Session Progress. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0,SIP/2.0/UDP > 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > Record-Route: > <sip:GATEWAY_2;lr;ftag=as0f6730bc>,<sip:OpenSER;lr=on;ftag=as0f6730bc>,<sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=VPST506071629460. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Contact: <sip:+17144828306@4.68.250.148:5060;transport=udp>. > Content-Type: application/sdp. > Content-Length: 143. > . > v=0. > o=- 1175161585 1175161586 IN IP4 209.244.42.237. > s=-. > c=IN IP4 OpenSER. > t=0 0. > m=audio 37160 RTP/AVP 0. > a=ptime:20. > a=nortpproxy:yes. > > ######################################################### > U +2.220013 OpenSER:5060 -> GATEWAY_2:5060 > CANCEL sip:+17608263100@GATEWAY_2:5060 SIP/2.0. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > To: <sip:9492666205@GATEWAY_1>. > CSeq: 102 CANCEL. > User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > . > > # > U +0.000015 OpenSER:5060 -> GATEWAY_2:5060 > CANCEL sip:+19496164708@GATEWAY_2:5060 SIP/2.0. > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK49bd.98a53e2.3. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > To: <sip:9492666205@GATEWAY_1>. > CSeq: 102 CANCEL. > User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > . > > # > U +0.008073 OpenSER:5060 -> ATA:5062 > CANCEL sip:9492666205@192.168.1.126:5062 SIP/2.0. > Max-Forwards: 10. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. > Via: SIP/2.0/UDP OpenSER;rport=5060;branch= z9hG4bK49bd.98a53e2.0. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > To: <sip:9492666205@GATEWAY_1>. > CSeq: 102 CANCEL. > Content-Length: 0. > . > > # > U +0.023630 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 487 Request Terminated. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-91e7. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Server: Bandwidth.com SDE (v1.0). > Content-Length: 0. > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32286 > req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+17608263100@GATEWAY_2:5060 > out_uri=sip:+17608263100@4.68.250.148:5060 via_cnt==1". > . > > # > U +0.000096 OpenSER:5060 -> GATEWAY_2:5060 > ACK sip:+17608263100@GATEWAY_2:5060 SIP/2.0. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-91e7. > CSeq: 102 ACK. > User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > . > > # > U +0.005775 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 487 Request Terminated. > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK49bd.98a53e2.3. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK529c9661;rport=5060. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-09f6. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Server: Bandwidth.com SDE (v1.0). > Content-Length: 0. > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32277 > req_src_ip=OpenSER req_src_port=5060 in_uri= sip:+19496164708@GATEWAY_2:5060 > out_uri=sip:+19496164708@4.68.250.148:5060 via_cnt==1". > . > > # > U +0.000089 OpenSER:5060 -> GATEWAY_2:5060 > ACK sip:+19496164708@GATEWAY_2:5060 SIP/2.0. > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK49bd.98a53e2.3. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-09f6. > CSeq: 102 ACK. > User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > . > > # > U +0.012529 ATA:5062 -> OpenSER:5060 > SIP/2.0 487 Request Terminated. > To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. > From: "Cell Phone TX" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 INVITE. > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK49bd.a8a53e2.0. > Via: SIP/2.0/UDP OpenSER;rport=5060;branch=z9hG4bK49bd.98a53e2.0. > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK49bd.5c8e4a8.0. > Via: SIP/2.0/UDP 64.24.35.73:5060 ;branch=z9hG4bK529c9661;rport=5060. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as0f6730bc>. > Server: Sipura/SPA841-3.1.4(a). > Content-Length: 0. > . > > # > U +0.000074 OpenSER:5060 -> ATA:5062 > ACK sip:9492666205@192.168.1.126:5062 SIP/2.0. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0 . > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73 . > To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. > CSeq: 102 ACK. > User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). > Content-Length: 0. > . > > # > U +0.003301 ATA:5062 -> OpenSER:5060 > SIP/2.0 200 OK. > To: <sip:9492666205@GATEWAY_1>;tag=cf04e5d3f5afda4fi0. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > CSeq: 102 CANCEL. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.a8a53e2.0. > Via: SIP/2.0/UDP OpenSER;rport=5060;branch= z9hG4bK49bd.98a53e2.0. > Record-Route: <sip:OpenSER;lr=on;ftag=as0f6730bc>. > Server: Sipura/SPA841-3.1.4(a). > Content-Length: 0. > . > > # > U +0.002123 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 200 ok -- no more pending branches. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.2. > From: "7135178240" <sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-91e7. > CSeq: 102 CANCEL. > Server: Bandwidth.com SDE (v1.0 ). > Content-Length: 0. > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32286 > req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+17608263100@GATEWAY_2:5060 > out_uri=sip:+17608263100@GATEWAY_2:5060 via_cnt==1". > . > > # > U +0.005996 GATEWAY_2:5060 -> OpenSER:5060 > SIP/2.0 200 ok -- no more pending branches. > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK49bd.98a53e2.3. > From: "7135178240" < sip:7135178240@64.24.35.73>;tag=as0f6730bc. > Call-ID: 132193751d2413ea41f9f02e378ec608(a)64.24.35.73. > To: <sip:9492666205@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-09f6. > CSeq: 102 CANCEL. > Server: Bandwidth.com SDE (v1.0). > Content-Length: 0. > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=32277 > req_src_ip=OpenSER req_src_port=5060 in_uri= sip:+19496164708@GATEWAY_2:5060 > out_uri=sip:+19496164708@GATEWAY_2:5060 via_cnt==1". > . > > ########exit > 450 received, 0 dropped > > > On 3/29/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > > > > > > > Brandon Armstead wrote: > > > Hello Klaus, > > > Any ideas, > > > > > > not without a SIP dump (ngrep port 5060) > > > > btw: please Cc: the list > > > > regards > > klaus > > > > > > > > > > > > thanks! > > > > > > On 3/29/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > >> > > >> > > >> > > >> Brandon Armstead wrote: > > >> > Hello everyone, > > >> > > > >> > I'm running into a issue, where I'm trying to implement parallel > > >> > forking, so basically on incoming calls from gateway, I want to > > then > > >> > Re-Invite, and send out the calls to multiple destinations, I am > > >> currently, > > >> > > >> FYI: The term "Re-INVITE" is wrong in this scenario. The proxy just > > >> forwards the INVITE doing parallel forking (in openser terminology: > > >> creates multiple branches). As sson as one phone picks up, this > > branch > > >> is the winning branch and the other branches will be canceled. > > >> > > >> A reINVITE is an in-dialog INVITE, e.g. used for modifying the media > > >> session (putting the call on hold, enable/disable video ...) > > >> > > >> regards > > >> klaus > > >> > > >> > using avp_db_query to grab the sip uri's and throwing them into an > > avp > > >> > list, > > >> > and avp_pushto("$ru", "$avp(i:1)/g"); This all works fine and > > good, > > >> invite > > >> > comes in, and then sends Re-Invites, however, there seems to be > > some > > >> > kind of > > >> > race condition where, it is not properly processing all of the > > >> Re-Invites, > > >> > to fork the calls out, and basically one of the calls picks up 183 > > >> early > > >> > media, then 200 OK, and terminates the other branches / > > calls. However > > >> > they > > >> > were never executed / no phone was ever picked up. Any ideas on > > what > > >> I'm > > >> > doing wrong, or what could possibly be happening? Any help is > > >> appreciated, > > >> > thank you. > > >> > > > >> > I am using OpenSER v1.1.1, example sipcap posted below: > > >> > > > >> > match: CALLEE > > >> > ################# > > >> > U +0.000000 GATEWAY_1:5060 -> OpenSER:5060 > > >> > INVITE sip:CALLEE@OpenSER:5060 SIP/2.0. > > >> > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as17d625b5>. > > >> > Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK0b52.a5ddb76.0. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK5dce8d77;rport=5060. > > >> > From: "CALLER" < sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>. > > >> > Contact: <sip:CALLER@64.24.35.73 >. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > User-Agent: US LEC VOIP. > > >> > Max-Forwards: 69. > > >> > Date: Wed, 28 Mar 2007 13:58:45 GMT. > > >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > >> > Content-Type: application/sdp. > > >> > Content-Length: 156. > > >> > . > > >> > v=0. > > >> > o=root 7184 7184 IN IP4 64.24.35.73. > > >> > s=session. > > >> > c=IN IP4 64.24.35.73. > > >> > t=0 0. > > >> > m=audio 17990 RTP/AVP 0. > > >> > a=rtpmap:0 PCMU/8000. > > >> > a=silenceSupp:off - - - -. > > >> > > > >> > # > > >> > U +0.002597 OpenSER:5060 -> GATEWAY_1:5060 > > >> > SIP/2.0 100 trying -- your call is important to us. > > >> > Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK0b52.a5ddb76.0;rport=5060. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK5dce8d77;rport=5060. > > >> > From: "CALLER" <sip:CALLER@64.24.35.73 >;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Server: OpenSer (1.1.1-notls (x86_64/linux)). > > >> > Content-Length: 0. > > >> > Warning: 392 OpenSER:5060 "Noisy feedback tells: pid=10278 > > >> > req_src_ip=GATEWAY_1 req_src_port=5060 in_uri= sip:CALLEE@OpenSER > > :5060 > > >> > out_uri=sip:+RE-INVITE_CALLEE_1@GATEWAY_2:5060 via_cnt==2". > > >> > . > > >> > > > >> > # > > >> > U +0.000090 OpenSER:5060 -> GATEWAY_2:5060 > > >> > INVITE sip:+RE-INVITE_CALLEE_1@GATEWAY_2:5060 SIP/2.0. > > >> > Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. > > >> > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as17d625b5>. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0. > > >> > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK5dce8d77;rport=5060. > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY _1>. > > >> > Contact: <sip:CALLER@64.24.35.73>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73 . > > >> > CSeq: 102 INVITE. > > >> > Max-Forwards: 68. > > >> > Date: Wed, 28 Mar 2007 13:58:45 GMT. > > >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > >> > Content-Type: application/sdp. > > >> > Content-Length: 174. > > >> > . > > >> > v=0. > > >> > o=root 7184 7184 IN IP4 64.24.35.73. > > >> > s=session. > > >> > c=IN IP4 OpenSER. > > >> > t=0 0. > > >> > m=audio 35998 RTP/AVP 0. > > >> > a=rtpmap:0 PCMU/8000. > > >> > a=silenceSupp:off - - - -. > > >> > a=nortpproxy:yes. > > >> > > > >> > # > > >> > U +0.000023 OpenSER:5060 -> GATEWAY_2:5060 > > >> > INVITE sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 SIP/2.0. > > >> > Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. > > >> > Record-Route: <sip:GATEWAY_1;lr=on;ftag=as17d625b5>. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. > > >> > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK5dce8d77;rport=5060. > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY _1>. > > >> > Contact: <sip:CALLER@64.24.35.73>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73 . > > >> > CSeq: 102 INVITE. > > >> > Max-Forwards: 68. > > >> > Date: Wed, 28 Mar 2007 13:58:45 GMT. > > >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > >> > Content-Type: application/sdp. > > >> > Content-Length: 174. > > >> > . > > >> > v=0. > > >> > o=root 7184 7184 IN IP4 64.24.35.73. > > >> > s=session. > > >> > c=IN IP4 OpenSER. > > >> > t=0 0. > > >> > m=audio 35998 RTP/AVP 0. > > >> > a=rtpmap:0 PCMU/8000. > > >> > a=silenceSupp:off - - - -. > > >> > a=nortpproxy:yes. > > >> > > > >> > # > > >> > U +0.022891 GATEWAY_2:5060 -> OpenSER:5060 > > >> > SIP/2.0 100 Trying bandwidth.com. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0. > > >> > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK5dce8d77;rport=5060. > > >> > From: "CALLER" < sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Server: Bandwidth.com SDE (v1.0). > > >> > Content-Length: 0. > > >> > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13590 > > >> > req_src_ip=OpenSER req_src_port=5060 > > >> > in_uri=sip:+RE-INVITE_CALLEE_1@GATEWAY_2:5060 > > >> > out_uri=sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060 via_cnt==3". > > >> > . > > >> > > > >> > # > > >> > U +0.000007 GATEWAY_2:5060 -> OpenSER:5060 > > >> > SIP/2.0 100 Trying bandwidth.com. > > >> > Via: SIP/2.0/UDP OpenSER;branch= z9hG4bK0b52.7164f3d6.1. > > >> > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. > > > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Server: Bandwidth.com SDE (v1.0). > > >> > Content-Length: 0. > > >> > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13590 > > >> > req_src_ip=OpenSER req_src_port=5060 > > >> > in_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 > > >> > out_uri=sip:+RE-INVITE_CALLEE_2@4.68.250.148:5060 via_cnt==3". > > >> > . > > >> > > > >> > ############## > > >> > U +2.054324 GATEWAY_2:5060 -> OpenSER:5060 > > >> > SIP/2.0 183 Session Progress. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0 ,SIP/2.0/UDP > > >> > GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP > > >> > 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. > > >> > Record-Route: > > >> > > > >> > > <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. > > >> > > >> > > > >> > From: "CALLER" < sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. > > >> > Content-Type: application/sdp. > > >> > Content-Length: 128. > > >> > . > > >> > v=0. > > >> > o=- 1175090325 1175090326 IN IP4 209.244.42.250. > > >> > s=-. > > >> > c=IN IP4 209.244.42.250. > > >> > t=0 0. > > >> > m=audio 61138 RTP/AVP 0. > > >> > a=ptime:20. > > >> > > > >> > # > > >> > U +0.000244 OpenSER:5060 -> GATEWAY_1:5060 > > >> > SIP/2.0 183 Session Progress. > > >> > Via: SIP/2.0/UDP > > >> > GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP > > >> > 64.24.35.73:5060 ;branch=z9hG4bK5dce8d77;rport=5060. > > >> > Record-Route: > > >> > > > >> > > <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. > > > > >> > > >> > > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. > > >> > Content-Type: application/sdp. > > >> > Content-Length: 143. > > >> > . > > >> > v=0. > > >> > o=- 1175090325 1175090326 IN IP4 209.244.42.250. > > >> > s=-. > > >> > c=IN IP4 OpenSER. > > >> > t=0 0. > > >> > m=audio 36002 RTP/AVP 0. > > >> > a=ptime:20. > > >> > a=nortpproxy:yes. > > >> > > > >> > ## > > >> > U +0.213133 GATEWAY_2:5060 -> OpenSER:5060 > > >> > SIP/2.0 200 OK. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0,SIP/2.0/UDP > > >> > GATEWAY_1;rport=5060;branch= z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP > > >> > 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. > > >> > Record-Route: > > >> > > > >> > > <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. > > > > >> > > >> > > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. > > >> > Content-Type: application/sdp. > > >> > Content-Length: 128. > > >> > . > > >> > v=0. > > >> > o=- 1175090325 1175090326 IN IP4 209.244.42.250. > > >> > s=-. > > >> > c=IN IP4 209.244.42.250. > > >> > t=0 0. > > >> > m=audio 61138 RTP/AVP 0. > > >> > a=ptime:20. > > >> > > > >> > # > > >> > U +0.004792 OpenSER:5060 -> GATEWAY_1:5060 > > >> > SIP/2.0 200 OK. > > >> > Via: SIP/2.0/UDP > > >> > GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP > > >> > 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. > > >> > Record-Route: > > >> > > > >> > > <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. > > > > >> > > >> > > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. > > >> > Content-Type: application/sdp. > > >> > Content-Length: 143. > > >> > . > > >> > v=0. > > >> > o=- 1175090325 1175090326 IN IP4 209.244.42.250. > > >> > s=-. > > >> > c=IN IP4 OpenSER. > > >> > t=0 0. > > >> > m=audio 36002 RTP/AVP 0. > > >> > a=ptime:20. > > >> > a=nortpproxy:yes. > > >> > > > >> > # > > >> > U +0.000146 OpenSER:5060 -> GATEWAY_2:5060 > > >> > CANCEL sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 SIP/2.0. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1 . > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > To: <sip:CALLEE@GATEWAY_1>. > > >> > CSeq: 102 CANCEL. > > >> > User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). > > >> > Content-Length: 0. > > >> > . > > >> > > > >> > # > > >> > U +0.028419 GATEWAY_2:5060 -> OpenSER:5060 > > >> > SIP/2.0 487 Request Terminated. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1 . > > >> > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK5dce8d77;rport=5060. > > >> > From: "CALLER" < sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY > > _1>;tag=2a6c315af837f0a6a989449d0bd9acf6-fcd9. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 INVITE. > > >> > Server: Bandwidth.com SDE (v1.0). > > >> > Content-Length: 0. > > >> > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13600 > > >> > req_src_ip=OpenSER req_src_port=5060 > > >> > in_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 > > >> > out_uri=sip:+RE-INVITE_CALLEE_2@4.68.250.148:5060 via_cnt==1". > > >> > . > > >> > > > >> > # > > >> > U +0.000098 OpenSER:5060 -> GATEWAY_2:5060 > > >> > ACK sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 SIP/2.0. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1 . > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > To: <sip:CALLEE@GATEWAY > > _1>;tag=2a6c315af837f0a6a989449d0bd9acf6-fcd9. > > >> > CSeq: 102 ACK. > > >> > User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). > > >> > Content-Length: 0. > > >> > . > > >> > > > >> > # > > >> > U +0.011645 GATEWAY_2:5060 -> OpenSER:5060 > > >> > SIP/2.0 200 ok -- no more pending branches. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > To: <sip:CALLEE@GATEWAY > > _1>;tag=2a6c315af837f0a6a989449d0bd9acf6-fcd9. > > >> > CSeq: 102 CANCEL. > > >> > Server: Bandwidth.com SDE (v1.0). > > >> > Content-Length: 0. > > >> > Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13600 > > >> > req_src_ip=OpenSER req_src_port=5060 > > >> > in_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 > > >> > out_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 via_cnt==1". > > >> > . > > >> > > > >> > # > > >> > U +0.030857 GATEWAY_1:5060 -> OpenSER:5060 > > >> > ACK sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp > > SIP/2.0. > > >> > Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK0b52.a5ddb76.2. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK29c21d5a;rport=5060. > > >> > Route: > > >> > > > <sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_2;lr;ftag=as17d625b5>. > > >> > From: "CALLER" < sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Contact: < sip:CALLER@64.24.35.73>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 ACK. > > >> > User-Agent: US LEC VOIP. > > >> > Max-Forwards: 69. > > >> > Content-Length: 0. > > >> > . > > >> > > > >> > # > > >> > U +0.000183 OpenSER:5060 -> GATEWAY_2:5060 > > >> > ACK sip:+RE-INVITE_CALLEE_1@4.68.250.148 :5060;transport=udp > > SIP/2.0. > > >> > Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.3. > > >> > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch= z9hG4bK0b52.a5ddb76.2 > > . > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK29c21d5a;rport=5060. > > >> > Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>. > > >> > From: "CALLER" < sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Contact: < sip:CALLER@64.24.35.73>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 102 ACK. > > >> > Max-Forwards: 68. > > >> > Content-Length: 0. > > >> > P-hint: rr-enforced. > > >> > . > > >> > > > >> > > > >> > > ##################################################################################################################################################################################################################################################################################################################################################################################################################################################################### > > > > >> > > >> > > > >> > U +71.203150 GATEWAY_1:5060 -> OpenSER:5060 > > >> > BYE sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp > > SIP/2.0. > > >> > Via: SIP/2.0/UDP GATEWAY_1;branch= z9hG4bK1b52.5ebf0565.0. > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK53c03317;rport=5060. > > >> > Route: > > >> > > > <sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_2;lr;ftag=as17d625b5>. > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Contact: < sip:CALLER@64.24.35.73>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 103 BYE. > > >> > User-Agent: US LEC VOIP. > > >> > Max-Forwards: 69. > > >> > Content-Length: 0. > > >> > . > > >> > > > >> > # > > >> > U +0.003835 OpenSER:5060 -> GATEWAY_2:5060 > > >> > BYE sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp > > SIP/2.0. > > >> > Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK1b52.25391854.0 . > > >> > Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK1b52.5ebf0565.0 > > . > > >> > Via: SIP/2.0/UDP 64.24.35.73:5060 > > ;branch=z9hG4bK53c03317;rport=5060. > > >> > Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>. > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Contact: < sip:CALLER@64.24.35.73>. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 103 BYE. > > >> > Max-Forwards: 68. > > >> > Content-Length: 0. > > >> > P-hint: rr-enforced. > > >> > . > > >> > > > >> > # > > >> > U +0.073996 GATEWAY_2:5060 -> OpenSER:5060 > > >> > SIP/2.0 200 OK. > > >> > Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK1b52.25391854.0,SIP/2.0/UDP > > >> > GATEWAY_1;rport=5060;branch=z9hG4bK1b52.5ebf0565.0,SIP/2.0/UDP > > >> > 64.24.35.73:5060;branch=z9hG4bK53c03317;rport=5060. > > >> > Record-Route: > > >> > > > <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>. > > >> > From: "CALLER" < sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 103 BYE. > > >> > Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. > > >> > Content-Length: 0. > > >> > . > > >> > > > >> > # > > >> > U +0.000100 OpenSER:5060 -> GATEWAY_1:5060 > > >> > SIP/2.0 200 OK. > > >> > Via: SIP/2.0/UDP > > >> > GATEWAY_1;rport=5060;branch=z9hG4bK1b52.5ebf0565.0 ,SIP/2.0/UDP > > >> > 64.24.35.73:5060;branch=z9hG4bK53c03317;rport=5060. > > >> > Record-Route: > > >> > > > <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>. > > >> > From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. > > >> > To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. > > >> > Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. > > >> > CSeq: 103 BYE. > > >> > Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. > > > > >> > Content-Length: 0. > > >> > . > > >> > > > >> > ################exit > > >> > 1036 received, 0 dropped > > >> > > > >> > Thanks Again everyone. > > >> > > > >> > > > >> > > > >> > > ------------------------------------------------------------------------ > > >> > > > >> > _______________________________________________ > > >> > Users mailing list > > >> > Users(a)openser.org > > >> > http://openser.org/cgi-bin/mailman/listinfo/users > > >> > > > > > > > > > > > > > > > -- > Brandon Armstead -- Brandon Armstead

1 0

Re: [Users] openser on AMD64 or intel Xeon64
by Abdurrahman Sahin 29 Mar '07

29 Mar '07

hi again; so just setting CPU will be OK? We have 2 fedora linux machines, one on Intel Xeon 64 bit and one on AMD64 bit. i'll go on with CPU setting. is there anything else to configure? best regards. 2007/3/29, Bodin Bruno <bbodin01(a)univ-lr.fr>: > Abdurrahman Sahin a écrit : > > hi; > > i'm new to this maillist; > > is there any openser distribution for AMD64 or Intel Xeon 64. > > i read sth on the site claims that it can be build via modifying > > parameters in make config files. > > but there is nothing which parameters need to be edited > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > In Suse I'have just to type : > > CPU=athlon64 make etc.... > or > export CPU=athlon64 > and normal commands > >

1 0

[Users] openser on AMD64 or intel Xeon64
by Abdurrahman Sahin 29 Mar '07

29 Mar '07

hi; i'm new to this maillist; is there any openser distribution for AMD64 or Intel Xeon 64. i read sth on the site claims that it can be build via modifying parameters in make config files. but there is nothing which parameters need to be edited

2 1

[Users] ReInvite Issue
by Brandon Armstead 29 Mar '07

29 Mar '07

Hello everyone, I'm running into a issue, where I'm trying to implement parallel forking, so basically on incoming calls from gateway, I want to then Re-Invite, and send out the calls to multiple destinations, I am currently, using avp_db_query to grab the sip uri's and throwing them into an avp list, and avp_pushto("$ru", "$avp(i:1)/g"); This all works fine and good, invite comes in, and then sends Re-Invites, however, there seems to be some kind of race condition where, it is not properly processing all of the Re-Invites, to fork the calls out, and basically one of the calls picks up 183 early media, then 200 OK, and terminates the other branches / calls. However they were never executed / no phone was ever picked up. Any ideas on what I'm doing wrong, or what could possibly be happening? Any help is appreciated, thank you. I am using OpenSER v1.1.1, example sipcap posted below: match: CALLEE ################# U +0.000000 GATEWAY_1:5060 -> OpenSER:5060 INVITE sip:CALLEE@OpenSER:5060 SIP/2.0. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as17d625b5>. Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK0b52.a5ddb76.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>. Contact: <sip:CALLER@64.24.35.73>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. User-Agent: US LEC VOIP. Max-Forwards: 69. Date: Wed, 28 Mar 2007 13:58:45 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 156. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 64.24.35.73. t=0 0. m=audio 17990 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U +0.002597 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK0b52.a5ddb76.0;rport=5060. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Server: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. Warning: 392 OpenSER:5060 "Noisy feedback tells: pid=10278 req_src_ip=GATEWAY_1 req_src_port=5060 in_uri=sip:CALLEE@OpenSER:5060 out_uri=sip:+RE-INVITE_CALLEE_1@GATEWAY_2:5060 via_cnt==2". . # U +0.000090 OpenSER:5060 -> GATEWAY_2:5060 INVITE sip:+RE-INVITE_CALLEE_1@GATEWAY_2:5060 SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as17d625b5>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>. Contact: <sip:CALLER@64.24.35.73>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Max-Forwards: 68. Date: Wed, 28 Mar 2007 13:58:45 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 174. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 OpenSER. t=0 0. m=audio 35998 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=nortpproxy:yes. # U +0.000023 OpenSER:5060 -> GATEWAY_2:5060 INVITE sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. Record-Route: <sip:GATEWAY_1;lr=on;ftag=as17d625b5>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>. Contact: <sip:CALLER@64.24.35.73>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Max-Forwards: 68. Date: Wed, 28 Mar 2007 13:58:45 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 174. . v=0. o=root 7184 7184 IN IP4 64.24.35.73. s=session. c=IN IP4 OpenSER. t=0 0. m=audio 35998 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=nortpproxy:yes. # U +0.022891 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 100 Trying bandwidth.com. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13590 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+RE-INVITE_CALLEE_1@GATEWAY_2:5060 out_uri=sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060 via_cnt==3". . # U +0.000007 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 100 Trying bandwidth.com. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13590 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 out_uri=sip:+RE-INVITE_CALLEE_2@4.68.250.148:5060 via_cnt==3". . ############## U +2.054324 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175090325 1175090326 IN IP4 209.244.42.250. s=-. c=IN IP4 209.244.42.250. t=0 0. m=audio 61138 RTP/AVP 0. a=ptime:20. # U +0.000244 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175090325 1175090326 IN IP4 209.244.42.250. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 36002 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. ## U +0.213133 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.0,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 128. . v=0. o=- 1175090325 1175090326 IN IP4 209.244.42.250. s=-. c=IN IP4 209.244.42.250. t=0 0. m=audio 61138 RTP/AVP 0. a=ptime:20. # U +0.004792 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_1;lr=on;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. Content-Type: application/sdp. Content-Length: 143. . v=0. o=- 1175090325 1175090326 IN IP4 209.244.42.250. s=-. c=IN IP4 OpenSER. t=0 0. m=audio 36002 RTP/AVP 0. a=ptime:20. a=nortpproxy:yes. # U +0.000146 OpenSER:5060 -> GATEWAY_2:5060 CANCEL sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 SIP/2.0. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. To: <sip:CALLEE@GATEWAY_1>. CSeq: 102 CANCEL. User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. . # U +0.028419 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK5dce8d77;rport=5060. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-fcd9. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 INVITE. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13600 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 out_uri=sip:+RE-INVITE_CALLEE_2@4.68.250.148:5060 via_cnt==1". . # U +0.000098 OpenSER:5060 -> GATEWAY_2:5060 ACK sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 SIP/2.0. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. To: <sip:CALLEE@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-fcd9. CSeq: 102 ACK. User-Agent: OpenSer (1.1.1-notls (x86_64/linux)). Content-Length: 0. . # U +0.011645 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 200 ok -- no more pending branches. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.1. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. To: <sip:CALLEE@GATEWAY_1>;tag=2a6c315af837f0a6a989449d0bd9acf6-fcd9. CSeq: 102 CANCEL. Server: Bandwidth.com SDE (v1.0). Content-Length: 0. Warning: 392 GATEWAY_2:5060 "Noisy feedback tells: pid=13600 req_src_ip=OpenSER req_src_port=5060 in_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 out_uri=sip:+RE-INVITE_CALLEE_2@GATEWAY_2:5060 via_cnt==1". . # U +0.030857 GATEWAY_1:5060 -> OpenSER:5060 ACK sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK0b52.a5ddb76.2. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK29c21d5a;rport=5060. Route: <sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_2;lr;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Contact: <sip:CALLER@64.24.35.73>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 ACK. User-Agent: US LEC VOIP. Max-Forwards: 69. Content-Length: 0. . # U +0.000183 OpenSER:5060 -> GATEWAY_2:5060 ACK sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK0b52.7164f3d6.3. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK0b52.a5ddb76.2. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK29c21d5a;rport=5060. Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Contact: <sip:CALLER@64.24.35.73>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 102 ACK. Max-Forwards: 68. Content-Length: 0. P-hint: rr-enforced. . ##################################################################################################################################################################################################################################################################################################################################################################################################################################################################### U +71.203150 GATEWAY_1:5060 -> OpenSER:5060 BYE sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP GATEWAY_1;branch=z9hG4bK1b52.5ebf0565.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK53c03317;rport=5060. Route: <sip:OpenSER;lr=on;ftag=as17d625b5>,<sip:GATEWAY_2;lr;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Contact: <sip:CALLER@64.24.35.73>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 103 BYE. User-Agent: US LEC VOIP. Max-Forwards: 69. Content-Length: 0. . # U +0.003835 OpenSER:5060 -> GATEWAY_2:5060 BYE sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp SIP/2.0. Record-Route: <sip:OpenSER;lr=on;ftag=as17d625b5>. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK1b52.25391854.0. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK1b52.5ebf0565.0. Via: SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK53c03317;rport=5060. Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Contact: <sip:CALLER@64.24.35.73>. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 103 BYE. Max-Forwards: 68. Content-Length: 0. P-hint: rr-enforced. . # U +0.073996 GATEWAY_2:5060 -> OpenSER:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OpenSER;branch=z9hG4bK1b52.25391854.0,SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK1b52.5ebf0565.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK53c03317;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 103 BYE. Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. Content-Length: 0. . # U +0.000100 OpenSER:5060 -> GATEWAY_1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP GATEWAY_1;rport=5060;branch=z9hG4bK1b52.5ebf0565.0,SIP/2.0/UDP 64.24.35.73:5060;branch=z9hG4bK53c03317;rport=5060. Record-Route: <sip:GATEWAY_2;lr;ftag=as17d625b5>,<sip:OpenSER;lr=on;ftag=as17d625b5>. From: "CALLER" <sip:CALLER@64.24.35.73>;tag=as17d625b5. To: <sip:CALLEE@GATEWAY_1>;tag=VPST506071629460. Call-ID: 681bcb047355d7d85d939d542534d721(a)64.24.35.73. CSeq: 103 BYE. Contact: <sip:+RE-INVITE_CALLEE_1@4.68.250.148:5060;transport=udp>. Content-Length: 0. . ################exit 1036 received, 0 dropped Thanks Again everyone. -- Brandon Armstead

3 2

[Users] XCAP for php4
by Helge Waastad 29 Mar '07

29 Mar '07

I, some reported T_Class error on xcap. Try adding a ';' at the end og require_once........ br hw

1 0

Re: RE: [Users] Simple xcap-emulator for OpenSER
by sukerry 28 Mar '07

28 Mar '07

Toni Barata, in Config.php, a ";" is missed before the term "class" ======= 2007-03-28 23:47:03 ======= >Hi Klaus, > > >I have been using xcap-0.2.1 without problems until yesterday. Today I have upgrade to the xcap-0.5 but the following error occurs: > >PHP Parse error: parse error, unexpected T_CLASS in /var/www/xcap-root/config.php on line 19 > >I have php4 on my system and the db_mysql.php file is php3 version. Could this be the problem? Should I downgrade php4 to php3? > >Thanks, >Toni > >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Klaus Darilion >Sent: quarta-feira, 28 de Mar?o de 2007 10:54 >To: sukerry >Cc: Klaus Darilion; Adrian Georgescu; users openser.org >Subject: Re: [Users] Simple xcap-emulator for OpenSER > >there is nothing to compile. Read the Readme. you have to install PEAR >and the modules mentioned in the READ. > >e.g. >pear install Auth_HTTP >pear install Log >pear install DB > >regards >klaus > > >sukerry wrote: >> Adrian Georgescu, >> >> >> Thanks for your great job! >> >> However , when I compile php, an error says that Log.php can not be found! >> >> Thanks >> >> sukerry >> >> >> >> ======= 2007-03-28 00:56:33======= >> >>> Hello, >>> >>> We wrote a simple xcap emulator to be able to manage the XCAP >>> documents used by OpenSER presence agent, by using PHP scripts in >>> combination with Apache web server. >>> >>> I have tested it against OpenSER with MySQL and Eyebeam for Apple. >>> Klaus Darilion has tested it against Postgress and Eyebeam for Windows. >>> >>> The software can be downloaded from: >>> >>> http://download.dns-hosting.info/XCAP/xcap-0.5.tar.gz >>> >>> Regards, >>> Adrian >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >> >> 　　　　　　　　sukerry >> 　　　　　　　　sukerry(a)126.com >> 　　　　　　　　　　2007-03-28 >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　sukerry 　　　　　　　　sukerry(a)126.com 　　　　　　　　　　2007-03-29

1 0

[Serusers] avp_sprintf in ser 0.9.6?
by Stefan Sayer 28 Mar '07

28 Mar '07

Hello, is there any function which can combine several avp values, possibly with constant strings, in ser 0.9.6? I am looking for something like avp_sprintf("$dst_avp", "P-App-Param: val1=$avp1; val2=avp2\r\n") but the only function I found was avp_write which can take either string or avp. Thanks Stefan -- Stefan Sayer Media Services Development stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

3 2

Re: [Users] Unauthorized Calls - PLEASE HELP!
by Mark Kent 28 Mar '07

28 Mar '07

Are you sure you deleted the Admin user, in the subscriber database? -mark

1 0

RE: [Users] Simple xcap-emulator for OpenSER
by Toni Barata 28 Mar '07

28 Mar '07

Hi Klaus, I have been using xcap-0.2.1 without problems until yesterday. Today I have upgrade to the xcap-0.5 but the following error occurs: PHP Parse error: parse error, unexpected T_CLASS in /var/www/xcap-root/config.php on line 19 I have php4 on my system and the db_mysql.php file is php3 version. Could this be the problem? Should I downgrade php4 to php3? Thanks, Toni -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Klaus Darilion Sent: quarta-feira, 28 de Março de 2007 10:54 To: sukerry Cc: Klaus Darilion; Adrian Georgescu; users openser.org Subject: Re: [Users] Simple xcap-emulator for OpenSER there is nothing to compile. Read the Readme. you have to install PEAR and the modules mentioned in the READ. e.g. pear install Auth_HTTP pear install Log pear install DB regards klaus sukerry wrote: > Adrian Georgescu, > > > Thanks for your great job! > > However , when I compile php, an error says that Log.php can not be found! > > Thanks > > sukerry > > > > ======= 2007-03-28 00:56:33======= > >> Hello, >> >> We wrote a simple xcap emulator to be able to manage the XCAP >> documents used by OpenSER presence agent, by using PHP scripts in >> combination with Apache web server. >> >> I have tested it against OpenSER with MySQL and Eyebeam for Apple. >> Klaus Darilion has tested it against Postgress and Eyebeam for Windows. >> >> The software can be downloaded from: >> >> http://download.dns-hosting.info/XCAP/xcap-0.5.tar.gz >> >> Regards, >> Adrian >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > > > 　　　　　　　　sukerry > 　　　　　　　　sukerry(a)126.com > 　　　　　　　　　　2007-03-28 > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] Status of Ottendorf and Presence
by Steve Blair 28 Mar '07

28 Mar '07

Would someone be able to tell me the status of the Ottendorf release? In addition to knowing when the stable release will be available I am also interested in when presence support will be available. Thanks,Steve

5 5

[Serusers] msilo module problem
by POON Ben 28 Mar '07

28 Mar '07

Hi all, I am trying to use the msilo module. On the website it says the dependency is only a db module and the tm module. Both of which are included in my ser.cfg file before the loadmodule for the msilo module. Can someone please help me find out what is wrong? Thanks. However, when I try to start ser, it gives me this: ************************************************************************ ****************** bash# ser -D -E 0(29857) parse error (166,17-18): unknown command, missing loadmodule? 0(29857) parse error (196,22-23): unknown command, missing loadmodule? 0(29857) parse error (243,20-21): unknown command, missing loadmodule? ERROR: bad config file (3 errors) Segmentation Fault (core dumped) ************************************************************************ ******************* I've checked the lines 166, 196, and 243, they all correspond to lines where I use m_dump and m_store. The config file is basically a copy of the one provided at http://www.iptel.org/ser/doc/modules/msilo Are there other dependencies not listed on the module page? I don't need SEMS or anything like that do I? Thank you very much, Ben

2 3

[Users] SIP Registrartions
by raviprakash sunkara 28 Mar '07

28 Mar '07

Hello Users , I have one OpenSER server , this can handle the Both Sip Registrar , location server, proxy server, But I need OpenSER only proxy that , When UAC send the Registrartion to OpenSER , and OpenSER send to Vendor Proxy ( Cisco Proxy server ). And other one I have 2 openSER , first one Registrar server say OP1 and other is OP2 UAC( register) -------->OP2-----------> OP1( here it have to save ) -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

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[Serusers] global contact list
by Barta János 28 Mar '07

28 Mar '07

Hello! I have a little problem with the (global) contact list subscription of the presence, please help me. I installed the presence server (0.10.99-dev35-pa-4.1), and it works without any problems. But I can not use the "Subscribe to contact list" of the eyeBeam. (The user subscribe list works fine.) I typed the next content into this field (of the eyeBeam): sip:bartaj-list@test.intra and my config file contains: sip:/var/xcap-root/rls-services/global# cat ./index <?xml version="1.0" encoding="UTF-8"?> <rls-services> <service uri="sip:bartaj-list@test.intra"> <list name="Human resources"> <entry uri="sip:test1@test.intra"> <display-name>bartaj</display-name> </entry> <entry uri="sip:test2@test.intra"> <display-name>12345</display-name> </entry> </list> <packages> <package>presence</package> </packages> </service> </rls-services> But I can see only the user subscribe list, and can't see the global one. Why? Can anybody help me? Thx, Jani _________________________________________________________________ Vásároljon kényelmesen! Több 10.000 ajánlattal, kedvező árakkal várjuk. http://cthandler.adverticum.net/?cturl=http%3A%2F%2Fwww.vasarlas.origo.hu%3…

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[Users] Radius
by jidn jidn 28 Mar '07

28 Mar '07

Hi list, I need to know if open ser have a module, that uses radius, for connect with Alepo Because I have Alepo for billing. Thanks, Pablo __________________________________________________ Preguntá. Respondé. Descubrí. Todo lo que querías saber, y lo que ni imaginabas, está en Yahoo! Respuestas (Beta). ¡Probalo ya! http://www.yahoo.com.ar/respuestas

1 0

Re: [Users] Simple xcap-emulator for OpenSER
by sukerry 28 Mar '07

28 Mar '07

Adrian Georgescu, Thanks for your great job! However , when I compile php, an error says that Log.php can not be found! Thanks sukerry ======= 2007-03-28 00:56:33======= >Hello, > >We wrote a simple xcap emulator to be able to manage the XCAP >documents used by OpenSER presence agent, by using PHP scripts in >combination with Apache web server. > >I have tested it against OpenSER with MySQL and Eyebeam for Apple. >Klaus Darilion has tested it against Postgress and Eyebeam for Windows. > >The software can be downloaded from: > >http://download.dns-hosting.info/XCAP/xcap-0.5.tar.gz > >Regards, >Adrian > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > 　　　　　　　　sukerry 　　　　　　　　sukerry(a)126.com 　　　　　　　　　　2007-03-28

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[Users] Content-type in PUBLISH request
by jh＠tutpro.com 28 Mar '07

28 Mar '07

i started to experiment with sending PUBLISH requests using pua_mi module. if i understood correctly from RFC 3903, PUBLISH request should contain a content-type header, such as Content-Type: application/pidf+xml PUBLISH request sent via pua_mi does not contain any content-type header. should one be added so that the receiver of PUBLISH request could know how to decode the body? another question related to this is, what event packages does openser presence server support? i assume that it supports "presence" event package (where content type usually is pidf+xml), but how about "message-summary" event package (RFC 3842) that could be used to publish message waiting events? -- juha

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[Users] OpenSER 1.2 support for sip:wizard
by SipWise Support 28 Mar '07

28 Mar '07

Hi all, We'd like to announce that sip:wizard (http://www.sipwise.com/wizard), the OpenSER configuration generator, now also supports OpenSER 1.2. Cheers, Your sip:wise Team

2 1

[Users] out of memory
by Papadopoulos Georgios 28 Mar '07

28 Mar '07

Hello, Lately our OpenSER 1.1.1 is running out of memory and I get the following message 4(25190) get_hdr_field: out of memory The result is that it stops processing any incoming request. This is a production system and of course it causes a great deal of frustration. The last time this happened I got the following statistics before restarting > ./openserctl fifo get_statistics Total statistics = 37 Total modules = 5 Module name = core; statistics=8 core:rcv_requests = 3259334 core:rcv_replies = 8816 core:fwd_requests = 736 core:fwd_replies = 137 core:drop_requests = 743 core:drop_replies = 0 core:err_requests = 0 core:err_replies = 43 Module name = shmem; statistics=6 shmem:total_size = 268435456 shmem:used_size = 1675864 shmem:real_used_size = 1835104 shmem:max_used_size = 4399048 shmem:free_size = 266600352 shmem:fragments = 17710 Module name = sl; statistics=9 sl:1xx_replies = 0 sl:2xx_replies = 1690410 sl:3xx_replies = 43 sl:4xx_replies = 334827 sl:5xx_replies = 1185629 sl:6xx_replies = 0 sl:sent_replies = 3210909 sl:sent_err_replies = 0 sl:received_ACKs = 743 Module name = tm; statistics=11 tm:received_replies = 8636 tm:relayed_replies = 7096 tm:local_replies = 1718 tm:UAS_transactions = 7003 tm:UAC_transactions = 94 tm:2xx_transactions = 1528 tm:3xx_transactions = 3 tm:4xx_transactions = 5129 tm:5xx_transactions = 468 tm:6xx_transactions = 0 tm:inuse_transactions = 0 Module name = usrloc; statistics=3 usrloc:location-users = 0 usrloc:location-contacts = 0 usrloc:location-expires = 7282 I am not sure how to read these numbers but I see a big "free_size" and still OpenSER is out of memory. Any suggestions? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Users] why b2bua always relay 403
by liu york 28 Mar '07

28 Mar '07

Dear all I use astersik b2bua and radius for auth and prepaid on pstn call,there is a big problem is that any responese from PSTN GW,for example 406,503...,the bubua always relay 403 to sip client.So i cant make the status clearly when pstn call,what's wrong with my b2bua,any cfg of bubua for me? many many thx.

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[Users] Openser Not Send Ack to Callee?
by 杨冬 28 Mar '07

28 Mar '07

hi, all: I'm new to here and just installed an instance of openser-1.2.0-notls version under CentOS 4.3 platform. Also I've added mysql db support and it seemed all is working well. But when I made a common call through openser, I found a problem that the openser didn't send ACK message to the callee. The flow is as follows: caller ---INVITE--> openser openser ---INVITE--> callee openser <--180-------- callee caller <----180------ openser openser <----200------- callee caller <----200------ openser caller ------ACK----> openser At this time when the caller send ACK to openser, openser didn't send ACK to the callee. So after 30 seconds, the callee released the call due to time out. Does anyone know what's the problem? Any help will be appreciate!

2 2

[Serusers] Problems with ACK
by Ricardo Martinez 27 Mar '07

27 Mar '07

Hello List. I have a problem with an ACK message for a CANCEL. I'm just wondering if anyone knows what could be happening here. It seems that SER can't route or "understand" this ACK ACK sip:0299005622408196@sipvoiss.desa.redvoiss.net SIP/2.0. v: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK2861608023. f: "5502203289" <sip:5502203289@sipvoiss.desa.redvoiss.net>;tag=1349608987. t: <sip:0299005622408196@sipvoiss.desa.redvoiss.net>;tag=f146818fa4. i: 5c83a04822964085e07aabd210e9d72f(a)192.168.1.100. CSeq: 4 ACK. l: 0. Could be something with the COMPACT form ??? This is the log from the debug : Mar 23 12:18:10 sipvoiss ser[5430]: t_lookup_request: start searching: hash=52251, isACK=1 Mar 23 12:18:10 sipvoiss ser[5430]: DEBUG: RFC3261 transaction matching failed Mar 23 12:18:10 sipvoiss ser[5430]: DEBUG: t_lookup_request: no transaction found Mar 23 12:18:10 sipvoiss ser[5430]: SER: forwarding ACK statelessly Mar 23 12:18:10 sipvoiss ser[5430]: DEBUG: mk_proxy: doing DNS lookup... Mar 23 12:18:10 sipvoiss ser[5430]: get_record: lookup(_sip._udp.sipvoiss.desa.redvoiss.net, 33) failed Mar 23 12:18:10 sipvoiss ser[5430]: sip_resolvehost: no SRV record found for sipvoiss.desa.redvoiss.net, trying 'normal' lookup... Mar 23 12:18:10 sipvoiss ser[5430]: check_via_address(10.0.0.248, 192.168.1.100, 0) Mar 23 12:18:10 sipvoiss ser[5430]: Sending: ACK sip:0299005622408196@sipvoiss.desa.redvoiss.net SIP/2.0^M Max-Forwards: 10^M Record-Route: <sip:10.0.0.246;ftag=2639303304;lr=on>^M Via: SIP/2.0/UDP 10.0.0.246;branch=0^M v: SIP/2.0/UDP 192.168.1.100:5060;received=10.0.0.248;branch=z9hG4bK365542204^M f: "5502203289" <sip:5502203289@sipvoiss.desa.redvoiss.net>;tag=2639303304^M t: <sip:0299005622408196@sipvoiss.desa.redvoiss.net>;tag=2746a580a4^M i: 81983e61d980fd3ef0046dafe9518284(a)192.168.1.100^M CSeq: 10 ACK^M l: 0^M ^M . Mar 23 12:18:10 sipvoiss ser[5430]: orig. len=338, new_len=477, proto=1 Mar 23 12:18:10 sipvoiss ser[5430]: DEBUG:destroy_avp_list: destroying list (nil) Mar 23 12:18:10 sipvoiss ser[5430]: receive_msg: cleaning up What does it means : "RFC3261 transaction matching failed" and why ??? The weird situation happend when almost the same message arrives without the Compact Form : V102 ACK sip:0299005622408196@sipvoiss.desa.redvoiss.net SIP/2.0. Via: SIP/2.0/UDP 10.0.0.237:5060;branch=z9hG4bK2364391837. From: "5502203289" <sip:5502203289@sipvoiss.desa.redvoiss.net>;tag=827237367. To: <sip:0299005622408196@sipvoiss.desa.redvoiss.net>;tag=6b466391a4. Call-ID: 43b2cc325b7fbe9f8790e4cee01dc4b3(a)10.0.0.237. CSeq: 2 ACK. Content-Length: 0. This message works ok!. .... ... Mar 23 16:30:06 sipvoiss ser[5974]: DEBUG: RFC3261 transaction matched, tid=424819551 Mar 23 16:30:06 sipvoiss ser[5974]: DEBUG: t_lookup_request: transaction found (T=0x404ce3e8) Mar 23 16:30:06 sipvoiss ser[5974]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Mar 23 16:30:06 sipvoiss ser[5974]: DEBUG: add_to_tail_of_timer[2]: 0x404ce430 Mar 23 16:30:06 sipvoiss ser[5974]: DEBUG:destroy_avp_list: destroying list (nil) Mar 23 16:30:06 sipvoiss ser[5974]: receive_msg: cleaning up Can someone give me a hand here? Thanks in advance Regards, Ricardo Martinez-

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[Serusers] New search-page
by Marcin Pazdro 27 Mar '07

27 Mar '07

Hi all, I'm going to improve the documentation search page (or rather to create new search engine). Here: http://www.iptel.org/ser_documentation_dynamic_search_page_functional_speci… I have placed a functional specification that describes main guidelines of new search page. I would like to get some feedback from you. Please let me know if you have some comments, tips or questions. pazdro :)

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Re: [Users] register more than one username by IP phone
by Klaus Darilion 27 Mar '07

27 Mar '07

forgotten to Cc the list ... Klaus Darilion wrote: > I think what you need are aliases. An alias maps from one SIP URI to > another. > > e.g. sip:kd@domain.com --> sip:klaus.darilion@domain.com > sip:+4315056416@domain.com --> sip:klaus.darilion@domain.com. > > Thus, if openser receives a request, you first lookup the alias and then > lookup the location table. You can either use the aliases table and use > lookup("aliases") or use the ALIAS_DB module. > > regards > klaus > > Bodin Bruno wrote: >> Klaus Darilion a écrit : >>> The only identifier which is REGISTERed is the current location of >>> the SIP client (ipaddress:port). This gets mapped to the SIP AoR (the >>> user's SIP address). >>> >>> A user can use several clients concurrently. Then this user will show >>> up multiple times in the location table - one entry for each SIP client. >>> >>> Hope this helps - I still don't understand what you want to achieve. >>> >>> regards >>> klaus >>> >>> Bodin Bruno wrote: >>>> Hi, >>>> Is it possible to register two (or more) identifiers by user, such >>>> as phone number, username or service ? >>>> Thank >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >> Hi, >> In some steps, >> >> what I do : >> >> step 1. Softphone clients are registered in openser with >> authentification (digest etc....) >> step 2. Openser use radius to check them (with ldap) >> step 3. Openser do save("location") to store softphone address. >> >> What I need : >> >> after step 3, authenticate username, and phone number of client are >> usable to call them. >> >> What I don't need : >> >> currently, the username used to call a softphone is the only choice of >> this user and this choice isn't limited. >> >> I'm sorry for my poor english, i'm french student. >> >> thank for your help >> >> >> >> >

3 2

[Serusers] Jabber w/ Ser
by SER LIST 27 Mar '07

27 Mar '07

Hello, has anyone successfuly worked with Jabber module? Especially forwarding messages to Hotmail, yahoo, etc. I am having issues and wanted to know whether there are issues in implementation of jaber module or not?Thanks in advance _________________________________________________________________ Your Space. Your Friends. Your Stories. Share your world with Windows Live Spaces. http://spaces.live.com/?mkt=en-ca

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[Users] Openser on Sun Ultrasparc T1 Processors - Mismatch architecture detection
by Sergio Gutierrez 27 Mar '07

27 Mar '07

Hi Henning. I tested the compilation and it does not detect the architecture. I think that maybe is because of uname -p outputs sparc, and uname -m outputs sun4v. Maybe, would it be possible to include the fix to this condition at the later if (line 83 at Makefile.defs) # fix sparc -> sparc64 ifeq ($(ARCH),sparc) ifeq ($(shell uname -m),sun4u) ARCH := sparc64 endif endif Kind regards On 3/27/07, Henning Westerholt <henning.westerholt(a)1und1.de> wrote: > > On Tuesday 27 March 2007 17:12, you wrote: > > Hi Henning. > > > > The patch applies OK. > > Hi Sergio, thank you. > And is the right architecture detected, if you do a fresh compile with > this > patch? > > Best regards, > > Henning >

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Re: [Serusers] re: CONTACT field malformed in a 302 reply
by Jiri Kuthan 27 Mar '07

27 Mar '07

Contact from SER is NOT malformed. It is 100% compliant to the RFC3261 spec (see sections 20.10 and 25 of RFC 3261). in the grammar, contact expands to name-addr|addr-spec, with addr-spec being URI. Exapmles in the addr-spec for also appear throughout the spec. Thus I suggest that the softswitch vendor makes himself familiar with this document. As a workaround you may wish trying some manual tricks to form a reply which will make the UAC equipment happy (append_to_reply). -jiri p.s. interestingly, gl_redirect must be your own feature, so if you developed some extra SER code updating the way of generating Contacts in 3xx should not be a big deal. I just occur to think that doing it in public SER doesn't make much sense, because if there are multiple ways to do things in SIP, whichever you choose, you will find someone who is silly enough only to accept the opposite one :-) At 20:57 26/03/2007, Jignesh Gandhi wrote: >Hello, > >I am using SER as a redirect server. >Recently I came across an issue where the 302 sent back by SER is >not liked by a soft switch. Particularly , the format of the CONTACT >field according to the softswitch.. > >Here is an excerpt of the 302 reply send back ... > >Session Initiation Protocol > Status-Line: SIP/2.0 302 MovedTemporarily > Status-Code: 302 > Resent Packet: True > Suspected resend of frame: 73 > Message Header > To: <<mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46>;tag=b27e1a1d33761e85846fc98f5f3a7e58.dfa5 > SIP to address: <mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46 > SIP tag: b27e1a1d33761e85846fc98f5f3a7e58.dfa5 > From: <http://172.20.20.46>172.20.20.46<<mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46>;tag=161dda6e > SIP Display info: <http://172.20.20.46>172.20.20.46 > SIP from address: <mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46 > SIP tag: 161dda6e > Via: SIP/2.0/UDP <http://10.99.99.140:9585>10.99.99.140:9585;branch=z9hG4bK-d87543-730866470-1--d87543-;rport=9585 > Call-ID: 95239a63f0347c53 > CSeq: 1 INVITE > Contact: sip:12345@172.20.20.37:5060 > Server: Sip EXpress router (0.8.14 (i386/linux)) > Content-Length: 0 > Warning: 392 <http://172.20.20.46:5060>172.20.20.46:5060 "Noisy feedback tells: pid=9550 req_src_ip=<http://10.99.99.140>10.99.99.140 req_src_port=9585 in_uri=<mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46 out_uri=sip:12345@172.20.20.37:5060 via_cnt==1 > >the softswitch is wanting the contact field with <sip:xxxx@xxx.xxx.xxx.xxx:5060> like a FROM or TO URI. > >Here is ser.cfg part that does the sl_send_reply(); > > # do a stateless redirect, if return code is correct > if (method=="INVITE") > { > xlog("L_INFO", "SourceIP <%is> \n"); > xlog("L_INFO", "From-uri<%fu>, r-uri <%ru> \n"); > if (!gl_redirect()) > { > sl_send_reply("480", "TemporarilyUnavailable"); > xlog("L_WARN", "Sending a 480 response with r-uri <%ru>\n"); > break; > } > else > { > sl_send_reply("302", "MovedTemporarily"); > xlog("L_DBG", "Sending a 302 response with r-uri <%ru>\n"); > break; > } > }; > >Any help is appreciated. > > >thanks, >-- >Jignesh Gandhi ><mailto:jigpgandhi@gmail.com>jigpgandhi(a)gmail.com >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Users] Openser on Sun Ultrasparc T1 Processors - Mismatch architecture detection
by Sergio Gutierrez 27 Mar '07

27 Mar '07

Hi everybody. I think that there is something missing on Makefile.defs. Sun Machines using the new line Processors (UltraSparc T1) are not correctly detected as a Sparc 64 bit machine. I think that architecture detection should include "sun4v" (uname -m output on a Sun Fire T1000 or T2000) as equivalent to Sun4u. Thanks. Sergio Gutiérrez.

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[Users] fifo,fifo_mode commands in 1.2.0-tls
by Steve Blair 27 Mar '07

27 Mar '07

I'm upgrading a 1.0 system to 1.2.0-tls and I get a few config file errors when trying to start the proxy using the previous openser.cfg file. It seems the fifo, fifo_mode and fifo_db_url commands have changed but I can only find a change log reference for fifo_mode. What are the correct replacements for these commands? Thanks,Steve

3 2

Re: [Users] Not able to access avp variables in reply route
by Asterisk Expert 27 Mar '07

27 Mar '07

On 3/26/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > > > Asterisk Expert wrote: > > So you mean it is not possible to access values set from one > > dialog(INVITE-200 OK) in (BYE-200OK) ? > > You can use avp_db_query to store a certain value into a table during > INVITE and fetch it back during BYE. Of course you need a key which is > identical during the dialog - e.g. callid+fromtag+totag. But totag will be null before relaying invite request, isn't it? Currently I did it with just call id. Let me know if it will create any problem. regards > klaus > > > Is there any alternative? I must have to use custom CDR not db or > radius. > > > > On 3/26/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > >> > >> > >> > >> Asterisk Expert wrote: > >> > I thought one call is one transaction isn't it? > >> > >> No. > >> > >> A transaction is a request till the final response, e.g. INVITE-200 OK. > >> or OPTIONS-200 Ok. > >> > >> In SIP terminology a call is named "dialog". A dilaog consists of > >> multiple transaction. The first transaction is the "dialog creating > >> transaction" (e.g. INVITE) and the last transaction in this dialog is > >> the "dialog terminating transcation" e.g. BYE. > >> > >> Thus, INVITE and BYE are seperate transactions which belong to the same > >> dialog. > >> > >> Openser is transaction statefull, but not dialog statefull. > >> > >> regards > >> klaus > >> > >> > > >> > On 3/26/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > >> >> > >> >> I guess you are trying to use an AVP created during INVITE in the > BYE > >> >> transaction. This is not possible as the BYE is a different > >> transaction > >> >> and AVP lifetime is limited to a single transaction. > >> >> > >> >> regards > >> >> klaus > >> >> > >> >> Asterisk Expert wrote: > >> >> > Hello, > >> >> > I'm trying to implement custom CDR using OpenSER. What I do is, > >> >> insert a > >> >> > record in cdr table on invite request. I get the id of the > inserted > >> row > >> >> > using LAST_INSERT_ID() function of mysql in avp_db_query and > >> store in > >> >> avp > >> >> > variable rowid. I set connect time in reply route when I receive > 200 > >> OK > >> >> for > >> >> > Invite request. When I get 200 OK for BYE request, I update the > cdr > >> >> record > >> >> > and store connect time, disconnect time, duration, etc. But at > this > >> >> > place(reply route) I get the null rowid so as connect time. > >> >> According to > >> >> tm > >> >> > module's onreply_avp_mode parameter, I should be able to see the > >> values > >> >> set > >> >> > from request route in reply route if I set that parameter to 1 > >> but it > >> >> > has no > >> >> > effect. I can not use either avp variables or script variables as > >> >> > transaction variables to record custom cdr. Can anyone guide me > >> >> > implementing > >> >> > this scenario or tell me what I've done wrong? > >> >> > > >> >> > > >> >> > > >> >> > >> > ------------------------------------------------------------------------ > >> >> > > >> >> > _______________________________________________ > >> >> > Users mailing list > >> >> > Users(a)openser.org > >> >> > http://openser.org/cgi-bin/mailman/listinfo/users > >> >> > >> > > >> > > >> > > >> > > > > > > > -- Regards Ruchir Brahmbhatt

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RE: [Users] load testing presence server
by Toni Barata 27 Mar '07

27 Mar '07

Hi Daniel, In a scenario in witch we have a winfo and presence Subscribe with a 70 sessions per second after a wile (3906 sessions in last test) the openser crashes. Can you take a look in the attached logs from /var/log/openser.log and from core dump (they are on the same attached file)? Thanks, Toni -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: terça-feira, 27 de Março de 2007 10:12 To: Toni Barata Cc: users(a)openser.org Subject: Re: [Users] load testing presence server Hello, On 03/26/07 19:12, Toni Barata wrote: > Hi Daniel, > > Increasing the children from 4 to 16 did turn into same results (I was already using debug=3 and fork=yes). > > But setting debug=0 did real turn in much better results. > looks like lot of syslog messages are printed. Can you check if your syslog is set asynchronous for openser? Cheers, Daniel > Thanks a lot, > Toni > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: segunda-feira, 26 de Março de 2007 16:53 > To: Toni Barata > Cc: users(a)openser.org > Subject: Re: [Users] load testing presence server > > Hello, > > if you use exactly the same config, then turn debug to a lower level > (e.g., 3), set fork=yes and children to 16 for better results. > > Cheers, > Daniel > > > On 03/26/07 18:48, Toni Barata wrote: > >> Hi Anka, >> >> >> >> In a scenario using Openser as a Presence Sever (using the config file >> example located in >> http://openser.org/dokuwiki/doku.php/presence:configuration-file), the >> execution of load tests (with sipp) with Subscribe Presence WInfo had >> as result a maximum of 20 sessions per second (We are using a Intel(R) >> Xeon(TM) CPU 3.00GHz with 2GB RAM), with lots of retransmissions if we >> increase to values bigger than 20 (Openser dos not crash). Is there >> any configurable parameter in the Presence module that allows >> increasing this value? >> >> >> >> Best regards, >> >> Toni >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >

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RE: [Users] load testing presence server
by Toni Barata 27 Mar '07

27 Mar '07

Hi Ovidiu, Thanks for the tip. But using the asynchronous mode it was not enough to increase the presence server performance if debug parameter is >= 3. Best Regards, Toni -----Original Message----- From: Ovidiu Sas [mailto:sip.nslu@gmail.com] Sent: segunda-feira, 26 de Março de 2007 22:24 To: Toni Barata Cc: daniel(a)voice-system.ro; users(a)openser.org Subject: Re: [Users] load testing presence server Hi Toni, Maybe this will help you: http://www.voice-system.ro/docs/ser-syslog/ar01s06.html Check the Note at the end of the page: NOTE: The '-' in front of "/var/log/ser.log" is to omit synchronizing the log file after every log message. If you choose to configure "syslog" in synchronous mode you must be aware that it has big impact over performances when the signaling traffic is high. Regards, Ovidiu Sas On 3/26/07, Toni Barata <toni-r-barata(a)ptinovacao.pt> wrote: > Hi Daniel, > > Increasing the children from 4 to 16 did turn into same results (I was already using debug=3 and fork=yes). > > But setting debug=0 did real turn in much better results. > > Thanks a lot, > Toni > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: segunda-feira, 26 de Março de 2007 16:53 > To: Toni Barata > Cc: users(a)openser.org > Subject: Re: [Users] load testing presence server > > Hello, > > if you use exactly the same config, then turn debug to a lower level > (e.g., 3), set fork=yes and children to 16 for better results. > > Cheers, > Daniel > > > On 03/26/07 18:48, Toni Barata wrote: > > > > Hi Anka, > > > > > > > > In a scenario using Openser as a Presence Sever (using the config file > > example located in > > http://openser.org/dokuwiki/doku.php/presence:configuration-file), the > > execution of load tests (with sipp) with Subscribe Presence WInfo had > > as result a maximum of 20 sessions per second (We are using a Intel(R) > > Xeon(TM) CPU 3.00GHz with 2GB RAM), with lots of retransmissions if we > > increase to values bigger than 20 (Openser dos not crash). Is there > > any configurable parameter in the Presence module that allows > > increasing this value? > > > > > > > > Best regards, > > > > Toni > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Users] register more than one username by IP phone
by Bodin Bruno 27 Mar '07

27 Mar '07

Hi, Is it possible to register two (or more) identifiers by user, such as phone number, username or service ? Thank

4 6

[Users] Contact header modification
by Michel Bensoussan 27 Mar '07

27 Mar '07

Hello. Is there a way to modify the contact header? For example: REGISTER REGISTER UA -----------------------------------------------> openser ----------------------------------------------------> Registrar IP:1.0.0.1 contact: ua(a)1.0.0.1 IP:1.0.0.2 contact: ua(a)1.0.0.2 The idea is to ensure that incoming call (INVITE) will path through the openser. Another setup is two interfaces, one public and one internal. Is it possible with openser? REGISTER REGISTER UA -----------------------------------------------> openser ----------------------------------------------------> Registrar IP:1.0.0.1 contact: ua(a)1.0.0.1 IP:1.0.0.2 IP:10.2.1.25 contact: ua(a)10.2.1.25 Thanks. Regards, Michel.

2 2

[Serusers] Record Routing
by KUMAR 27 Mar '07

27 Mar '07

I am using SER 0.9.6 along with MS RTC Client API for the client application. When I used the line: if(method!=REGISTER){ record_route(); } It caused in a one way communication only in case of audio as well as in case of in case of instant messaging, a new communication session had to be established for each message. But when i removed the record_route() and used sth like: if(method==ACK){ t_relay(); } When my code included both the above code snippets, it never reached the later: if (method==ACK)... part. Can anyone please tell me why when record_route() was switched on, the control never reached the later part. Thank you kumar

2 1

RE: [Users] Problem configuring the openSER with the DISC
by priyanka.bhargava＠wipro.com 27 Mar '07

27 Mar '07

Hi, I have still not reached the point of authentication, the failure is happening at establishment on the TCP connection between openSER and Diameter Client. Thanks, Priyanka ________________________________ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, March 27, 2007 4:03 PM To: Priyanka Bhargava (WT01 - IP-Multimedia Carrier & Ent Networks) Subject: Re: [Users] Problem configuring the openSER with the DISC Hello Can u check the port forwoarding of diameter clinet and server, if there are in remote. In diameter server , client file is there, in that u need to give the openser username(a)domain.openSER and Password and nas ( n.w access server ) file in this U server has to know the DC and same in DC server file will this need to tell the DS. I guess, Because , Radius and Diameter are Same purpose using in openSER and other AAA purpose. But in My experience Radius is Best, On 3/27/07, priyanka.bhargava(a)wipro.com < priyanka.bhargava(a)wipro.com <mailto:priyanka.bhargava@wipro.com> > wrote: Hi Ravi, Thanks for your quick response. I'm actually connecting openSER with the diameter client. But the client is not accepting the connection. The configuration between the diameter client and server are already done. And they are communicating properly the only problem is the communication between the Diameter Client and the openSER. Thanks, Priyanka ________________________________ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, March 27, 2007 2:16 PM To: Priyanka Bhargava (WT01 - IP-Multimedia Carrier & Ent Networks) Cc: users(a)openser.org Subject: Re: [Users] Problem configuring the openSER with the DISC Hello Priyanka, Diameter and radius both are same purpose. DialMeter Server never communicate with OpenSER, Vice versa, So Need the DiaMeter client, Which play the main role to Both openSER or Diameter, U problem in Diameter Clinet and server Configuration, -UA -------->OpenSER-->DiameterClient ---> Diameter Server. in DiC u should configured the authentation, where the Server is located. I'm Using the Radius for AAA, Thanks & Regards Sunkara Ravi Prakash Hyd..... On 3/27/07, priyanka.bhargava(a)wipro.com < priyanka.bhargava(a)wipro.com> wrote: Hi All, I'm trying use to Diameter Server Client (DISC) for the authentication purpose (using the auth_diameter module). But I'm facing issues while creating a connection between openSER and DISC. The diameter client does not accept the connection from the openSER. Please let me know if somebody has tried this out and can pass me the config files and the code changes if any. Thanks, Priyanka The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[Users] Invalid Table Version
by Martin Klisch 27 Mar '07

27 Mar '07

Hello, after upgrading our MySQL 4.0.x to 5 and compiling the openser(1.1.1)-mysql modul with the new mysql libraries i always get this error: [ID 272136 local1.error] table_version(): Invalid type (2) or nul (0) version columns for subscriber The tableversion is the right one and i can access the openser tables with the openser-user via regular mysql console client. now i tried it with mysql 4.1 - it's the same problem. anyone has an idea, what the problem could be?

2 3

RE: [Users] PRESENCE module issues
by Geir O. Jensen 27 Mar '07

27 Mar '07

Well, I did what you said... Grabbed the SVN (today, Tuesday 27th), purged the old db, modules, binaries and libraries ... Recompiled, ran openser_mysql.sh create and now I get this error; PRESENCE:mod_init: Wrong version v1 for table <presentity>, need v0 ... :-/ Now what - Geir > -----Original Message----- > From: Anca Vamanu [mailto:anca@voice-system.ro] > Sent: Monday, March 26, 2007 7:01 PM > To: Geir O. Jensen > Subject: Re: [Users] PRESENCE module issues > > Hello, > > Don't worry. It will for sure be up and running by then. > > I was not talking about the offical release, but the SVN > version of OpenSer1.2. > To download it use: > > # svn co > https://openser.svn.sourceforge.net/svnroot/openser/branches/1 > .2 openser More information at: > http://www.openser.org/mos/view/Download/ > > It must work now :) . > > Best regards, > Anca > > > > Geir O. Jensen wrote: > > >Okay, I downloaded the 1.2.0 tarball on the 19th of March. Was it > >changed after that date? The date stamp on the > >http://www.openser.org/pub/openser/1.2.0/src/ server location says > >March 12th... > > > >Hopefully I'll be able to do this within a few days... Easter is > >looming and things slow down around here then :D > > > >Thanks for your help so far! > > > >- Geir > > > > > > > >>-----Original Message----- > >>From: Anca Vamanu [mailto:anca@voice-system.ro] > >>Sent: Monday, March 26, 2007 10:17 AM > >>To: G.O. Jensen > >>Cc: daniel(a)voice-system.ro > >>Subject: Re: [Users] PRESENCE module issues > >> > >>Hello, > >> > >>I was in fact refering to the newst version of OpenSer 1.2.0. > >>It has also been updated. > >>You can also take the svn version- it will work with the > modules from > >>1.2.0, but you have to make a change in the > >>database- adding 2 new columns in active_watcher_table > >>- so you should run the database script again. It has a transport > >>issue fixed which is still under testing. > >> > >>Best regards, > >> > >>Anca Vamanu > >> > >>G.O. Jensen wrote: > >> > >> > >> > >>>Hm, righto. Well, I used the "official" downloadable tarball from > >>>openser.org. Maybe they should rush out a 1.2.1 patch asap then :D > >>> > >>>So, do I need to grab all modules and OpenSER core from > SVN too, or > >>>while the "updated" presence module work with the "stable" 1.2.0 > >>>OpenSER release? > >>> > >>>Anca Vamanu said: > >>> > >>> > >>> > >>> > >>>>Hello, > >>>> > >>>>I found the problem, much simpler that we might have > >>>> > >>>> > >>thought. You use > >> > >> > >>>>an old version, at least a week old. There was a bug there that I > >>>>fixed last week. > >>>>You should take the latest version on svn and it should work. > >>>> > >>>>Best regards, > >>>> > >>>>Anca Vamanu > >>>> > >>>> > >>>>G.O. Jensen wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>Hey, I really want to get this working so all the effort I > >>>>> > >>>>> > >>can pour > >> > >> > >>>>>into it :D I will do more testing over the weekend > though - I have > >>>>>two servers so I can see if the problems crops up on both. > >>>>> > >>>>>The clients use are eyeBeam 1.5 (Windows) and Gaim > 2.0beta (Linux). > >>>>>OpenSER crashes when I register the eyeBeam client and the > >>>>> > >>>>> > >>presence > >> > >> > >>>>>exchange starts. Fork is indeed set to yes in my Config > >>>>> > >>>>> > >>(which I can > >> > >> > >>>>>get you a copy of over the weekend aswell to help > reproduce it at > >>>>>your > >>>>>end...) > >>>>> > >>>>>(I'm at home now so I can't access all data easily...) > >>>>> > >>>>>Thanks, > >>>>>G.O. Jensen > >>>>>UNINETT > >>>>> > >>>>>Anca Vamanu said: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>Thanks for your effort. > >>>>>>Unfortunatelly the backtrace is not revealing at all. It seems > >>>>>>compulsory to reproduce your scenario to be able to fix > >>>>>> > >>>>>> > >>this. And I > >> > >> > >>>>>>just can't manage to do this. > >>>>>>What clients are you using? > >>>>>>Is fork yes or no in your configuration? > >>>>>> > >>>>>>Many thanks. > >>>>>> > >>>>>>G.O. Jensen wrote: > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>>Anca-Maria Vamanu said: > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>>>I can't reproduce what you got there and I can't figure > >>>>>>>> > >>>>>>>> > >>out from > >> > >> > >>>>>>>>the log what goes wrong. > >>>>>>>>Could you please send be the backtrace of the generated core. > >>>>>>>>Use: gdb openser core > >>>>>>>> >bt > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>Sure can do... this error was reproducible for me (I > tried 3 or 4 > >>>>>>>times...) > >>>>>>> > >>>>>>>Main routing script does a if method= then handle_() > >>>>>>> > >>>>>>> > >>thingy... and > >> > >> > >>>>>>>then returns. > >>>>>>> > >>>>>>>root@test-sip:/# gdb openser core GNU gdb 6.4-debian Copyright > >>>>>>>2005 Free Software Foundation, Inc. > >>>>>>>GDB is free software, covered by the GNU General > Public License, > >>>>>>>and you are welcome to change it and/or distribute > copies of it > >>>>>>>under certain conditions. > >>>>>>>Type "show copying" to see the conditions. > >>>>>>>There is absolutely no warranty for GDB. Type "show > >>>>>>> > >>>>>>> > >>warranty" for > >> > >> > >>>>>>>details. > >>>>>>>This GDB was configured as "i486-linux-gnu"...Using host > >>>>>>>libthread_db library "/lib/tls/i686/cmov/libthread_db.so.1". > >>>>>>> > >>>>>>>Core was generated by `/usr/local/sbin/openser -P > >>>>>>>/var/run/openser.pid -m > >>>>>>>32 -u root -g root'. > >>>>>>>Program terminated with signal 11, Segmentation fault. > >>>>>>> > >>>>>>>warning: Can't read pathname for load map: Input/output error. > >>>>>>>Reading symbols from /lib/tls/i686/cmov/libdl.so.2...done. > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libdl.so.2 > Reading symbols > >>>>>>> > >>>>>>> > >>>>>>>from /lib/tls/i686/cmov/libresolv.so.2...done. > >>>>>> > >>>>>> > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libresolv.so.2 Reading > >>>>>>>symbols from /usr/lib/i686/cmov/libssl.so.0.9.8...done. > >>>>>>>Loaded symbols for /usr/lib/i686/cmov/libssl.so.0.9.8 > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/lib/i686/cmov/libcrypto.so.0.9.8...done. > >> > >> > >>>>>>>Loaded symbols for /usr/lib/i686/cmov/libcrypto.so.0.9.8 > >>>>>>>Reading symbols from /lib/tls/i686/cmov/libc.so.6...done. > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libc.so.6 Reading symbols > >>>>>>> > >>>>>>> > >>>>>>>from /lib/ld-linux.so.2...done. > >>>>>> > >>>>>> > >>>>>>>Loaded symbols for /lib/ld-linux.so.2 Reading symbols from > >>>>>>>/usr/lib/libz.so.1...done. > >>>>>>>Loaded symbols for /usr/lib/libz.so.1 Reading symbols from > >>>>>>>/usr/local/lib/openser/modules/mysql.so...done. > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/mysql.so > >>>>>>>Reading symbols from /lib/tls/i686/cmov/libm.so.6...done. > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libm.so.6 Reading symbols > >>>>>>> > >>>>>>> > >>>>>>>from /usr/lib/libmysqlclient.so.14...done. > >>>>>> > >>>>>> > >>>>>>>Loaded symbols for /usr/lib/libmysqlclient.so.14 > Reading symbols > >>>>>>> > >>>>>>> > >>>>>>>from /lib/tls/i686/cmov/libcrypt.so.1...done. > >>>>>> > >>>>>> > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libcrypt.so.1 > >>>>>>> > >>>>>>> > >>Reading symbols > >> > >> > >>>>>>>from /lib/tls/i686/cmov/libnsl.so.1...done. > >>>>>> > >>>>>> > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libnsl.so.1 > >>>>>>> > >>>>>>> > >>Reading symbols > >> > >> > >>>>>>>from /usr/local/lib/openser/modules/sl.so...done. > >>>>>> > >>>>>> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/sl.so > >>>>>>>Reading symbols from > /usr/local/lib/openser/modules/tm.so...done. > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/tm.so > >>>>>>>Reading symbols from > /usr/local/lib/openser/modules/rr.so...done. > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/rr.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/maxfwd.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/maxfwd.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/usrloc.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/usrloc.so > >>>>>>>Reading symbols from > >>>>>>>/usr/local/lib/openser/modules/registrar.so...done. > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/registrar.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/textops.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/textops.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/xlog.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/xlog.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/enum.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/enum.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/uri_db.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/uri_db.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/options.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/options.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/auth.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/auth.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/auth_db.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/auth_db.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/avpops.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/avpops.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/mi_fifo.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/mi_fifo.so > >>>>>>>Reading symbols from > >>>>>>> > >>>>>>> > >>/usr/local/lib/openser/modules/presence.so...done. > >> > >> > >>>>>>>Loaded symbols for /usr/local/lib/openser/modules/presence.so > >>>>>>>Reading symbols from /usr/lib/libxml2.so.2...done. > >>>>>>>Loaded symbols for /usr/lib/libxml2.so.2 Reading symbols from > >>>>>>>/lib/tls/i686/cmov/libnss_compat.so.2...done. > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libnss_compat.so.2 > >>>>>>>Reading symbols from /lib/tls/i686/cmov/libnss_nis.so.2...done. > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libnss_nis.so.2 > >>>>>>>Reading symbols from > /lib/tls/i686/cmov/libnss_files.so.2...done. > >>>>>>>Loaded symbols for /lib/tls/i686/cmov/libnss_files.so.2 > >>>>>>>#0 0x080b7eed in fm_free (qm=0x8170b40, p=0xb7a8ffdf) at > >>>>>>>mem/f_malloc.c:130 > >>>>>>>130 frag->u.nxt_free=*f; > >>>>>>>(gdb) bt > >>>>>>>#0 0x080b7eed in fm_free (qm=0x8170b40, p=0xb7a8ffdf) at > >>>>>>>mem/f_malloc.c:130 > >>>>>>>#1 0xb7a8ce8f in handle_subscribe (msg=0x8186ca8, str1=0x0, > >>>>>>>str2=0x0) > >>>>>>> at subscribe.c:1099 > >>>>>>>#2 0x08052d56 in do_action (a=0x8178f48, msg=0x8186ca8) at > >>>>>>>action.c:883 > >>>>>>>#3 0x08051faa in run_action_list (a=0x8178e68, > msg=0x8186ca8) at > >>>>>>>action.c:131 > >>>>>>>#4 0x08054423 in do_action (a=0x81790d8, msg=0x8186ca8) at > >>>>>>>action.c:801 > >>>>>>>#5 0x08051faa in run_action_list (a=0x8178ba0, > msg=0x8186ca8) at > >>>>>>>action.c:131 > >>>>>>>#6 0x08054423 in do_action (a=0x817bc88, msg=0x8186ca8) at > >>>>>>>action.c:801 > >>>>>>>#7 0x08051faa in run_action_list (a=0x81770c8, > msg=0x8186ca8) at > >>>>>>>action.c:131 > >>>>>>>#8 0x080552c5 in run_top_route (a=0x81770c8, msg=0x0) at > >>>>>>>action.c:111 > >>>>>>>#9 0x08082769 in receive_msg ( > >>>>>>> buf=0x8140180 "SUBSCRIBE sip:oak@test-sip.uninett.no > >>>>>>> > >>>>>>> > >>SIP/2.0\r\nVia: > >> > >> > >>>>>>>SIP/2.0/UDP > >>>>>>>158.38.62.97:16472;branch=z9hG4bK-d87543-2b2f3c5a00212916 > >>>>>>> > >>>>>>> > >>-1--d87543-;rport\r\nMax-Forwards: > >> > >> > >>>>>>>69\r\nContact: <sip:gojensen@158.38.62.97:16472>"..., > >>>>>>> len=635, rcv_info=0xbfae1328) at receive.c:156 #10 > >>>>>>> > >>>>>>> > >>0x080b112a in > >> > >> > >>>>>>>udp_rcv_loop () at udp_server.c:465 > >>>>>>>#11 0x08070a08 in main_loop () at main.c:834 > >>>>>>>#12 0x08072045 in main (argc=9, argv=0xbfae14e4) at main.c:1399 > >>>>>>> > >>>>>>>// geir.o.jensen(a)uninett.no > >>>>>>> UNINETT AS, NO-7465 Trondheim, NORWAY > >>>>>>> Fax: +47 73 55 79 01, Phn: +47 73 55 79 00 > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>// geir.o.jensen(a)uninett.no > >>>>> UNINETT AS, NO-7465 Trondheim, NORWAY > >>>>> Fax: +47 73 55 79 01, Phn: +47 73 55 79 00 > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>> > >>>// geir.o.jensen(a)uninett.no > >>> UNINETT AS, NO-7465 Trondheim, NORWAY > >>> Fax: +47 73 55 79 01, Phn: +47 73 55 79 00 > >>> > >>> > >>> > >>> > >> > >> > > > > > > > >

3 3

RE: [Users] load testing presence server
by Toni Barata 27 Mar '07

27 Mar '07

Hi Daniel, Increasing the children from 4 to 16 did turn into same results (I was already using debug=3 and fork=yes). But setting debug=0 did real turn in much better results. Thanks a lot, Toni -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: segunda-feira, 26 de Março de 2007 16:53 To: Toni Barata Cc: users(a)openser.org Subject: Re: [Users] load testing presence server Hello, if you use exactly the same config, then turn debug to a lower level (e.g., 3), set fork=yes and children to 16 for better results. Cheers, Daniel On 03/26/07 18:48, Toni Barata wrote: > > Hi Anka, > > > > In a scenario using Openser as a Presence Sever (using the config file > example located in > http://openser.org/dokuwiki/doku.php/presence:configuration-file), the > execution of load tests (with sipp) with Subscribe Presence WInfo had > as result a maximum of 20 sessions per second (We are using a Intel(R) > Xeon(TM) CPU 3.00GHz with 2GB RAM), with lots of retransmissions if we > increase to values bigger than 20 (Openser dos not crash). Is there > any configurable parameter in the Presence module that allows > increasing this value? > > > > Best regards, > > Toni > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

3 2

[Users] Re: memory issues
by Papadopoulos Georgios 27 Mar '07

27 Mar '07

Hi Daniel, Does this problem affect 1.1.1 or only 1.2? Is there a patch to use for 1.1.1? thank you George -------------------------------------------------------------------------------- Hello Ovidiu, On 03/23/07 17:04, Ovidiu Sas wrote: > Hi Daniel, > > Can we backport this one to 1.2? already done, two minutes after the commit in trunk. Cheers, Daniel > > > Regards, > Ovidiu Sas > > On 3/22/07, Daniel-Constantin Mierla <daniel at voice-system.ro <http://openser.org/cgi-bin/mailman/listinfo/users> > wrote: >> Hello, >> >> the supposed fragmentation turned out to be a mem leak in pkg. Please >> take the latest SVN version and try again to see if you got same >> results. >> >> Thanks, >> Daniel >> >> On 03/19/07 18:52, Christian Schlatter wrote: >> > ... >> >>> The memory statistics indeed show a high number of memory fragments: >> >>> >> >>> before 'out of memory': >> >>> >> >>> shmem:total_size = 536870912 >> >>> shmem:used_size = 59607040 >> >>> shmem:real_used_size = 60106488 >> >>> shmem:max_used_size = 68261536 >> >>> shmem:free_size = 476764424 >> >>> shmem:fragments = 9897 >> >>> >> >>> after 'out of memory' (about 8000 calls per process): >> >>> >> >>> shmem:total_size = 536870912 >> >>> shmem:used_size = 4171160 >> >>> shmem:real_used_size = 4670744 >> >>> shmem:max_used_size = 68261536 >> >>> shmem:free_size = 532200168 >> >>> shmem:fragments = 57902 >> >>> >> >>>> >> >>>> You can try to compile openser with -DQM_JOIN_FREE (add it in DEFS >> >>>> variable of Makefile.defs) and test again. Free fragments should be >> >>>> merged and fragmentation should not occur -- processing will be >> >>>> slower. We will try for next release to provide a better solution >> >>>> for that. >> >>> >> >>> Compiling openser with -DQM_JOIN_FREE did not help. I'm not sure how >> >>> big of a problem this fragmentation issue is. >> >> What is the number of fragments with QM_JOIN_FREE after flooding? >> > >> > The numbers included above are with QM_JOIN_FREE enabled. >> > >> >>> Do you think it would make sense to restart our production openser >> >>> instances from time to time just to make sure they're not running >> >>> into this memory fragmentation limits? >> >> The issue will occur only when the call rate reaches the limits of >> >> the proxy's memory. Otherwise the chunks are reused. Transactions and >> >> avps are rounded up to be sure there will be minimized the number of >> >> different sizes for memory chunks. It wasn't reported too often, >> >> maybe that's why no big attention was paid to it. This memory system >> >> is in place since the beginning of ser. Alternative is to use sysv >> >> shared memory, but is much slower, along with libc private memory >> >> manager. >> > >> > I've done some more testing and the same out-of-memory stuff happens >> > when I run sipp with 10 calls per second only. I tested with >> > 'children=1' and I only could get through about 8200 calls (again >> > those 8000 calls / process). And this is with QM_JOIN_FREE enabled. >> > >> > Memory statistics: >> > >> > before: >> > shmem:total_size = 536870912 >> > shmem:used_size = 2311976 >> > shmem:real_used_size = 2335720 >> > shmem:max_used_size = 2465816 >> > shmem:free_size = 534535192 >> > shmem:fragments = 183 >> > >> > after: >> > shmem:total_size = 536870912 >> > shmem:used_size = 1853472 >> > shmem:real_used_size = 1877224 >> > shmem:max_used_size = 2465816 >> > shmem:free_size = 534993688 >> > shmem:fragments = 547 >> > >> > So I'm not sure if this is really a fragmentation issue. 10 cps surely >> > doesn't reach the proxy's memory. >> > >> > Thoughts? >> > >> > Christian >> > >> > >> > >> >> Cheers, >> >> Daniel >> >> >> >>> >> >>> thanks, >> >>> Christian >> >>> >> >>>> >> >>>> Cheers, >> >>>> Daniel >> >>>> >> >>>> On 03/18/07 01:21, Christian Schlatter wrote: >> >>>>> Christian Schlatter wrote: >> >>>>> ... >> >>>>>> >> >>>>>> I always had 768MB shared memory configured though, so I still >> >>>>>> can't explain the memory allocation errors I got. Some more test >> >>>>>> runs revealed that I only get these errors when using a more >> >>>>>> production oriented config that loads more modules than the one >> >>>>>> posted in my earlier email. I now try to figure out what exactly >> >>>>>> causes these memory allocation errors that happen reproducibly >> >>>>>> after about 220s at 400 cps. >> >>>>> >> >>>>> I think I found the cause for the memory allocation errors. As >> >>>>> soon as I include an AVP write operation in the routing script, I >> >>>>> get 'out of memory' messages after a certain number of calls >> >>>>> generated with sipp. >> >>>>> >> >>>>> The routing script to reproduce this behavior looks like (full >> >>>>> config available at >> >>>>> http://www.unc.edu/~cschlatt/openser/openser.cfg <http://www.unc.edu/~cschlatt/openser/openser.cfg> ): >> >>>>> >> >>>>> route{ >> >>>>> $avp(s:ct) = $ct; # commenting this line solves >> >>>>> # the memory problem >> >>>>> >> >>>>> if (!method=="REGISTER") record_route(); >> >>>>> if (loose_route()) route(1); >> >>>>> >> >>>>> if (uri==myself) rewritehost("xx.xx.xx.xx"); >> >>>>> route(1); >> >>>>> } >> >>>>> >> >>>>> route[1] { >> >>>>> if (!t_relay()) sl_reply_error(); >> >>>>> exit; >> >>>>> } >> >>>>> >> >>>>> An example log file showing the 'out of memory' messages is >> >>>>> available at http://www.unc.edu/~cschlatt/openser/openser.log <http://www.unc.edu/~cschlatt/openser/openser.log> . >> >>>>> >> >>>>> Some observations: >> >>>>> >> >>>>> - The 'out of memory' messages always appear after about 8000 test >> >>>>> calls per worker process. One call consists of two SIP >> >>>>> transactions and six end-to-end SIP messages. An openser with 8 >> >>>>> children handles about 64'000 calls, whereas 4 children only >> >>>>> handle about 32'000 calls. The sipp call rate doesn't matter, only >> >>>>> number of calls. >> >>>>> >> >>>>> - The 8000 calls per worker process are independent from the >> >>>>> amount of shared memory available. Running openser with -m 128 or >> >>>>> -m 768 does not make a difference. >> >>>>> >> >>>>> - The more AVP writes are done in the script, the less calls go >> >>>>> through. It looks like each AVP write is leaking memory (unnoticed >> >>>>> by the memory statistics). >> >>>>> >> >>>>> - The fifo memory statistics do not reflect the 'out of memory' >> >>>>> syslog messages. Even if openser does not route a single SIP >> >>>>> message because of memory issues, the statistics still show a lot >> >>>>> of 'free' memory. >> >>>>> >> >>>>> >> >>>>> All tests were done with openser SVN 1.2 branch on Ubuntu dapper >> >>>>> x86. I think the same is true for 1.1 version but I haven't tested >> >>>>> that yet. >> >>>>> >> >>>>> >> >>>>> Christian >> >>>>> >> >>> >> >>> >> > >> > >> > _______________________________________________ >> > Users mailing list >> > Users at openser.org <http://openser.org/cgi-bin/mailman/listinfo/users> >> > http://openser.org/cgi-bin/mailman/listinfo/users <http://openser.org/cgi-bin/mailman/listinfo/users> >> > >> >> _______________________________________________ >> Users mailing list >> Users at openser.org <http://openser.org/cgi-bin/mailman/listinfo/users> >> http://openser.org/cgi-bin/mailman/listinfo/users <http://openser.org/cgi-bin/mailman/listinfo/users> >> > Γιώργος Παπαδόπουλος Altec Telecoms R&D Department Πάτμου 14, 151 23 Μαρούσι τηλ.: +30 211 687 2933 fax.: +30 211 687 2911 e-mail: geop(a)altectelecoms.gr Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Users] pua_usrloc and expire time
by H Quintana 27 Mar '07

27 Mar '07

Hello, I'm using the pua_usrloc for IP phones with no presence options. When I restart the IP phone, the first REGISTER makes the module to send the PUBLISH message and the presentity is stored in the table with the proper expiration time. For the next REGISTER, it seems the module is using the old "expires" time instead taking the new value because the presentity expires some seconds after that. For further registrations, it's happening the same because the status is always offline after the second REGISTER. It doesn't matter the db_mode used for the usrloc module. Any suggestions? Thanks for your help, Humberto Quintana ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

2 1

[Users] I need save("location") but with some change ...
by Bodin Bruno 27 Mar '07

27 Mar '07

Hello, I currently use this code : if (is_method("REGISTER")) { if (!radius_www_authorize("mydomain")) { www_challenge("mydomain", "0"); exit; }; xlog("new user : $au"); save("location")) ; exit; }; But save function add a new item in location table with these param : Username, domain, contact ....user_agent, socket, methods. Username use by register module isn't Authentification's username how can I change it ? Is it possible to add more location for a user ? ( Like username and numer phone ) Thank for your help.

3 2

[Users] Watchers table not being updated
by Sebastian Murray-Roberts 27 Mar '07

27 Mar '07

Hi, I'm trying to set up OpenSER as a presence server without using force_active (ie to allow authorization). I have two x-lite clients which register with OpenSER and can subscribe to each other. I get a prompt on each client asking whether to authorize the other, but if I click yes no presence info is displayed. If I restart either of the x-lites the authorization message appears again. I know this is because the subs_status field of the watchers table is not being changed from "pending" to "active". If I run a packet capture when starting up x-lite, it sends the 200 OK to the notify provided by OpenSER, but when I click "allow" in the authorization box, nothing is sent to OpenSER. Has anyone got presence working with x-lite without using force_active? If so, is there anything I'm missing? My config file is as follows: debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=udp:192.168.1.107:5060 children=4 mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "maxfwd.so " loadmodule "textops.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "presence.so" loadmodule "avpops.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 3) modparam("rr", "enable_full_lr", 1) modparam("presence|^usrloc$", "db_url"," mysql://user:pass@localhost /openser") modparam("presence", "force_active", 0) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip: 192.168.1.107:5060") route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!is_method("REGISTER")) record_route(); if (loose_route()) { route(1); exit; }; if (!uri==myself) { route(1); exit; }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; }; if( is_method("PUBLISH|SUBSCRIBE")) { route(2); exit; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; } Thanks in advance, Sebastian -- Sebastian Murray-Roberts (B.Sc Hons) Server Developer Yeigo Communications http://www.yeigo.com Tel: +27 21 409 7807 Fax: +27 21 409 7050 Mobile: +27 72 183 5998 CONFIDENTIALITY NOTICE: E-MAIL This e-mail is confidential. If you receive it by mistake, please delete it and notify the sender immediately. If you are not the intended recipient, you may not use, disclose, distribute, copy or publish any information contained herein.

2 1

[Serusers] re: CONTACT field malformed in a 302 reply
by Jignesh Gandhi 27 Mar '07

27 Mar '07

Hello, I am using SER as a redirect server. Recently I came across an issue where the 302 sent back by SER is not liked by a soft switch. Particularly , the format of the CONTACT field according to the softswitch.. Here is an excerpt of the 302 reply send back ... Session Initiation Protocol Status-Line: SIP/2.0 302 MovedTemporarily Status-Code: 302 Resent Packet: True Suspected resend of frame: 73 Message Header To: <sip:12345@172.20.20.46>;tag= b27e1a1d33761e85846fc98f5f3a7e58.dfa5 SIP to address: sip:12345@172.20.20.46 SIP tag: b27e1a1d33761e85846fc98f5f3a7e58.dfa5 From: 172.20.20.46<sip:12345@172.20.20.46>;tag=161dda6e SIP Display info: 172.20.20.46 SIP from address: sip:12345@172.20.20.46 SIP tag: 161dda6e Via: SIP/2.0/UDP 10.99.99.140:9585 ;branch=z9hG4bK-d87543-730866470-1--d87543-;rport=9585 Call-ID: 95239a63f0347c53 CSeq: 1 INVITE * Contact: sip:12345@172.20.20.37:5060 * Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 172.20.20.46:5060 "Noisy feedback tells: pid=9550 req_src_ip=10.99.99.140 req_src_port=9585 in_uri=sip:12345@172.20.20.46out_uri= sip:12345@172.20.20.37:5060 via_cnt==1 the softswitch is wanting the contact field with <sip:xxxx@xxx.xxx.xxx.xxx:5060> like a FROM or TO URI. Here is ser.cfg part that does the sl_send_reply(); # do a stateless redirect, if return code is correct if (method=="INVITE") { xlog("L_INFO", "SourceIP <%is> \n"); xlog("L_INFO", "From-uri<%fu>, r-uri <%ru> \n"); if (!gl_redirect()) { sl_send_reply("480", "TemporarilyUnavailable"); xlog("L_WARN", "Sending a 480 response with r-uri <%ru>\n"); break; } else { sl_send_reply("302", "MovedTemporarily"); xlog("L_DBG", "Sending a 302 response with r-uri <%ru>\n"); break; } }; Any help is appreciated.** thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com

3 2

Re: [Serusers] SER - Config files
by samuel 27 Mar '07

27 Mar '07

First of all, please CC to the list!!! it->refers to the routing script you are using that by default is ser.cfg. You can change it starting ser with -f parameter and placing after the config file. I recomend you to look into iptel.org and get the general overview of the project. samuel. 2007/3/26, Goh, Tee-Yong <gohty(a)m1.com.sg>: > > > > Hi Samuel, > > Thank you for your support. > Therefore, you have mentioned that ser.cfg is just an example and do not > need to modify. However, as you have mentioned "You would definetely need to > modify it to make SER behave as your requirements dictates", what does "it" > are you refering to? The files in the SER or just ser.cfg? > > Please advice. > > Thank you. > > Regards, > Howard > > > -----Original Message----- > From: samuel [mailto:samu60@gmail.com] > Sent: Monday, March 26, 2007 5:10 PM > To: Goh, Tee-Yong > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] SER - Config files > > > ser.cfg is, usually, the routing script that dictates how SER behaves > on receiving SIP requests and responses. > It's just an example that should work wihtout modificaction and is > meant as a starting point to get into SER. You would definetely need > to modify it to make SER behave as your requirements dictates. > > look for more information in iptel.org and google... > > Good Luck! > > sam > > 2007/3/25, Goh, Tee-Yong <gohty(a)m1.com.sg>: > > > > > > > > Hi all, > > > > May i seek your expertise advice. I have found the config file name > > "ser.cfg" and "ser.cfg~" in the SER-0.9.6. > > However, i do not know what are they used for? Can i make any modification > > to them? > > > > Please advice. > > > > Thank you. > > > > Regards, > > Howard > > > > ________________________________ > > This email (including any attachment) is subject to the following > > disclaimer: > > > http://m1.com.sg/M1/misc/disclaimer________________________________ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ________________________________ > This email (including any attachment) is subject to the following > disclaimer: > http://m1.com.sg/M1/misc/disclaimer________________________________ >

1 0

[Users] Openser Compilation problem using Optimized GCC
by Sergio Gutierrez 27 Mar '07

27 Mar '07

Hello everybody. I discovered another issue when compiling openser on Solaris 10 running on Sparc. If I use standard gcc version 3.4.3, it compiles and runs Ok when it is compiled on 32 bits (Standard compilation without special command line options). But If I use the GCC edition for SPARC Systems (GCCFS) by Sun Microsytems (which is based on GCC 4.0.3 and is an optimized edition for Solaris on SPARC), openser fails to start complaining about table versions. Below appears the log output, and the backtrace of the generated core file: WARNING: no fork mode 0(1120) fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(1120) fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560 0(1120) fm_malloc_init(fc800000, 33554432), start=fc800000 0(1120) shm_mem_init: success 0(0) core dump limits set to 0 0(0) DEBUG:add_avp_galias: registering <serial_branch> for avp id <16725044> 0(0) INFO: statistics manager successfully initialized 0(0) DEBUG: init_mod: sl 0(0) StateLess module - initializing 0(0) DEBUG: MD5 calculated: 224bba3fcb82b3a99110f369fb8df55a 0(0) DEBUG: register_stats: sl 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) INFO:ul_init_locks: locks array size 4096 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: found <db_use_table> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_init> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_close> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_query> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_fetch_result> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_raw_query> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_free_result> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_insert> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_delete> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_update> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_replace> in module mysql [/toip2/lib/openser/modules/] 0(0) DEBUG: register_stats: usrloc 0(0) DEBUG: register_mi: usrloc 0(0) DEBUG: init_mod: registrar 0(0) registrar - initializing 0(0) find_cmd_export_t: found <load_sl>(0) in module sl [/toip2/lib/openser/modules/] 0(0) find_cmd_export_t: found <ul_bind_usrloc>(1) in module usrloc [/toip2/lib/openser/modules/] 0(0) DEBUG: register_stats: registrar 0(0) DEBUG: init_mod: tm 0(0) TM - initializing... 0(0) Call-ID initialization: '47395976' 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) Warning: uac does not spread across the whole hash table 0(0) DEBUG: MD5 calculated: b5fd95724ef23fee9ce49ee6c27239d6 0(0) DEBUG: MD5 calculated: d0730e359e0208514db6efb58e0ac8b9 0(0) DEBUG: register_stats: tm 0(0) DEBUG: register_mi: tm 0(0) DEBUG: init_mod: mysql 0(0) mysql: MySQL client version is 5.0.37 0(0) DEBUG: init_mod: auth 0(0) AUTH module - initializing 0(0) find_cmd_export_t: found <load_sl>(0) in module sl [/toip2/lib/openser/modules/] 0(0) xl_parse_name: name [s:rpid] index [0] 0(0) xl_parse_vname: avp [s:rpid] 0(0) DEBUG: init_mod: auth_db 0(0) AUTH_DB module - initializing 0(0) find_mod_export: found <db_use_table> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_init> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_close> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_query> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_fetch_result> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_raw_query> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_free_result> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_insert> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_delete> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_update> in module mysql [/toip2/lib/openser/modules/] 0(0) find_mod_export: found <db_replace> in module mysql [/toip2/lib/openser/modules/] 0(0) find_cmd_export_t: found <bind_auth>(0) in module auth [/toip2/lib/openser/modules/] 0(0) find_cmd_export_t: found <load_sl>(0) in module sl [/toip2/lib/openser/modules/] 0(0) DEBUG: init_mod: auth_radius 0(0) auth_radius - Initializing 0(0) auth_radius: No `Cisco' vendor in Radius dictionary 0(0) find_cmd_export_t: found <bind_auth>(0) in module auth [/toip2/lib/openser/modules/] 0(0) DEBUG: init_mod: acc 0(0) ACC - initializing 0(0) find_cmd_export_t: found <load_tm>(0) in module tm [/toip2/lib/openser/modules/] 0(0) xl_parse_name: name [Via] index [-2] 0(0) INFO:xl_parse_name: using hdr type (1) instead of <Via> 0(0) fixing radius_www_authorize, line 63 0(0) fixing www_challenge, line 65 0(0) fixing save, line 69 0(0) db_init: Connection 'mysql://openser:openserrw@localhost/openser' not found in pool 0(0) new_connection: Opening MySQL connection: mysql://openser:openserrw@localhost/openser 0(0) new_connection: Connection type is Localhost via UNIX socket 0(0) new_connection: Protocol version is 10 0(0) new_connection: Server version is 5.0.37-log 0(0) table_version(): Invalid type (2) or nul (0) version columns for location 0(0) register_udomain(): Error while querying table version 0(0) pool_remove: Removing connection from the pool 0(0) domain_fixup(): Error while registering domain 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 69 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration Segmentation Fault (core dumped) # GDB backtrace (gdb) bt #0 0x000305a8 in process_count () #1 0x00030518 in kill_all_children () #2 0x00033ab8 in main () Is there any incompatibility when using GCC 4.X to compile openser? Thanks in advance for your help. Sergio Gutiérrez

1 0

[Users] "client did not present a certificate"(OpenSER 1.2 + EyeBeam 1.5 + TLS) !! need help!!
by yanlin 27 Mar '07

27 Mar '07

Hi, all, i have been trying to test TLS support on OpenSER with EyeBeam client, but in no vain. OpenSER keep complaining that "client did not present a certificate". really need help! thanks in advance. here is some info of my environment: 1) OpenSER 1.2 and EyeBeam 1.5 2) run "openserctl tls rootCA", create "cacert.pem" under /etc/openser/tls/rootCA/. 3) run "openserctl tls userCERT", create "user-calist.pem user-cert.pem user-cert_req.pem user-privkey.pem" under /etc/openser/tls/user/. 4) i have set openser.cfg as follow: disable_tls = 0 listen = tls:172.22.14.61:5061 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = TLSv1 tls_certificate = /etc/openser/tls/user/user-cert.pem" tls_private_key = "/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/etc/openser/tls/user/user-calist.pem" 5) copy "/etc/openser/tls/rootCA/cacert.pem" created at step 2) to EyeBeam clinet machine, which was a Windows XP machine, run "certmrg.msc" there, import this certificate to WindowXP "root certificate store". when run ... error occur. OpenSER complaint that "client did not present a certificate", and EyeBeam receive a "503 certificate name mismath". Any advise will be very appreciate !! yan lin yanlin(a)fortinet.com 2007-3-26

3 3

[Serusers] New search-page
by Marcin Pazdro 26 Mar '07

26 Mar '07

Hi all, I'm going to improve the documentation search page (or rather to create new search engine). Here: http://www.iptel.org/ser_documentation_dynamic_search_page_functional_speci… I have placed a functional specification that describes main guidelines of new search page. I would like to get some feedback from you. Please let me know if you have some comments, tips or questions. pazdro :)

2 1

[Users] Openser fails when compiled on Solaris 64bit
by Sergio Gutierrez 26 Mar '07

26 Mar '07

Hi I am trying to run Openser compiled on 64 bits on a SPARC Machine running Solaris 10. When I create a simple configuration file for testing radius integration, Openser starts to consume the whole memory reservation (-m parameter) and fails with segmentation fault error. after several tests, I have found that the error is caused by save() function (registrar module). This is the main route my configuration file: route { if(method=="REGISTER") { if(!radius_www_authorize("")) { www_challenge("", "0"); return; }; if(!save("location")) { sl_reply_error(); }; return; } else { } } Thanks in advance for your help. Sergio G.

2 1

[Users] load testing presence server
by Toni Barata 26 Mar '07

26 Mar '07

Hi Anka, In a scenario using Openser as a Presence Sever (using the config file example located in http://openser.org/dokuwiki/doku.php/presence:configuration-file), the execution of load tests (with sipp) with Subscribe Presence WInfo had as result a maximum of 20 sessions per second (We are using a Intel(R) Xeon(TM) CPU 3.00GHz with 2GB RAM), with lots of retransmissions if we increase to values bigger than 20 (Openser dos not crash). Is there any configurable parameter in the Presence module that allows increasing this value? Best regards, Toni

2 1

[Users] Not able to access avp variables in reply route
by Asterisk Expert 26 Mar '07

26 Mar '07

Hello, I'm trying to implement custom CDR using OpenSER. What I do is, insert a record in cdr table on invite request. I get the id of the inserted row using LAST_INSERT_ID() function of mysql in avp_db_query and store in avp variable rowid. I set connect time in reply route when I receive 200 OK for Invite request. When I get 200 OK for BYE request, I update the cdr record and store connect time, disconnect time, duration, etc. But at this place(reply route) I get the null rowid so as connect time. According to tm module's onreply_avp_mode parameter, I should be able to see the values set from request route in reply route if I set that parameter to 1 but it has no effect. I can not use either avp variables or script variables as transaction variables to record custom cdr. Can anyone guide me implementing this scenario or tell me what I've done wrong? -- Regards Ruchir Brahmbhatt Ecosmob Technologies

2 3

[Users] openserctl alias show ???
by Daryl Sanders 26 Mar '07

26 Mar '07

How can I view a list of aliases via openserctl? I tried "openserctl alias show", however this seems to return a list of registered UA's not aliases. - Daryl

4 5

[Users] Can not access script variables in openser 1.2.0
by Asterisk Expert 26 Mar '07

26 Mar '07

Hello, Sorry last mail sent accidentally. I've initialized some SCRIPT VARIABLES in request route. I modify the values of some of these SCRIPT VARIABLES in reply route. But I'm not able to get value of some of those variables in either request or reply route. Any idea? -- Regards Ruchir Brahmbhatt

2 2

[Serusers] Changing expires in 200 OK for Register
by Baljeet Gill 26 Mar '07

26 Mar '07

Hi All, Is there a way to change the "expires" value in the 200 response (sent by SER) for REGISTER? Thanks, Baljeet To access the latest news from CSR copy this link into a web browser: http://www.csr.com/email_sig.php To get further information regarding CSR, please visit our Investor Relations page at http://ir.csr.com/csr/about/overview

2 1

[Users] rewritehost
by Kro Kamil 26 Mar '07

26 Mar '07

hello, suppose that i am calling xxxxx(a)10.0.0.9 and i want to send this call to 10.0.0.10. is there any difference between rewritehost(10.0.0.10); and $ruri=sip:xxxxx@10.0.0.10; thanks _________________________________________________________________ Live.com'u deneyin: çevrimiçi dünyanızı bir araya getirin; haberler, spor, hava durumu ve çok daha fazlası. http://www.live.com/getstarted

2 1

[Serusers] FIFO file
by Meidiana Ten 26 Mar '07

26 Mar '07

Hi, I'd like to know if we had 2 SER config file (home proxy and voicemail server) in same machine, do I have to setup different fifo file? For example, SER home proxy has fifo file /tmp/ser_fifo so what fifo I have to use for SER voicemail server? And how do I create the other fifo file? Thanx Regards, Meidiana --------------------------------- Don't be flakey. Get Yahoo! Mail for Mobile and always stay connected to friends.

5 4

[Users] No audio when openSER is forworded to Asterisk server in the Same LAN network.
by raviprakash sunkara 26 Mar '07

26 Mar '07

Hello Users . For last 2 days onwards i'm fighting with this promlem, In integrating with Asterisk and openSER, No audio when openSER is forworded to Asterisk server in the Same LAN network. OpenSER and Asterisk are in the Same LAN network. With NAT , i forward the the port to 192.168.2.11 of openSER sip port and rtp port. With out NAT is Working Fine , openSER ip is 192.168.2.11 Asterisk is in 192.168.2.75 alias=sip.hyperion-tech.com if ( method =="INVITE") { if(uri=~"sip:1234567@sip.hyperion-tech.com || uri=~" sip:12345678@sip.hyperion-tech.com") { route(12); }; }; route[12] { rewrtitehostport("192.168.2.75:5060"); # ASTERISK SERVER if(!t_relay()) { sl_send_error(); exit; }; } -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

1 0

Re: [Users] Not able to access avp variables in reply route
by Asterisk Expert 26 Mar '07

26 Mar '07

Yes I know radius accounting will work but I have to do it without radius. So I'm doing it manually using avp_db_query. On 3/26/07, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > Hi > > U can use the Radius for account > > with Acc of radius some avp functions are there > > u means > like this > onroute_reply[] > { > if( status==200) > { > i f ( method == "invite" ) > accouting start of avp module > if methosd == bye) > acctoung stop of avp module > } > } > > > On 3/26/07, Asterisk Expert < asterisk.expert(a)gmail.com> wrote: > > > Hello, > > I'm trying to implement custom CDR using OpenSER. What I do is, insert a > > record in cdr table on invite request. I get the id of the inserted row > > using LAST_INSERT_ID() function of mysql in avp_db_query and store in avp > > variable rowid. I set connect time in reply route when I receive 200 OK for > > Invite request. When I get 200 OK for BYE request, I update the cdr record > > and store connect time, disconnect time, duration, etc. But at this > > place(reply route) I get the null rowid so as connect time. According to tm > > module's onreply_avp_mode parameter, I should be able to see the values set > > from request route in reply route if I set that parameter to 1 but it has no > > effect. I can not use either avp variables or script variables as > > transaction variables to record custom cdr. Can anyone guide me implementing > > this scenario or tell me what I've done wrong? > > -- > > Regards > > Ruchir Brahmbhatt > > Ecosmob Technologies > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com > M:+91 9985077535 > www.hyperion-tech.com > Client and Parent company :- www.august-networks.com -- Regards Asterisk Expert

1 0

[Users] No audio when openSER is forwarded to Asterisk server in the Same LAN network.
by raviprakash sunkara 26 Mar '07

26 Mar '07

Hello Users . For last 2 days onwards i'm fighting with this promlem, In integrating with Asterisk and openSER, No audio when openSER is forworded to Asterisk server in the Same LAN network. OpenSER and Asterisk are in the Same LAN network. With NAT , i forward the the port to 192.168.2.11 of openSER sip port and rtp port. With out NAT is Working Fine , openSER ip is 192.168.2.11 Asterisk is in 192.168.2.75 alias=sip.hyperion-tech.com if ( method =="INVITE") { if(uri=~"sip:1234567@sip.hyperion-tech.com || uri=~" sip:12345678@sip.hyperion-tech.com") { route(12); }; }; route[12] { rewrtitehostport("192.168.2.75:5060"); # ASTERISK SERVER if(!t_relay()) { sl_send_error(); exit; }; } -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

1 0

[Users] VOIP Patent Bomb Goes Off In U.S.
by michael papet 26 Mar '07

26 Mar '07

For those of you that haven't heard, here's a nice rundown on Verizon's patent enforcement activities. http://ipurbia.com/2007/03/verizon-patent-analysis.html This news makes me sad.

2 1

[Users] Openser not starting
by Knight Tiger 26 Mar '07

26 Mar '07

Hi I ran [root@openser ~]# openser start Listening on udp: A.B.C.D [A.B.C.D]:5060 udp: 127.0.0.1 [127.0.0.1]:5060 tcp: A.B.C.D [A.B.C.D]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 Aliases: tcp: localhost.my.domain:5060 tcp: localhost:5060 tcp: dhcpwl009.testdomain.com:5060 udp: localhost.my.domain:5060 udp: localhost:5060 udp: dhcpwl009.testdomain.com:5060 but I dont see any process running with the ps command So I checked the config files using -c [root@openser ~]# openser -c Listening on udp: A.B.C.D [A.B.C.D]:5060 udp: 127.0.0.1 [127.0.0.1]:5060 tcp: A.B.C.D [A.B.C.D]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 Aliases: tcp: localhost.my.domain:5060 tcp: localhost:5060 tcp: dhcpwl009.testdomain.com:5060 udp: localhost.my.domain:5060 udp: localhost:5060 udp: dhcpwl009.testdomain.com:5060 config file ok, exiting... 0(16175) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy [root@openser ~]# So the config files are okay but then openser does not start. So I tried root@openser % openserctl moni [cycle #: 1; if constant make sure server lives] and the process does not die.. I had to manually kill the process. I checked the subscriber table on the database and it was corect. I had registered a few users and it reflected them. The Postgres database is running. What am I doing wrong ? Any help would be appreciated. Thanks Knight

2 1

[Users] OpenSER as a Load Balancer for Asterisk and RE-INVITEs
by Edoardo Serra 25 Mar '07

25 Mar '07

Hi all, I have an OpenSER acting as a registrar server and as a load balancer (with dispatcher module) for a cluster of PSTN Gateways based on Asterisk. When an inbound call is coming from the PSTN, Asterisks dial SIP/user@openser and the call is properly forwarded to the correct user in OpenSER location table But, if the user wants, for example, to put the call On-Hold, its user agent sends a RE-INVITE to OpenSER. My problem is that OpenSER (correctly) dispatch the call to a random Asterisk GW, and, if it's not the GW which is handling the call, On-Hold does not work and another call is originated. Tnx in advance for help Edoardo My openser.cfg follows # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # ----------- global configuration parameters ------------------------ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" #uid=nobody #gid=nobody # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/exec.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("usrloc", "timer_interval", 120) modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "db_url", "mysql://user:pass@192.168.252.5/db") modparam("uri_db", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "max_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("nathelper", "natping_interval", 20) # Ping interval 20 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("dispatcher", "force_dst", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) { sl_send_reply("405", "Method Not Allowed"); exit; } if (!method=="REGISTER") { record_route(); }; if ((src_ip==ip.of.asterisk.1) || (src_ip==ip.of.asterisk.1)) { if (!lookup("location")) { sl_send_reply("486", "Busy here"); exit; }; if (!t_relay()) { sl_reply_error(); }; exit; }; if (nat_uac_test("3")) { if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) { fix_nated_contact(); force_rport(); setflag(6); # Mark as NATed } } if (method=="REGISTER") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_from()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } } # loose-route processing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); exit; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); if ((src_ip!=ip.of.asterisk.1) && (src_ip!=ip.of.asterisk.2)) { ds_select_dst("1", "0"); } if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 2

[Users] roadmap web page
by Ovidiu Sas 24 Mar '07

24 Mar '07

Hi, Now that openser 1.2 was released, it is time to update the roadmap :) http://www.openser.org/mos/view/Roadmap/ Regards, Ovidiu Sas

1 0

[Users] Re: Openser not starting (Knight Tiger)
by vallimamod abdullah 24 Mar '07

24 Mar '07

Hello Knight, On 3/24/07, "Knight Tiger" <caugar(a)gmail.com> wrote on behalf of users-request(a)openser.org <users-request(a)openser.org>: > [...] > So the config files are okay but then openser does not start. > [...] Have you had a look on the logs ? There are sometimes config errors that are not detected during the check but only at startup. So you can try with: debug=9 fork=no log_stderror=yes in your config and run openser directly (i.e. not via the init script) to get verbose output to the console. -- Regards, -vma .

1 0

[Users] Can not access script variables
by Asterisk Expert 24 Mar '07

24 Mar '07

Hello, I've initialized some script variables in request route. I modify the values of some of these SCRIPT VARIABLES -- Regards Asterisk Expert

1 0

[Users] Comparable regex function in openser
by CARTWRIGHT, CORY C (ATTSNET) 23 Mar '07

23 Mar '07

Hello all, I am attempting to add a line to the SDP of a reply using subst. In perl I can do it like this. $test1 =~ s/(.+)\sRTP\/AVP\s18\s(.+)/$1 RTP\/AVP 18 $2\na=fmtp:18 annexb=no\n/g; This ensures I have the whole "m=audio 28104 RTP/AVP 18 101 19" string in the replacement I started with this: subst('/(.*) RTP\/AVP 18 (.*)/\1 RTP\/AVP 18 \2\r\na=fmtp:18 annexb=no\r\n/g'); As most of you probably can see this does not work. Does anyone know if this is possible? Thanks. This e-mail and any files transmitted with it are the property of AT&T Connecticut, are confidential, and are intended solely for the use of the individuals or entities to whom it is addressed. If you are not a named recipient, or otherwise have reason to believe that you have received this message in error, please notify the sender at phone 203-420-7225, and delete this message immediately from your computer. Any other use, retention, dissemination, forwarding, printing, or copying of this e-mail is strictly prohibited.

2 3

[Users] OpenSER 1.2 and NAT
by Andreas Granig 23 Mar '07

23 Mar '07

Hi, When migrating my config from 1.1 to 1.2, far-end NAT traversal doesn't work anymore. Here is what I do: modparam("usrloc", "nat_bflag", 6) ... # register handling: if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { xlog("L_INFO", "NATed REGISTER\n"); fix_nated_register(); setbflag(6); } if(!save("location")) { # error handling ... } The log message is printed, so NAT is detected. The received-column in the DB also shows the correct NAT-address, but the flag-column is always 0. Any advice? Regards, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

2 3

[Users] Openser 1.2.0 and exec_dset
by Tobias Lindgren 23 Mar '07

23 Mar '07

Hi all, I'm using exec_dset to run a script which takes the Request-URI and returns a new Request-URI based on the number dialed. In ser.cfg I'm doing this: if (exec_dset("lookup.sh \"$$SIP_RURI\"")) { # Route call } else { # Reply with a cause } Recently a customer started to use SIP-X, and his INVITES with Request-URIs looks like this: sip:1001@192.168.0.1:5060;sipx-noroute=Voicemail Ser responds with "500 Server Internal Error" to these INVITEs, which is due to that my script fails. If I run OpenSER in the foreground with debugging I can see this: 0(20509) exec:w_exec_dset: executing [lookup.php "$SIP_RURI"] 0(20509) exec_str: rtrim sh: line 1: sipx-noroute=Voicemail: command not found How do I correctly escape the SIP_RURI in my openser.cfg to prevent my script from failing? Or is the RURI to long or something? If I run my script from commandline with the request uri, everything is fine. Br, /Tobias

2 6

[Users] PRESENCE module issues
by Geir O. Jensen 23 Mar '07

23 Mar '07

Setup; OpenSER 1.2.0 Fresh install (no SERWEB, all the rest) running on MySQL 5.0.22 (Ubuntu 6.06) Two subscribers that publish status, but neither can see status of the other... They are both in the presentity table and listed in the watchers/active_watchers table. When I change status in the client the following occurs on the server; PRESENCE:update_presentity: querying presentity PRESENCE:get_subs_dialog:querying database table = active_watchers PRESENCE: get_subs_dialog:The query for subscribtion for [user]= gojensen,[domain]= test-sip.uninett.no for [event]= presence returned no result PRESENCE:query_db_notify: Could not get subs_dialog from database PRESENCE:update_presentity: Could not send Notify PRESENCE: handle_publish: sip-etag: [SIP-ETag: a.1174552555.9851.2^M /31] I looked through the archives and google very quickly and found something about wrong record sizes and types (for postgresql)... However neither seems to apply in my case. Has anyone got PRESENCE module working, or more importantly, does anyone know how to solve the above issue? // sip:gojensen@uninett.no | h323: 004710012 pstn: +47 73 55 79 23 | fax: +47 73 55 79 01

4 6

Re: [Users] OpenSER crashes when attempting to use null avp variable
by Daniel-Constantin Mierla 23 Mar '07

23 Mar '07

Hello, On 03/23/07 13:59, Asterisk Expert wrote: > Hello Daniel, > I just figured out. I forgot to set db url for avpops :)) > But in this case openser should not start and give error. not? does now, thanks for reporting (just committed on svn). > Instead it started and when executed mysql query, it crashed. > I have one more query. I read that if we set onreply_avp_mode=1 in tm > module, we can access avp variables in reply routes. I have done the > same. I set some avp value in request route and update mysql db based > on that value in reply route but I'm getting null :( > What do you think will be the problem? I will check it. Cheers, Daniel > > > On 3/23/07, *Daniel-Constantin Mierla* <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro> > wrote: > > Hello, > > i tried to reproduce, but I got no same result. If the avp is > null, the > returned values is false in comparisons. > > Could you send log messages and backtrace? > > Cheers, > Daniel > > > On 03/23/07 10:37, Asterisk Expert wrote: > > Hello, > > I'm using openser 1.2.0 . I have following section in my > openser.cfg. > > > > avp_db_query("SELECT active, accountcode FROM openser_ip_auth WHERE > > ip_addr='$src_ip'","$avp(s:active);$avp(s:accountcode)"); > > xlog("L_INFO", "\n$avp(s:active):::$avp(s:accountcode)\n"); > > if (!avp_check("$avp(s:active)","eq/1/I")) > > { > > xlog("L_ERR", "$si [$Tf]: $rm $fu -> $ru: REGISTER: bad > > credentials, ask for credentials\n"); > > > > #authentication failed > > www_challenge("", "0"); > > setflag(1); > > return; > > } > > Basically I'm doing IP based authentication. I have used the similar > > method in version 1.1 but openser 1.2.0 crashes if the query in > > avp_db_query returns no rows which keeps those two avp variables > null. > > Is this a bug or I'm doing something wrong? > > I also tried "==" operator for comparison but it crashes whenever it > > is trying to access one of those two variables. > > -- > > Regards > > Ruchir Brahmbhatt > > Ecosmob Technologies > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > Regards > Ruchir Brahmbhatt

2 2

[Users] memory issues
by Christian Schlatter 23 Mar '07

23 Mar '07

I'm doing sipp performance measurements with openser 1.2 and I was surprised on how much memory the openser processes do allocate when confronted with a high call-per-second load. I observed the following behavior: One sipp instance creates SIP requests that get routed through openser and forwarded to another sipp instance running in UAS mode. The sipp scenario looks like INVITE ----------> 100 <---------- 180 <---------- 200 <---------- ACK ----------> Pause [ 10.0s] BYE ----------> 200 <---------- Nothing special here. The openser instance record routes the SIP requests and forwards them in transaction stateful mode (t_relay) to the sipp UAS instance. The routing script looks like: route{ if (!method=="REGISTER") record_route(); if (loose_route()) route(1); if (uri==myself) rewritehost("xx.xx.xx.xx"); route(1); } route[1] { if (!t_relay()) sl_reply_error(); exit; } ( xx.xx.xx.xx is the IP of sipp UAS) Running sipp with a call rate of 500 cps results in 45 MB of memory allocation per openser process after a few seconds. What could cause this behavior, message queuing maybe? But then again having a message queue of 40+ MB seems to be unrealistic. And the memory consumption goes up the longer the test runs until openser dies because of out-of-memory. I observed the same behavior with openser 1.1 and 1.2. Is this a memory leak or did I miss some settings? And what's the best method to debug memory issues with openser? thanks in advance, Christian

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[Users] No Audio when integrating openSER and Asterisk , in NAT
by raviprakash sunkara 23 Mar '07

23 Mar '07

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - -> Asterisk (voicemail sserver ). Please help me -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

1 0

[Users] OpenSER crashes when attempting to use null avp variable
by Asterisk Expert 23 Mar '07

23 Mar '07

Hello, I'm using openser 1.2.0. I have following section in my openser.cfg. avp_db_query("SELECT active, accountcode FROM openser_ip_auth WHERE ip_addr='$src_ip'","$avp(s:active);$avp(s:accountcode)"); xlog("L_INFO", "\n$avp(s:active):::$avp(s:accountcode)\n"); if (!avp_check("$avp(s:active)","eq/1/I")) { xlog("L_ERR", "$si [$Tf]: $rm $fu -> $ru: REGISTER: bad credentials, ask for credentials\n"); #authentication failed www_challenge("", "0"); setflag(1); return; } Basically I'm doing IP based authentication. I have used the similar method in version 1.1 but openser 1.2.0 crashes if the query in avp_db_query returns no rows which keeps those two avp variables null. Is this a bug or I'm doing something wrong? I also tried "==" operator for comparison but it crashes whenever it is trying to access one of those two variables. -- Regards Ruchir Brahmbhatt Ecosmob Technologies

2 1

[Users] are there sites that use OpenSER server?
by 김형선 23 Mar '07

23 Mar '07

Hello. Are there sites that use OpenSER server? Like 'iptel.org' (don't care commercial site or not) thanks. 새로운 기부문화의 씨앗, 해피빈

4 3

[Serusers] (no subject)
by KUMAR 23 Mar '07

23 Mar '07

thank you, that was helpful. kumar

1 0

[Serusers] Sip Softphone
by Jai Rangi 23 Mar '07

23 Mar '07

Hello, We are looking for a sip based softphone that can be customized according to our need. Something like Skype, so that use have to enter only username and password. If it can be open source that will be great. If not we can also pay some one who can do that for us. Please let me know if any one can do this or suggest me where to find that. Thank you, -JP

4 3

[Users] openser1.2.0+diameter
by 赵俊杰 23 Mar '07

23 Mar '07

Hello,ereryone I want to use diameter for sip accounting under ipv6.But I don't know how to configure it .Who can give me some suggestion? Thank you very much.My sip server is openser 1.2.0. Zhao Junjie 2007-03-22

2 1

[Serusers] groups
by KUMAR 23 Mar '07

23 Mar '07

yes, i was talking about the 'group' module. I still see it in 0.9.6. Moreover, there's even a table 'grp' in the ser database. Even if it has been removed now, i would like to know what it is/was used for. thank you kumar

2 1

[Serusers] Groups
by KUMAR 23 Mar '07

23 Mar '07

can you please tell me or point me to a link as to what are groups used for in SER. thank you kumar

2 1

[Serusers] INFO: Prague User Group Meetings Presentations Now Available
by Greger V. Teigre 23 Mar '07

23 Mar '07

To all of you who participated in the Prague User Group meetings on Monday and Wednesday: It was great to meet you, get a face behind the the name, and thanks for all the good discussions we had! There were people I didn't have the chance to talk to, I'm sure we get the opportunity another time :-) To those of you who did not have the possibility to attend: The presentations from Monday can be found on iptel.org in PDF format, together with a brief summary. http://www.iptel.org/ser/usergroup/prague/summary The Wednesday meeting was available as an mp3 stream and many people listened in and participated on IRC. http://www.iptel.org/ser_workshop The full workshop will soon be made available, as well as a summary of the conclusions and most important topic. A separate announcement will be made when available. g-)

2 1

[Serusers] SIP Accounting Module problem
by POON Ben 23 Mar '07

23 Mar '07

Hi folks, I'm trying to get my accounting services working, but am having trouble doing so... I am using the sample ser.cfg file, and am having a problem loading the acc.so module. In the ser.cfg file, I try added the line: loadmodule "/usr/sfw/lib/ser/modules/acc.so" to the bottom of the loadmodule section. I haven't done any modparam or scripting yet, even so, when I restart the ser service, I get: Starting SER : cat: cannot open /var/run/ser.pid. If I comment or remove the loadmodule line, ser will load again. Is there anything else I need to add or configure before loading this module? I went to /usr/sfw/lib/ser/modules to make sure acc.so is there, and it is. Any help is greatly appreciated. Thank you very much. Ben

3 7

[Users] OpenSER not sending responses to IP in via header
by Tim Madorma 23 Mar '07

23 Mar '07

Hi, I have an OpenSER behind a BIG IP load balancer. The scenario is as follows: 1) The UAC will initiate the call with an INVITE that has the REQ URI set to the IP of the BIG IP. 2) The OpenSER has an alias configured for the IP address of the BIG IP so it receives the incoming INVITE, but when it sends a 100 response, it does not send it to the UAC address that is specified in the VIA header, rather it sends it to the source IP (which is the IP of the BIG IP load balancer). This is undesirable. 3) After the UAS receives the proxied INVITE, it responds with a 100, then answers with a 200. 4) When the OpenSER receives the 100, it forwards this back to the BIG IP load balancer! First, I am surprised that it fowards a 100 at all, and second it is undesirable to send any responses to the BIG IP load balancer since the via header indicates the IP of the UAC. 5) When the OpenSER receives the 200, it forwards it back to the BIG IP also.. again, I don't know why it isn't sending to the IP address of the via - which is that of the UAC. Any suggestions would be most helpful. Here are my related openser.cfg settings: # LOCAL IP ADDRESS AND PORTS listen=65.185.232.62:5061 listen=65.185.232.62:5062 listen=65.185.232.62:5063 # IP ADDRESS AND PORTS OF THE BIG IP LOAD BALANCER alias=65.185.233.104:5061 alias=65.185.233.104:5062 alias=65.185.233.104:5063 thanks, Tim

3 3

[Serusers] group info
by KUMAR 23 Mar '07

23 Mar '07

Hi, I want to store the buddylist from the client application on the SER. How do i go about it. I guess, i can do it using the groups but cant' really find any resources to do the studying part. Can you please point me as to where i can find the information regarding this.?? thank you kumar

2 1

[Users] OpenSER sometimes forwarding 100 response
by Tim Madorma 23 Mar '07

23 Mar '07

Hi, I'm using version 1.2 of OpenSER. After proxying an INVITE from a UAC to a UAS, the UAS replies with a 100 Trying. *Sometimes* the OpenSER forwards this 100 Trying to the UAC. Any idea how to disable this? I think this can be done using onreply_route, but it seems that a received 100 should not be forwarded by default. thanks, Tim

2 1

[Users] reason of interlocking with Asterisk server
by 김형선 23 Mar '07

23 Mar '07

Hello. I think that some sites use OpenSER with Asterisk server. But Why? (for accounting? or supplementary service?) Aren't they same? In conceptionally, OpenSER is central VoIP server, and Asterisk is IP-PBX for apart site. right? What is the structure of service being applied, in generally? OpenSER, Asterisk,... What is the role of them? thank you. 새로운 기부문화의 씨앗, 해피빈

2 1

[Users] Openser + postgres not starting
by Knight Tiger 23 Mar '07

23 Mar '07

Hi I have installed Openser v 1.2 no tls. I have Postgres version 8.0.7. I tried openser start [root@openser /usr/local/lib]# openser start ERROR: bad config file (1 errors) [root@openser /usr/local/lib]# openser -c 0(29611) ERROR: load_module: could not open module </usr/local/lib/openser/modules/postgres.so>: /usr/local/lib/openser/modules/postgres.so: Undefined symbol "PQescapeStringConn" 0(29611) parse error (41,13-14): failed to load module ERROR: bad config file (1 errors) 0(29611) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy I then ran openserctl moni cycle #1, if constant make sure server lives My openser.cfg file # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ...... .... I checked for the PQescapeStringConn and found this www.mail-archive.com/devel@openser.org.msg05182.html but I could not make anything out of it I dont want authentication just yet. Can anyone tell me what is wrong ? Thanks in advance Caugar

3 3

[Users] an example of xcap_table entry?
by jh＠tutpro.com 22 Mar '07

22 Mar '07

is there somewhere an example off xcap_table entry that would show who is allowed to subscribe the presence of an AoR? -- juha

3 2

[Users] OpenSER performance tests
by Bogdan-Andrei Iancu 22 Mar '07

22 Mar '07

Hi everybody, based on the newly 1.2.0 OpenSER release, we ran some tests in order to give some realistic numbers about the performances of OpenSER. I think it is a very useful information for our community or other interested people. For more, please refer to the news bulletin: http://www.openser.org/mos/view/News/NewsItem/OpenSER-1.2.0---Perfomance-Te… or directly to the tests results web page: http://www.openser.org/docs/openser-performance-tests/ regards, bogdan

6 11

[Serusers] ACK for non-2XX not record-routed
by Emmanuel Hislen 22 Mar '07

22 Mar '07

Hi, I've been using SER (0.9.6) as a Proxy doing Loose Routing between 2 endpoints. If I place a call I can see how the ACK for the 2XX response to the INVITE is properly handled (Route header removed, Record-Route inserted). Now with the call still up, I do a re-INVITE which is rejected with a 415. The re-INVITE request sent by UAC contained a Route header, as it was an in-dialog request in a record-routed dialog. Now as per RFC3261 section 17.1.1.3: If the INVITE request whose response is being acknowledged had Route header fields, those header fields MUST appear in the ACK. This is to ensure that the ACK can be routed properly through any downstream stateless proxies. My UAC complied with the above, but SER seemed to choke on this: whereas the ACK for 2XX was successfully routed by SER in the original INVITE, the ACK for non-2XX (looking exactly the same: same Request URI, same IP destination, same Route header, ... different Via branch of course) is not handled properly: yes the ACK reaches UAS but it still has the Route header in it, and no Record-Route was added. And I'm guessing that if there had been another proxy in the Route Set on the way to the UAS it would have been bypassed. SER traces for ACK-2XX: > 0(13574) ===> Loose Routing logic... > 0(13574) parse_headers: flags=-1 > 0(13574) DEBUG: t_newtran: msg id=89 , global msg id=88 , T on entrance=0xffffffff > 0(13574) parse_headers: flags=-1 > 0(13574) parse_headers: flags=60 > 0(13574) t_lookup_request: start searching: hash=41794, isACK=1 > 0(13574) parse_headers: flags=28 > 0(13574) DEBUG: t_lookup_request: e2e proxy ACK found > 0(13574) parse_headers: flags=4 > 0(13574) DEBUG: totag for e2e ACK found: 0 > 0(13574) SER: forwarding ACK statelessly SER traces for ACK-non-2XX: > 0(13574) ===> Loose Routing logic... > 0(13574) parse_headers: flags=-1 > 0(13574) DEBUG: t_newtran: msg id=92 , global msg id=91 , T on entrance=0xffffffff > 0(13574) parse_headers: flags=-1 > 0(13574) parse_headers: flags=60 > 0(13574) t_lookup_request: start searching: hash=41807, isACK=1 > 0(13574) DEBUG: RFC3261 transaction matched, tid=007d7195499e33a1 > 0(13574) DEBUG: t_lookup_request: transaction found (T=0xb616ae48) > 0(13574) DEBUG: cleanup_uac_timers: RETR/FR timers reset > 0(13574) DEBUG: add_to_tail_of_timer[2]: 0xb616ae90 > 0(13574) DEBUG:destroy_avp_list: destroying list (nil) > 0(13574) receive_msg: cleaning up This doesn't seem right, am I missing something here? Thanks, Emmanuel.

3 5

Re: [Users] To: header in e.164 format - adding a plus
by Daryl Sanders 22 Mar '07

22 Mar '07

I'd love to use uac_replace_to(), but to my knowledge it does not exist. I read through the UAC module docs and there is no mention of it. Any other posibilities? If I were to use subst(), can it be applied it in such a way that it does not cause problems with the dialog? - Daryl On 3/21/07, Alan Crosswell <alan(a)columbia.edu> wrote: > Let me guess -- Level 3? They're the only ones that make that bogus > requirement AFAIK. And, the subst() is of course wrong, you need to do > uac_replace_to() or break the dialog otherwise.... > /a > > Daryl Sanders wrote: > > One of my providers requires the To: header to be in e.164 format with > > a "+" in front of it. Since there is no uac_replace_to() function yet, > > what are my options for adding this plus symbol? > > > > Before anyone suggests it... I cannot just use another provider. I > > must find a solution to add the plus symbol. > > > > Thanks, > > Daryl > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users >

3 3

[Users] SUBSCRIBE/NOTIFY problem
by Vadim Berezniker 22 Mar '07

22 Mar '07

The setup is: Polycom 601 => OpenSER => Asterisk The SUBSCRIBE from the phone reaches asterisk just fine and the OK is returned properly to the phone. The NOTIFY messages don't get to the phone, however. SER sends it to the requested URI (which is phone@ser), which ends up being relayed back to Asterisk.

1 1

[Users] parameter of route(n)
by jh＠tutpro.com 22 Mar '07

22 Mar '07

is it so that currently parameter n of call route(n) needs to be an integer constant, i.e., that use of a pseudo variable is not possible? -- juha

4 8

[Users] dispatcher, failure route and force_send_socket problem
by Klaus Darilion 22 Mar '07

22 Mar '07

Hi! Once again I have a problem with force_send_socket and branches in failure route. Scenario: normal route block --> call SIP client after timeout --> failure route in failure route send to gateway via dispatcher module The gateway accepts signaling only from a certain port, thus I have to use force_send_socket. The problem is, that if I call force_send_socket() in failure route, it sets the send_Socket for branches[0]. Thus if I call force_Send_socket before ds_select_domain it sets branches[0], but the sendsocket is not copied into the new branch when ds_select_domain internally calls append_branch. With version 1.1 I can use the ugly workaround to manually call append_branch too, as this copies another branch and also copies the sendsocket. Then I have 2 new branches where branches[1] gets dropped by teh firewall (wrong sendseocket) but branches[2] succeeds. But with version 1.2 the sendsocket is never copied from branches[0] and I can't set it manually for a certain branch. Are there any workarounds (for version 1.1. and 1.2) thanks klaus version 1.1: -- Klaus Darilion nic.at

1 1

[Users] [user] openser TLS and eyebeam
by Bodin Bruno 22 Mar '07

22 Mar '07

I made certificat, I add user folder to openser ser config (with lines it must) and I add certificat in Internet explorer for client. But eyebeam say certificat failed. Anyone have experience with eyebeam and tls ? thank.

2 1

[Users] OpenSer Asterisk
by Michael Chan 22 Mar '07

22 Mar '07

Hi, OpenSer Support, I read the OpenSer and Asterisk real time intergation and follow the config instruction but still no result that we can make PSTN call through Asterisk without using the name and PW of OpenSer on the UA Sip Client (X-lite). On the Asterisk side, we assigned a user name and PW to OpenSer, but not sure where we can create script on OpenSer side that allow all calls to Asterisk without using the shared secret name and PW of OpenSer. Please any help will be grateful. MC -------------------------------------------------------------------------------- I am using the free version of SPAMfighter for private users. It has removed 234 spam emails to date. Paying users do not have this message in their emails. Try SPAMfighter for free now!

2 1

RE: [Serusers] re: composing SIP messages
by Goh, Tee-Yong 22 Mar '07

22 Mar '07

Hi Greger, Thank you for sharing the information. May i know which part of the program in the SER can i find FIFO? Please advice. Thanks. Regards, Howard -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Greger V. Teigre Sent: Thursday, March 22, 2007 3:10 PM To: Jignesh Gandhi Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] re: composing SIP messages No, but there is a FIFO interface that allows inserting messages into SER. For anything but trivial cases, you will probably want to do it from SEMS. g-) Jignesh Gandhi wrote: Is there a way to compose SIP messages such as INVITE from scratch using SER ? I am new to SER... So any help would be appreciated thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

1 0

[Serusers] re: composing SIP messages
by Jignesh Gandhi 22 Mar '07

22 Mar '07

Is there a way to compose SIP messages such as INVITE from scratch using SER ? I am new to SER... So any help would be appreciated thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com

2 1

[Users] Compiling Openser for best performance on Solaris SPARC
by Sergio Gutierrez 21 Mar '07

21 Mar '07

Hi. Currently I am installing Openser 1.2.0 on a Sun Fire T2000 running Solaris 10. Although I used an optimized version of GCC for SPARC Systems, and used the recommended settings to generate accurate code for my Processor (UltraSPARC T1), the results of the performance tests are just 50% of the presented ones on the web page. My setup is the following: Proxy and UAS: Sun Fire T2000, 8 GB RAM, 1 UltraSPARC T1 1.2 GHZ 6 Core. UAC's: Two domain of a Sun Fire 12K, 4 GB RAM, 4 UlraSPARC III+ 900 Mhz Software versions listed as on the webpage. openser -V output: version: openser 1.2.0-notls (sparc/solaris) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT-NOSMP ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, select, /dev/poll. svnrevision: unknown @(#) $Id: main.c 1827 2007-03-12 15:22:53Z bogdan_iancu $ main.c compiled on 21:43:51 Mar 19 2007 with gcc 4.0.3 I would thank any idea about how could I improve the performnace of my setup. I think that because of the hardware specifications, the result should be at least equal than listed. Thanks in advance. Kind regards. Sergio Gutiérrez.

2 7

[Serusers] Acc not writing to MySQL
by POON Ben 21 Mar '07

21 Mar '07

Hi all, I am having a lot of trouble making the acc module work... I've googled and tried a lot of sample config but am still having no luck. Can someone please help me to get accounting on my server? Below is my ser.cfg file. It's basically the sample one with some editing. The auth stuff work really nicely :) Everything seems to load fine when I do a "ser -D -E" However, I noticed that nothing is ever entered into the acc table in the ser database. I am running ser 0.9.6 with mysql 5.0.37. Thank you so much! Ben # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) listen=1.1.1.1 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias="localhost" alias=1.1.1.1 /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://admin:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # Report missed calls setflag(2); append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # Set accounting flag setflag(1); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

1 0

[Users] To: header in e.164 format - adding a plus
by Daryl Sanders 21 Mar '07

21 Mar '07

One of my providers requires the To: header to be in e.164 format with a "+" in front of it. Since there is no uac_replace_to() function yet, what are my options for adding this plus symbol? Before anyone suggests it... I cannot just use another provider. I must find a solution to add the plus symbol. Thanks, Daryl

3 2

[Users] Fresh OpenSER 1.2.0 Openserctl can only add 1 user?!
by Geir O. Jensen 21 Mar '07

21 Mar '07

I have two server, one in production and one for testing. Trying to debug something else entirely I upgraded the production server from 1.1.0 to 1.2.0 yesterday. Everything seems to work fine on it. So I wiped my testbed and compiled OpenSER 1.2.0 on the testing system, installed the mysql db and started it up. Worked fine. I then began adding testaccounts on the testsystem, - first user ok... - second user, not so ok... # openserctl add test test gojensen(a)uninett.no MySql password for user 'xxx@localhost': ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 ERROR: introducing the new user 'test' to the database failed Of course, I couldn't add a third or fourth either, so I'm basically stuck with one user and the admin account. Any idea where to start debugging this?! (As a note, I added 3 test accounts on the production server with no problems... Their config files are basically identical - the testing server only adds the presence module...) // sip:gojensen@uninett.no | h323: 004710012 pstn: +47 73 55 79 23 | fax: +47 73 55 79 01

2 2

[Users] Wesip C2C
by Helmut Kuper 21 Mar '07

21 Mar '07

Hello, after upgrading to wesip 0.1.3 I'm not able anymore to start c2c application. in WeSIP's logfile I can find this exception: 2007-03-21 18:11:52 StandardManager[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-03-21 18:11:52 StandardManager[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-03-21 18:11:52 HTMLManager: ManagerServlet.install[/click2call-08_11_06] java.lang.NoClassDefFoundError: com/voztele/wesip/apps/click2call/C2CDefaultServlet at java.lang.ClassLoader.defineClass1(Native Method) at java.lang.ClassLoader.defineClass(Unknown Source) at java.security.SecureClassLoader.defineClass(Unknown Source) at java.net.URLClassLoader.defineClass(Unknown Source) at java.net.URLClassLoader.access$100(Unknown Source) at java.net.URLClassLoader$1.run(Unknown Source) at java.security.AccessController.doPrivileged(Native Method) at java.net.URLClassLoader.findClass(Unknown Source) at java.lang.ClassLoader.loadClass(Unknown Source) at sun.misc.Launcher$AppClassLoader.loadClass(Unknown Source) at java.lang.ClassLoader.loadClass(Unknown Source) at org.apache.catalina.loader.WebappClassLoader.loadClass(WebappClassLoader.java:1379) at org.apache.catalina.loader.WebappClassLoader.loadClass(WebappClassLoader.java:1307) at com.voztele.sipservlet.core.StandardSipWrapper.loadServlet(StandardSipWrapper.java:753) at com.voztele.sipservlet.core.StandardSipWrapper.load(StandardSipWrapper.java:701) at org.apache.catalina.core.StandardHttpContext.loadOnStartup(StandardHttpContext.java:3382) at org.apache.catalina.core.StandardHttpContext.start(StandardHttpContext.java:3605) at com.voztele.sipservlet.core.ConvergedContext.start(ConvergedContext.java:259) at org.apache.catalina.core.ContainerBase.addChildInternal(ContainerBase.java:773) at org.apache.catalina.core.ContainerBase.addChild(ContainerBase.java:759) at org.apache.catalina.core.StandardHttpHost.addChild(StandardHttpHost.java:552) at com.voztele.sipservlet.core.ConvergedHost.addChild(ConvergedHost.java:157) at com.voztele.sipservlet.core.ConvergedHostDeployer.install(ConvergedHostDeployer.java:264) at com.voztele.sipservlet.core.ConvergedHost.install(ConvergedHost.java:170) at org.apache.catalina.servlets.ManagerServlet.install(ManagerServlet.java:663) at org.apache.catalina.servlets.HTMLManagerServlet.install(HTMLManagerServlet.java:265) at org.apache.catalina.servlets.HTMLManagerServlet.doPost(HTMLManagerServlet.java:245) at javax.servlet.http.HttpServlet.service(HttpServlet.java:760) at javax.servlet.http.HttpServlet.service(HttpServlet.java:853) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:200) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:146) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:206) #################################################################################### When I cwitch back to wesip 0.1.2 C2C is loading and starting fine. I use click2call-08_11_06 When I search for the missed class C2CDefaultServlet I get this: hk@vm-en-test0:~/wesip/wesip$ find . -name "C2CDefaultServlet*" ./wesipapps/click2call-08_11_06/WEB-INF/classes/com/voztele/wesip/apps/click2call/C2CDefaultServlet.class So it#s there, but wesip doesn#t find it. Wher do I have to tell weSIP the location of that class ? regards helmut

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Re: [Users] NAT Problem with OpenSER behind load balancer
by Daniel-Constantin Mierla 21 Mar '07

21 Mar '07

You should keep cc-ing mailing lists, so people coming after a while, with same issue, will be reading mail thread and see that the given solution was right. Thanks, Daniel On 03/20/07 21:42, Tim Madorma wrote: > Got it. thanks so much! > > Tim > > On 3/20/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> On 03/20/07 17:39, Tim Madorma wrote: >> > Hi, >> > >> > I have put the OpenSER version 1.2 behind a BIG IP load balancer which >> > provides a static private IP address (10.1.1.20) via NAT to the >> > OpenSER. In our application, the clients will send to one of the three >> > public addresses that are defined as aliases: >> > >> > alias="65.185.232.62:5061" >> > alias="65.185.232.62:5062" >> > alias="65.185.232.62:5063" >> you have to remove quotes around the value of alias parameters. As it is >> now, the value between quotes is considered to be hostname. >> alias=65.185.232.62:5061 >> alias=65.185.232.62:5062 >> alias=65.185.232.62:5063 >> >> Same case as it was reported by Juha, today with loose routing ... >> >> Cheers, >> Daniel >> >> > >> > An incoming OPTIONS message is received with the host in the REQ URI >> > set to 65.185.232.62:5063 is received and when it hits this section of >> > openser.cfg: >> > >> > if (!uri==myself) { >> > # mark routing logic in request >> > append_hf("P-hint: outbound\r\n"); >> > route(1); >> > }; >> > >> > it executes within the if conditional and tries to route the OPTIONS >> > message. I am wondering why uri does not match myself. The debug >> > output from the log is as follows (it seems that the alias assignments >> > are not working): >> > >> > >> > Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] >> > grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 >> > 0.1.1.20] >> > Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] >> > grep_sock_info - checking if port 5061 matches port 5063 >> > Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] >> > grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 >> > 0.1.1.20] >> > Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] >> > grep_sock_info - checking if port 5062 matches port 5063 >> > Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] >> > grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 >> > 0.1.1.20] >> > Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] >> > grep_sock_info - checking if port 5063 matches port 5063 >> > Mar 19 16:41:11 homer openser[2267]: [ID 140248 local0.debug] >> > DEBUG:check_self: host != me >> > >> > >> > Any suggestions? I have not tried this in any other version. Is this a >> > possible bug in 1.2? >> > >> > thanks, >> > Tim >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >

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[Users] ENC: Presence Module - Eyebeam 1.5
by Hildebrando Trannin 21 Mar '07

21 Mar '07

Hi all, I configured the presence module in OpenSER v1.2 and it looks like ok. But when tested with EyeBeam 1.5 (buil 30037) it doesn't work. I used the WireShark to see the SIP messages and observed that in NOTIFY messages there aren't PIDF document, neither message body. Furthermore, OpenSER didn't parse xml BODY from PRESENTITY table when try to build notify. The PIDF document that EyeBeam creates and the error that OpenSER gave me when try to parser this document are below. Can someone help me with this problem? Is it a bug from OpenSER? Regards, Hildebrando PIDF Document <?xml version="1.0" encoding="UTF-8" standalone="no" ?> <pr:presence xmlns:pr="urn:ietf:params:xml:ns:pidf" entity="sip:hildebrando@voip.maritaca" xmlns:caps="urn:ietf:params:xml:ns:pidf:caps" xmlns:cipid="urn:ietf:params:xml:ns:pidf:cipid" xmlns:counterpath="www.counterpath.com/presence/ext" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"> <pr:tuple id="se43ac217"> <pr:status> <pr:basic>open</pr:basic> </pr:status> <pr:note>Busy</pr:note> <rpid:user-input last-input="2007-03-19T19:21:49Z">active</rpid:user-input> <pr:timestamp>2007-03-19T19:21:49Z</pr:timestamp> </pr:tuple> <dm:person id="p0d4e4429"> <rpid:activities> <rpid:busy/> </rpid:activities> <dm:note>Busy</dm:note> </dm:person> </pr:presence> OpenSER Error 2(6498) Chegou um PUBLISH de (hildebrando) para (sip:hildebrando@voip.maritaca) 2(6498) PRESENCE:update_presentity: querying presentity 2(6498) PRESENCE:get_subs_dialog:querying database table = active_watchers 2(6498) PRESENCE:get_p_notify_body: querying presentity 2(6498) PRESENCE:agregate_xmls: ERROR while parsing xml body message 2(6498) PRESENCE:notify:dialog informations: 2(6498) [p_user]= hildebrando [p_domain]= voip.maritaca [w_user]= gustavo [w_domain]= voip.maritaca 2(6498) [event]= presence [staus]= active [expires]= 3340 2(6498) [to_tag]= 10.6503.1174328485.15 [from_tag]= 80746145

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[Users] xmpp module
by Sebastian Murray-Roberts 21 Mar '07

21 Mar '07

Hi, I just have a question about the new xmpp module. I have set up an OpenSER box as a presence server using the presence module, and everything is working fine, messaging and presence. Now I would like to allow those users to communicate with users logged into a jabber server. My first question is can presence information be exchanged between OpenSER and the jabber server? If so, how is this achieved since the xmpp module only has an xmpp_send_message() method. Also, just to make sure I understand how the module works, would this be a correct summary of the module's functionality for a SIP to Jabber message with a response: User A registers with OpenSER. User B registers with jabber server (OpenFire) User A sends a message to sip:User B*<jabber_domain>@<gateway_domain> with from details : sip:User A*<openser_domain>@<gateway_domain> OpenSER forwards the request to the gateway Gateway translates the message and sends it to the jabber server Jabber server delivers message to User B Jabber server sends response to gateway with to details: User A*<openser_domain>@<gateway_domain> Gateway forwards response to sip:User A@<openser_domain> Is that more or less correct, or am I off the mark? Finally, if that is the case, would User A need to be registered on the jabber server already? I've never worked with presence before, so I'm still trying to get to grips with it. Any comments/insights are much appreciated. Many thanks, Sebastian -- Sebastian Murray-Roberts (B.Sc Hons) Server Developer Yeigo Communications http://www.yeigo.com Tel: +27 21 409 7807 Fax: +27 21 409 7050 Mobile: +27 72 183 5998 CONFIDENTIALITY NOTICE: E-MAIL This e-mail is confidential. If you receive it by mistake, please delete it and notify the sender immediately. If you are not the intended recipient, you may not use, disclose, distribute, copy or publish any information contained herein.

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[Users] Commercial SIP Server
by Michel Bensoussan 21 Mar '07

21 Mar '07

Hello Is there a way to know which commercial SIP servers are based on openser or ser? Or if a specific SIP server is based on openser or ser? Thanks. Regards, Michel.

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答复: [Users] ERROR 1062 (23000): Duplicate entry
by 郭军 21 Mar '07

21 Mar '07

Why don't you use openserctl add ....to add new user? -----邮件原件----- 发件人: alper ozbilen [mailto:alperbilen@hotmail.com] 发送时间: 2007年3月21日 16:45 收件人: users(a)openser.org 主题: [Users] ERROR 1062 (23000): Duplicate entry Dear All, I can not insert a new subscriber from mysql console. When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks null. So, i can only insert one new subscriber. When i attempt to insert second one, i get error written below: ERROR 1062 (23000): Duplicate entry '' for key 3 Note : My insert sentence like that: insert into subscriber (username, domain, password, email_address) values ("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); I am confused about what i should to do to fix null statement of ha1 & ha2 & and phplib_id fields. on the other side, there is no probem to add new user through openserctl What was the problem? I will wait your kind suggestion. Regards, Alper Ozbilen _________________________________________________________________ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema… _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] RHEL/Fedora equivalents of deb packages in RPM SPEC
by Alan Crosswell 21 Mar '07

21 Mar '07

The BuildPrereq lines in the RPM SPEC file for Fedora Core (openser.spec.fedora) supplied with 1.2.0 references debian names for the package (e.g. libxml2-dev -> libxml2-devel). Before I waste too much time finding the equivalent RHEL and/or Fedora Core packages that do the job, I thought I would ask her if anyone else has done so. If not, I'll make sure to submit patches for the SPECs once I figure them out. I am really looking to build on RHEL4 but clearly will need to grab some RPMs from Fedora Core.... Please no religious rants about Debian vs. RHEL vs. Fedora vs. SuSE and so on:-) /a

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[Users] info-color of openserctl
by Andreas Granig 21 Mar '07

21 Mar '07

Hi, Could we please change the color of minfo from yellow to, say, orange in openserctl? It's impossible to read on a console with white background. Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

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[Users] domain module in 1.2
by Andreas Granig 21 Mar '07

21 Mar '07

Hi, In openser 1.1, when I use db_mode=0 (no caching) for domain module, I get this error: /usr/local/openser-1.1# ./sbin/openserctl domain show 500 command 'domain_dump' not available In openser 1.2, the first time I execute the command, I get nothing, the second time openserctl just hangs until I terminate it with ctrl-c: /usr/local/openser-1.2# ./sbin/openserctl domain show /usr/local/openser-1.2# ./sbin/openserctl domain show ./sbin/openserctl: line 83: /tmp/openser_fifo: Interrupted system call Terminated Same thing for "domain reload" btw. Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

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[Users] Only db_mode=0 works with dbtext loaction table in 1.2
by Daryl Sanders 21 Mar '07

21 Mar '07

When using db_mode=1,2 or 3 I get the following error when starting OpenSER 1.2 using dbtext as the usrloc DB. 1(23052) preload_udomain(): Error while doing db_query 1(23052) ERROR:ul:child_init(1): Error while preloading domain 'location' 1(23052) ERROR:init_mod_child: Error while initializing module usrloc, rank 1 I am wondering if their may be a problem with the columns in the location table file. I tried two different methods of creating the initial table files... 1.) sc.dbtext dbtext /usr/local/etc/openser/dbtext/ 2.) textdb.sh create /usr/local/etc/openser/dbtext/ The first method has problems with the version file that I tried to repair by trying to mimic the mysql table structure. The second method seems to work ok with the exception of the error I mentioned above when starting OpenSER. Any help would be greatly appreciated! Thanks, Daryl

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RE: [Users] DB migration
by Leonid Fainshtein 21 Mar '07

21 Mar '07

Thank you, Brad. I missed MySQL RENAME DATABASE command. It is because the option appears in MySQL starting from version 5.1 only and I have version 5.0. It seems that for older MySQL the following sequence will work: openser_mysqldb.sh migrate openser openser12 mysqldump -u root -pxxxxx -c -t --ignore-table=openser12.version openser12 > /tmp/openser12.bak openser_mysqldb.sh reinit echo "delete from subscriber;" | mysql -u openser -pxxxxx openser echo "delete from admin_priveleges;" | mysql -u openser -pxxxxx openser mysql -u openser -pxxxxx openser < /tmp/openser12.bak Leonid -----Original Message----- From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] Sent: Tuesday, March 20, 2007 7:23 PM To: Leonid Fainshtein; users(a)openser.org Subject: RE: [Users] DB migration In MySQL at least: rename database <old name> <new name> Once you've confirmed that the migration was successful, you can run a backup of the old datbase, drop it, and then rename the new one. - Brad > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Leonid Fainshtein > Sent: Tuesday, March 20, 2007 12:43 PM > To: users(a)openser.org > Subject: [Users] DB migration > > Hi, > I didn't understand the proposed process of MySQL DB migration from > ver.1.1 to 1.2 > For example, the result of "openser_mysqldb.sh migrate > openser openser12" will be a new database openser12. > But how can I copy/rename openser12 to openser? Does it mean > that from now, all of my auxiliary software must work with openser12? > > Thank you, > Leonid Fainshtein > > > ******************************************** > This outgoing mail message was scanned by McAfee GroupShield Engine > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

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[Users] Problem with dialog module
by Leonid Fainshtein 21 Mar '07

21 Mar '07

I just to tested the issue discussed in http://openser.org/pipermail/users/2007-March/009777.html (OpenSer removes the did parameter from re-INVITE) with version 1.2.0. Unfortunatelly the problem still exists... Leonid ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

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[Users] Presence and Postgresql
by H Quintana 21 Mar '07

21 Mar '07

Hi everybody, I'm using Xlite 3.0 and postgresql 8.2.3, the information about the presentity is properly stored in the table. The query brings the value for the field "body" but after the "conversion" to string the value is empty. Is Someone running the presence module with postgres? The log is at: http://cut.and.paste.org/index.php?id=952 I added this to notify.c in the line 514: DBG("PRESENCE:NOTIFY:Body array n=%d, content:<%.*s>\n", i, body_array[i]->len, body_array[i]->s); and in the log you will find: 0(14253) PRESENCE:NOTIFY:Body array n=0, content:<> before the line: 0(14253) PRESENCE:agregate_xmls: ERROR while parsing xml body message Thanks for your help, Humberto Quintana ____________________________________________________________________________________ TV dinner still cooling? Check out "Tonight's Picks" on Yahoo! TV. http://tv.yahoo.com/

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[Users] lcr, branch route and setdsturi
by Andreas Granig 21 Mar '07

21 Mar '07

Hi, Is it possible to set the dst-uri in a branch-route (in openser 1.1)? It would be useful if you could for example do least-cost-routing with lcr, where some of the gateways need an extra hop via an outbound-proxy, which could be marked with a prefix in the gw-table, like: route[1] { # let's assume uri = 'sip:12345@something' load_gws(); next_gw(); # lcr module selects a gw according to lcr patterns, and some of # them needs an outbound-proxy, which are marked by prefix 'out' t_on_failure("1"); t_on_branch("1"); # do something else and t_relay() } failure_route[1] { if( /* error code which triggers failover */ ) { if(next_gw()) { t_on_branch("1"); t_on_failure("1"); t_relay(); } else { /* error */ } } } branch_route[1] { # check if it's marked for using an outbound proxy: if(uri =~ "^sip:out[0-9]+@") setdsturi("sip:my.outbound.ip:5060"); } Would this work? Comments? Regards, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

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[Serusers] NAT Handling - SER behind a router
by TZieleniewski 21 Mar '07

21 Mar '07

Hi!! Right now I have SER based voip system but inside the local network. I would like to enable the external internet connection. Is it possible to realize the NAT handling but with SER-RTPProxy or SER-MediaProxy not with the public IP so still in the local net but with the strict forwarding rules on the router. where router ofcaurse has the public IP So that for instance having SER and media proxy inside local network but with whole SIP traffic routed to SER and RTP to media proxy?? Is one of the above methods more recommened at the moment with ser2?? Thank you in advance Best Tomasz

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[Serusers] Call forwarding
by Jai Rangi 21 Mar '07

21 Mar '07

Hello, I have a ser and asterisk setup. Users register on ser and all the national and international calls go through asterisk. Asterisk handles the billing part for long distance calls. Now I got an issue with call forwarding. Here is the situation. Useragent A calls Usergent B. User B has setup the call the call forwarding to his cell phone. So when user A calls user B, ser check that user B has set the forwarding and send the call to Asterisk box. Asterisk box sees user A in the callerID and bills user A for that call. We want user B to be billed for that call, cause he is forwarding the calls to his cell phone (long distance number). How can I achieve this. Thank you, -Jai

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Re: 答复: [Users] ERROR 1062 (23000): Duplicateentry
by sukerry 21 Mar '07

21 Mar '07

郭军,您好！ please modify openserctrlrc, set ?WEB?(some what) to be yes ======= 2007-03-21 16:49:35 您在来信中写道：======= >Why don't you use openserctl add ....to add new user? > >-----邮件原件----- >发件人: alper ozbilen [mailto:alperbilen@hotmail.com] >发送时间: 2007年3月21日 16:45 >收件人: users(a)openser.org >主题: [Users] ERROR 1062 (23000): Duplicate entry > >Dear All, > >I can not insert a new subscriber from mysql console. >When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks >null. So, i can only insert one new subscriber. When i attempt to insert >second one, i get error written below: > >ERROR 1062 (23000): Duplicate entry '' for key 3 > >Note : My insert sentence like that: >insert into subscriber (username, domain, password, email_address) values >("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); > >I am confused about what i should to do to fix null statement of ha1 & ha2 & >and phplib_id fields. > >on the other side, there is no probem to add new user through openserctl > >What was the problem? I will wait your kind suggestion. > >Regards, >Alper Ozbilen > >_________________________________________________________________ >Exercise your brain! Try Flexicon. >http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema… > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　sukerry 　　　　　　　　sukerry(a)126.com 　　　　　　　　　　2007-03-21

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[Users] ERROR 1062 (23000): Duplicate entry
by alper ozbilen 21 Mar '07

21 Mar '07

Dear All, I can not insert a new subscriber from mysql console. When I look at subscireber table, ha1 & ha1b & and phplib_id fields looks null. So, i can only insert one new subscriber. When i attempt to insert second one, i get error written below: ERROR 1062 (23000): Duplicate entry '' for key 3 Note : My insert sentence like that: insert into subscriber (username, domain, password, email_address) values ("testuser","myfon.com","testpasswd","emailoftestuser(a)myfon.com"); I am confused about what i should to do to fix null statement of ha1 & ha2 & and phplib_id fields. on the other side, there is no probem to add new user through openserctl What was the problem? I will wait your kind suggestion. Regards, Alper Ozbilen _________________________________________________________________ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglinema…

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[Users] OpenSER and FWD
by Michel Bensoussan 21 Mar '07

21 Mar '07

Hello. How can I configure OpenSER to outgoing all call to FWD and more important to ensure that all INCOMING call will go through OpenSER? Thanks. Regards, Michel.

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[Users] RPID single quotes problem
by Daryl Sanders 20 Mar '07

20 Mar '07

Is there an easy way to strip the single quotes from RPID in DBTEXT? When I use sc.dbtext it automatically adds single quotes to the entry like so... '17145551212' As a result when I add RPID to my header with the folowing line, I get the single quotes in the header. append_hf("P-Asserted-Identity: \"User\" <sip:+$avp(s:rpid)@x.x.x.x>\r\n"); Regards, Daryl

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Re: [Users] loose_route problem with 1.2
by jh＠tutpro.com 20 Mar '07

20 Mar '07

it turned out that if i have in openser.cfg listen=x.x.x.x alias="sip.foo.bar:5060" port=5060 auto_aliases=no then openser advertises at startup: Aliases: *: sip.foo.bar:5060:* and loose_route DOES NOT recognize that Route header <sip:sip.foo.bar;lr;transport=TCP> is local. on the other hand, if i comment line #auto_aliases=no openser advertises at startup: Aliases: tcp: sip.foo.bar:5060 udp: sip.foo.bar:5060 *: sip.foo.bar:5060:* and it DOES recognize that above Route header is local. if i uncomment auto_aliases=no and change alias="sip.foo.bar:5060" to alias="udp:sip.foo.bar:5060" alias="tcp:sip.foo.bar:5060" then loose_route DOES NOT recognize that the route header is local. my question is, how should i write the alias statements so that the route header is recognized local even when i have auto_aliases=no? -- juha

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Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Kostas Marneris 20 Mar '07

20 Mar '07

Hello, What I understand (from Frame12) Session Initiation Protocol Request-Line: BYE sip:000003@1.255.ua_priv__IP SIP/2.0 is that your GW does not comply with the fixed SIP UA Contact address Contact: <sip:000003@213.156.ua_pub_IP:1176> which SER seems to correctly fix it (from Frame6). GW must send the BYE request back to the fixed addr : 213.156.ua_pub_IP:1176. Try a different PSTN-GW or try to isolate the RTP-Proxy solution. I am really confused of your ser.cfg. You use force_rtp_proxy("l") on Loose Route Section / INVITE block, but you don't use force_rtp_proxy() on INVITE handler (Route[3]) and you don't also use rtp_proxy on BYE block at Loose Route Section. I tried this senario without Mediaproxy or RTPproxy with success. regards, Kostas --- K.Marneris(a)otenet.gr ----- Original Message ----- From: "Fabio Macchi" <f.macchi(a)keeptelecom.com> To: <greger(a)teigre.com> Cc: "'Kostas Marneris'" <K.Marneris(a)otenet.gr>; <serusers(a)lists.iptel.org> Sent: 20 March 2007 19:56 Subject: R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hi Greger, I attached an ethereal SIP call trace of a test call ( summary and detailed, I simple maskerade final ip numbers ): below only the INVITE relayed from proxy to gateway: Session Initiation Protocol Request-Line: INVITE sip:9999001234@194.244.gatewayIP:5060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:194.244.Proxy__IP;ftag=12e1e2e19d527792;lr=on> Via: SIP/2.0/UDP 194.244.Proxy__IP;branch=z9hG4bKb24f.5133a2c4.0 Transport: UDP Sent-by Address: 194.244.Proxy__IP Branch: z9hG4bKb24f.5133a2c4.0 Via: SIP/2.0/UDP 1.255.ua_priv__IP;rport=1176;received=213.156.ua_pub_IP;branch=z9hG4bK35ca7a df63e3094f Transport: UDP Sent-by Address: 1.255.ua_priv__IP RPort: 1176 Received: 213.156.ua_pub_IP Branch: z9hG4bK35ca7adf63e3094f From: "000003" <sip:000003@194.244.Proxy__IP>;tag=12e1e2e19d527792 SIP Display info: "000003" SIP from address: sip:000003@194.244.Proxy__IP SIP tag: 12e1e2e19d527792 To: <sip:9999001234@194.244.Proxy__IP> SIP to address: sip:9999001234@194.244.Proxy__IP Contact: <sip:000003@213.156.ua_pub_IP:1176> Contact Binding: <sip:000003@213.156.ua_pub_IP:1176> URI: <sip:000003@213.156.ua_pub_IP:1176> SIP contact address: sip:000003@213.156.ua_pub_IP:1176 Supported: replaces Call-ID: 953e8996cfcc4ccc(a)1.255.ua_priv__IP CSeq: 60577 INVITE Sequence Number: 60577 Method: INVITE User-Agent: Grandstream HT386 1.0.3.64 FXS0 Max-Forwards: 16 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 327 Message body As you can see, contact informations are correctly fixed with the pubblic UA address, but when the callee hungs up, the BYE is relayed to the private IP address: am I missing something ? In this invite I see UA private address only in VIA: does BYE look to this parameter ? Later caller hangs up too, and the OK is relayed to the correct IP/port. Any help would be high appreciate, thanks in advance. Fabio _____ Da: Greger V. Teigre [mailto:greger@teigre.com] Inviato: martedμ 20 marzo 2007 9.59 A: Fabio Macchi Cc: 'Kostas Marneris'; serusers(a)lists.iptel.org Oggetto: Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Normally this happens because you haven't fixed the Contact if the original INVITE or OK. g-) Fabio Macchi wrote: First, thanks for answer. I've tryed your trik and in effect this solve the problem of the '200 ok' forwarded to the UA, but my problem still remain alive: when BYE is sent from Gateway, it reaches correctly SER, but it still forward it to the private UA address. I was wondering about the nat_uac_test in this case, as the source of the BYE message is the gateway ( not natted ) and not the UA. Have any idea about this ? Fabio -----Messaggio originale----- Da: Kostas Marneris [mailto:K.Marneris@otenet.gr] Inviato: giovedμ 15 marzo 2007 20.39 A: Fabio Macchi Cc: serusers(a)lists.iptel.org Oggetto: Re: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hello, I was working on about the same problem today either with 'Mediaproxy solution' or with 'SER's Nathelper only solution' . The NAT issue is a nightmare, not because of SER but because of different implementations on NAT boxes. Actually my problem was : if the NATed UA send a BYE to SER, SER forward it to PSTN-GW, then the '200 Ok' Response from PSTN-GW is forwarded by SER to UA to the wrong port (Contact or Via header port). I used the following block on Loose Route section, (because BYE is loose_routed if you use Record-Route), and it seems to work. # --------------------------------------- # Loose Route Section # --------------------------------------- if (loose_route()) { # mark routing logic in request if (method == "BYE") { if (nat_uac_test("22")) { xlog("L_NOTICE", "*** LR -> NATed BYE - Use force_rport()"); force_rport(); }; }; route(1); break; }; I faced up your second problem too. The solution was to move the NAT handling block before proxy_authorize block. I think that the different behaviour does not come with the 'standard RFC1918 addresses', but with the different NAT type. I realize that the provisional mesgs '100 Trying' and '407 Proxy Authentication Required' are relayed back to the real IP addr of NATed UA (this is correct), but to the WRONG port (that of Contact/Via header and not the signalling received port). It seems that these mesgs use the IP address part of 'Received' field of Location DB but not the port. It happens to work if NAT box use the SAME port (eg. 5060) on NAT translation (10.10.10.1:5060 --> Real_IP:5060) (eg. with a SAGEM1500 Router) But it does not work if NAT box doesn't use the same port (10.10.10.1:5060 --> Real_IP:38181) I think that this has to be verified by SER developers or SER experts. Kostas --- K.Marneris(a)otenet.gr ----- Original Message ----- From: "Fabio Macchi" <mailto:f.macchi@keeptelecom.com> <f.macchi(a)keeptelecom.com> To: <mailto:serusers@lists.iptel.org> <serusers(a)lists.iptel.org> Sent: 15 March 2007 19:34 Subject: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hi all, I'm running the following schema: UA ( possibly natted ) -> SER -> PSTN Gateway I have a problem with UA belonging to a particular network with private address not RFC1918 compliant ( class 1.x.x.x ), SER and PSTN Gateway have pubblic address. The problem is that, after a succesfull call, if the PSTN gateway send a BYE to SER, then SER forward BYE to the private address of UA instead of pubblic one. I don't understand which is the section that handle BYE messages and how can I solve this problem: anyone may help ? Second, another question: with this particular network I had problem with INVITE too, because SER was sending "proxy authorization request" to the wrong TCP port. To solve this, I've moved the nat handling ( with force_rport ) before the proxy_authorize block and it's working, but why this is not necessary on standard RFC1918 compliant natted address ? Many thanks for any explanation Fabio _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------------------------------------------------------------------- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.15/728 - Release Date: 20/03/2007 08:07

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Re: R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Greger V. Teigre 20 Mar '07

20 Mar '07

Thanks for the detailed trace, I wouldn't have spent on this without it ;-) Its' the ACK for the OK that is not done fix_nated_contact for. In fact, checking the getting started rtpproxy file, it seems that this error is also found there. All messages coming from behind NAT should have the contact fixed. Strictly speaking, RFC3261 states that the dialog state should be created from the information found in the INVITE, not the ACK on an OK. Your BYE generating entity does this error. I have seen this problem before, but has not really thought about it from the getting started config file perspective. It is thus safer to change Contact on ALL messages behind NAT, regardless of they are loose routed or not. g-) Fabio Macchi wrote: > > Hi Greger, > > > > I attached an ethereal SIP call trace of a test call ( summary and > detailed, I simple maskerade final ip numbers ): below only the INVITE > relayed from proxy to gateway: > > > > Session Initiation Protocol > > Request-Line: INVITE sip:9999001234@194.244.gatewayIP:5060 SIP/2.0 > > Method: INVITE > > [Resent Packet: False] > > Message Header > > Record-Route: <sip:194.244.Proxy__IP;ftag=12e1e2e19d527792;lr=on> > > Via: SIP/2.0/UDP 194.244.Proxy__IP;branch=z9hG4bKb24f.5133a2c4.0 > > Transport: UDP > > Sent-by Address: 194.244.Proxy__IP > > Branch: z9hG4bKb24f.5133a2c4.0 > > Via: SIP/2.0/UDP > 1.255.ua_priv__IP;rport=1176;received=213.156.ua_pub_IP;branch=z9hG4bK35ca7adf63e3094f > > Transport: UDP > > Sent-by Address: 1.255.ua_priv__IP > > RPort: 1176 > > Received: 213.156.ua_pub_IP > > Branch: z9hG4bK35ca7adf63e3094f > > From: "000003" <sip:000003@194.244.Proxy__IP>;tag=12e1e2e19d527792 > > SIP Display info: "000003" > > SIP from address: sip:000003@194.244.Proxy__IP > > SIP tag: 12e1e2e19d527792 > > To: <sip:9999001234@194.244.Proxy__IP> > > SIP to address: sip:9999001234@194.244.Proxy__IP > > Contact: <sip:000003@213.156.ua_pub_IP:1176> > > Contact Binding: <sip:000003@213.156.ua_pub_IP:1176> > > URI: <sip:000003@213.156.ua_pub_IP:1176> > > SIP contact address: sip:000003@213.156.ua_pub_IP:1176 > > Supported: replaces > > Call-ID: 953e8996cfcc4ccc(a)1.255.ua_priv__IP > > CSeq: 60577 INVITE > > Sequence Number: 60577 > > Method: INVITE > > User-Agent: Grandstream HT386 1.0.3.64 FXS0 > > Max-Forwards: 16 > > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE > > Content-Type: application/sdp > > Content-Length: 327 > > Message body > > > > As you can see, contact informations are correctly fixed with the > pubblic UA address, but when the callee hungs up, the BYE is relayed > to the private IP address: am I missing something ? > > In this invite I see UA private address only in VIA: does BYE look to > this parameter ? > > Later caller hangs up too, and the OK is relayed to the correct IP/port. > > > > Any help would be high appreciate, thanks in advance. > > > > Fabio > > > >

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[Users] NAT Problem with OpenSER behind load balancer
by Tim Madorma 20 Mar '07

20 Mar '07

Hi, I have put the OpenSER version 1.2 behind a BIG IP load balancer which provides a static private IP address (10.1.1.20) via NAT to the OpenSER. In our application, the clients will send to one of the three public addresses that are defined as aliases: alias="65.185.232.62:5061" alias="65.185.232.62:5062" alias="65.185.232.62:5063" An incoming OPTIONS message is received with the host in the REQ URI set to 65.185.232.62:5063 is received and when it hits this section of openser.cfg: if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; it executes within the if conditional and tries to route the OPTIONS message. I am wondering why uri does not match myself. The debug output from the log is as follows (it seems that the alias assignments are not working): Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5061 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5062 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5063 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 140248 local0.debug] DEBUG:check_self: host != me Any suggestions? I have not tried this in any other version. Is this a possible bug in 1.2? thanks, Tim

3 2

RV: RV: [Users] WeSIP session question
by Antonio Abajo 20 Mar '07

20 Mar '07

Buenas tardes Tele In the capture, I see the following call flow. GW WESIP MS ------ INV(S1)-----> -------INV (S2) ------> <--------183 ----------- ------- 183 -----------> <----------ACK --------- <---------503 ---------- ... 14:07:57 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Request Received <<<<<<<<< INVITE sip:390104491079@82.215.163.67 SIP/2.0 Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=6a4fa6b1 To: <sip:390104491079@82.215.163.67> Call-ID: 1783604896-30787@SVIGateway CSeq: 1 INVITE Contact: <sip:82.215.163.5:5060> Content-Type: application/sdp Content-Length: 213 14:07:58 19Mar2007 DEBUG SipRequest [SipProcessor[4]]- >>>>>>>>> Sending Request >>>>>>>>> INVITE sip:199@82.215.133.50 SIP/2.0 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67> CSeq: 1 INVITE Content-Type: application/sdp Call-ID: 11111783604896-30787@SVIGateway Contact: <sip:82.215.163.67:5060;transport=udp> Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 Content-Length: 21314:07:58 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Response Received <<<<<<<<< SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 Date: Mon, 25 Mar 2002 04:04:50 GMT Call-ID: 11111783604896-30787@SVIGateway Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,UPDATE,REGISTER,SUBSCRIBE,NOTIFY,PRACK,RE FER Allow-Events: telephone-event Contact: <sip:199@82.215.133.50:5060> Content-Type: application/sdp Content-Disposition: oSystemsSIP-GW-UserAg Content-Length: 235 14:07:58 19Mar2007 DEBUG SipResponse [SipProcessor[4]]- >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=6a4fa6b1 To: <sip:390104491079@82.215.163.67> CSeq: 1 INVITE Call-ID: 11111783604896-30787@SVIGateway Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 9756 1908 IN IP4 82.215.133.50 s=SIP Call c=IN IP4 82.215.133.50 t=0 0 m=audio 19450 RTP/AVP 0 99 c=IN IP4 82.215.133.50 a=rtpmap:0 PCMU/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[3]]- <<<<<<<<< Request Received <<<<<<<<< ACK sip:199@82.215.133.50 SIP/2.0 Via: SIP/2.0/UDP 82.215.163.67;branch=z9hG4bK07de.99a696f6.0 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A Call-ID: 11111783604896-30787@SVIGateway To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 CSeq: 1 ACK User-Agent: OpenSer (1.2.0-pre6-notls (i386/linux)) Content-Length: 0 14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Response Received <<<<<<<<< SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 Date: Mon, 25 Mar 2002 04:04:50 GMT Call-ID: 11111783604896-30787@SVIGateway Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=41 Content-Length: 0 The signalling is not correct but, I have looked your code and believe that it is right, revise it and you check if you have the last version of WeSIP and you return to try, because the last release commited in HYPERLINK "http://www.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1"http://ww w.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1 was fixed a bug with the ApplicationSessions. In my test application I reply your behaviour(with 5xx, with AppSession attributes) and all have working perfectly, I dont understand nothing :( Sorry Good Luck. Antonio -----Mensaje original----- De: tele [mailto:tele@plexia.com] Enviado el: martes, 20 de marzo de 2007 16:46 Para: users(a)openser.org CC: antonio.abajo(a)voztele.com Asunto: Re: RV: [Users] WeSIP session question Hola Antonio, Your example works great and the code is more clear and correct than mine :) i was not clear, my problem was only with the 503 Service Unavailable response, that means a 5xx Server Failure, is it possibile that when the container or openser with seas recive a 5xx response is not able to do for example a creation of a new invite to UA3 recover from the same session?. Or probably i'm occur in a 473 response for an incorrect use of SipApplicationSession. anyway with a 480 Temporarily Unavailable response from the UA2 i solve my problem. thank you very much :tele On Mon, 2007-03-19 at 19:27 +0100, Antonio Abajo wrote: > Hola tele ;) > > I have tested an example similar to yours, and it works ok. It's de > following: > > UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2 > | > [3] > ---OK----_> UA3 > > [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA > behaviour. In do INVITE. I have implemented the following: > > protected void doInvite(SipServletRequest invite){ > SipServletRequest otherInvite = sf.createRequest(invite, false); > SipURI sipUri = sf.createSipURI("UA2","proxy"); > otherInvite.setRequestURI(sipUri); > otherInvite.getSession().setAttribute("REQUEST", otherInvite); > > otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession()); > > invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession()); > invite.getSession().setAttribute("REQUEST", invite); > otherInvite.send(); > } > > [2] UA2 declines the call and a 486 response is received by WeSIP and > process it in doErrorResponse: > > protected void doErrorResponse(SipServletResponse errorResponse) { > switch(errorResponse.getStatus()){ > case 486: > SipServletRequest request = (SipServletRequest) > errorResponse.getSession().getAttribute("REQUEST"); > SipSession otherSession = (SipSession) > errorResponse.getSession().getAttribute("PEER_SESSION"); > SipURI sipUri = sf.createSipURI("UA3", "proxy"); > request.setRequestURI(sipUri); > SipServletRequest newRequest = > sf.createRequest(request,false); > newRequest.getSession().setAttribute("REQUEST", newRequest); > newRequest.getSession().setAttribute("PEER_SESSION", > otherSession); > otherSession.setAttribute("PEER_SESSION", > newRequest.getSession()); > newRequest.setHeader("X-SSVTPBX", "yes"); > newRequest.send(); > break; > } > } > > [3] UA3 Recevies the second call and it takes down. > > > I do not understand the cause of error. I annex you the code of simple sip > application example that replies to your problem. You can start it up and > checking if it works. Else if work for you, I would attempt watching the > particular case when a 5XX responses are received. > > Sorry... > > Best regards. > Antonio. > > -----Mensaje original----- > De: tele [mailto:tele@plexia.com] > Enviado el: lunes, 19 de marzo de 2007 14:33 > Para: Antonio Abajo Álvarez > Asunto: RE: [Users] WeSIP session question > > Hola Antonio, > > The scenario is more complex, i'll try to explain it: > > PSTN > | > | > UA---->GW ------> WeSIP(B2BUA) > | > | > mediaserver > > > GW: 82.215.163.5 > WeSIP: 82.215.163.67 > MS: 82.215.133.50 > > in the mediaserver there is an vxml script that i play in early media > and in case of particular event return to wesip a 503 temporaly > unavailable or e 410 Gone. So my B2BUA application have control of this > and can do stuff with the 503 and 410. > > in particular, in case of 503 temporaly unavailable i get the upstream > session and create a new invite to the media server for play another > announcement associated. in case of 410 gone i generate a new invite to > the GW with the original URI request for the correct termination. > > Yes when the session is removed i'm able to send another call. > > attached here the logs and the servlet. > > don't care about the hardcoded IP and the repeated code :-) > i'm doing testing.. > > regards, > > :tele > > > On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote: > > Hi Tele, > > > > I don't understand very well the problem., for what I understand you have > > the following: > > > > > > UA1 --------------------- WeSIP --------------------- UA2 > > > > ---INV/4XX-6XX/ACK (SS1) --> ----INV/4XX-6XX/ACK(SS2)--> > > > > You try to send another call and receive the 473 response of WeSIP... > > > > ------INV/473/ACK (SS1) --> > > > > And when the session has been removed you can send another call. > > > > If it is the case, I understand that the 473 response is send from > > application or from openser script configuration, because the internal > > behaviour of SIP doesn't send this response automatically. > > > > Can you verify if it is the case? > > > > Thank you very much... > > > > Antonio. > > > > -----Mensaje original----- > > De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre > > de tele > > Enviado el: lunes, 19 de marzo de 2007 12:35 > > Para: users(a)openser.org > > Asunto: [Users] WeSIP session question > > > > Hi, > > > > I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx > > correctly terminated, when i try to send another call to WeSIP i recive > > a "473 Filtered destination" then for send another call i've to wait > > WeSIP complete some management with session: > > > > 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: > > Recycling processor SipProcessor[3] > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e6006f] in state [3] with lifetime of :74490 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e68616] in state [3] with lifetime of :40281 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e6006f] in state [3] with lifetime of :134500 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e68616] in state [3] with lifetime of :100291 > > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 > > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- Remove AppSession > > [B4D6C9C4288784A68E032D20AB78BD8E] > > > > When i see Remove AppSession i'm able to send another call... > > I've read the sip servlet spec about that and i'm trying to invalidate() > > or setExpires() to SiApplicationSession in case of failed call. but it's > > not clear how to do yet. > > > > i can provide the full debug if needed. > > > > thank you! > > > > regards > > > > :tele > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49

1 0

[Users] DB migration
by Leonid Fainshtein 20 Mar '07

20 Mar '07

Hi, I didn't understand the proposed process of MySQL DB migration from ver.1.1 to 1.2 For example, the result of "openser_mysqldb.sh migrate openser openser12" will be a new database openser12. But how can I copy/rename openser12 to openser? Does it mean that from now, all of my auxiliary software must work with openser12? Thank you, Leonid Fainshtein ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

2 1

[Users] Help with dbtext location table - Almost there!
by Daryl Sanders 20 Mar '07

20 Mar '07

I have been struggling to get usrloc to work with... modparam("usrloc", "db_mode", 1) OpenSER 1.2 starts fine but gives the following error when a phone tries to register: 2(26634) db_insert_ucontact(): Error while inserting contact 2(26634) ERROR:usrloc:insert_ucontact: failed to insert in database Regards, Daryl

1 0

RV: [Users] WeSIP session question
by Antonio Abajo 20 Mar '07

20 Mar '07

Hola tele ;) I have tested an example similar to yours, and it works ok. It's de following: UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2 | [3] ---OK----_> UA3 [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA behaviour. In do INVITE. I have implemented the following: protected void doInvite(SipServletRequest invite){ SipServletRequest otherInvite = sf.createRequest(invite, false); SipURI sipUri = sf.createSipURI("UA2","proxy"); otherInvite.setRequestURI(sipUri); otherInvite.getSession().setAttribute("REQUEST", otherInvite); otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession()); invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession()); invite.getSession().setAttribute("REQUEST", invite); otherInvite.send(); } [2] UA2 declines the call and a 486 response is received by WeSIP and process it in doErrorResponse: protected void doErrorResponse(SipServletResponse errorResponse) { switch(errorResponse.getStatus()){ case 486: SipServletRequest request = (SipServletRequest) errorResponse.getSession().getAttribute("REQUEST"); SipSession otherSession = (SipSession) errorResponse.getSession().getAttribute("PEER_SESSION"); SipURI sipUri = sf.createSipURI("UA3", "proxy"); request.setRequestURI(sipUri); SipServletRequest newRequest = sf.createRequest(request,false); newRequest.getSession().setAttribute("REQUEST", newRequest); newRequest.getSession().setAttribute("PEER_SESSION", otherSession); otherSession.setAttribute("PEER_SESSION", newRequest.getSession()); newRequest.setHeader("X-SSVTPBX", "yes"); newRequest.send(); break; } } [3] UA3 Recevies the second call and it takes down. I do not understand the cause of error. I annex you the code of simple sip application example that replies to your problem. You can start it up and checking if it works. Else if work for you, I would attempt watching the particular case when a 5XX responses are received. Sorry... Best regards. Antonio. -----Mensaje original----- De: tele [mailto:tele@plexia.com] Enviado el: lunes, 19 de marzo de 2007 14:33 Para: Antonio Abajo Álvarez Asunto: RE: [Users] WeSIP session question Hola Antonio, The scenario is more complex, i'll try to explain it: PSTN | | UA---->GW ------> WeSIP(B2BUA) | | mediaserver GW: 82.215.163.5 WeSIP: 82.215.163.67 MS: 82.215.133.50 in the mediaserver there is an vxml script that i play in early media and in case of particular event return to wesip a 503 temporaly unavailable or e 410 Gone. So my B2BUA application have control of this and can do stuff with the 503 and 410. in particular, in case of 503 temporaly unavailable i get the upstream session and create a new invite to the media server for play another announcement associated. in case of 410 gone i generate a new invite to the GW with the original URI request for the correct termination. Yes when the session is removed i'm able to send another call. attached here the logs and the servlet. don't care about the hardcoded IP and the repeated code :-) i'm doing testing.. regards, :tele On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote: > Hi Tele, > > I don't understand very well the problem., for what I understand you have > the following: > > > UA1 --------------------- WeSIP --------------------- UA2 > > ---INV/4XX-6XX/ACK (SS1) --> ----INV/4XX-6XX/ACK(SS2)--> > > You try to send another call and receive the 473 response of WeSIP... > > ------INV/473/ACK (SS1) --> > > And when the session has been removed you can send another call. > > If it is the case, I understand that the 473 response is send from > application or from openser script configuration, because the internal > behaviour of SIP doesn't send this response automatically. > > Can you verify if it is the case? > > Thank you very much... > > Antonio. > > -----Mensaje original----- > De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre > de tele > Enviado el: lunes, 19 de marzo de 2007 12:35 > Para: users(a)openser.org > Asunto: [Users] WeSIP session question > > Hi, > > I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx > correctly terminated, when i try to send another call to WeSIP i recive > a "473 Filtered destination" then for send another call i've to wait > WeSIP complete some management with session: > > 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: > Recycling processor SipProcessor[3] > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e6006f] in state [3] with lifetime of :74490 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e68616] in state [3] with lifetime of :40281 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e6006f] in state [3] with lifetime of :134500 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e68616] in state [3] with lifetime of :100291 > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- Remove AppSession > [B4D6C9C4288784A68E032D20AB78BD8E] > > When i see Remove AppSession i'm able to send another call... > I've read the sip servlet spec about that and i'm trying to invalidate() > or setExpires() to SiApplicationSession in case of failed call. but it's > not clear how to do yet. > > i can provide the full debug if needed. > > thank you! > > regards > > :tele > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.13/726 - Release Date: 18/03/2007 15:34

2 1

[Users] mi_fifo question
by Andreas Granig 20 Mar '07

20 Mar '07

Hi, I have a OpenSER 1.2 with mi_fifo running, but adding an alias fails: /usr/local/openser-1.2# ./sbin/openserctl alias add 1000 sip:foo1000@mydomain.org database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 entering fifo_cmd ul_add aliases 1000(a)mydomain.org sip:foo1000@mydomain.org 0 1.00 0 0 4294967295 400 Too few or too many arguments FIFO command was: :ul_add:openser_receiver_17638 aliases 1000(a)mydomain.org sip:foo1000@mydomain.org 0 1.00 0 0 4294967295 In etc/openser/openserctlrc I have ALIASES_TYPE="UL", and the mi_fifo module config looks like this: 88 loadmodule "mi_fifo.so" 89 modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") 90 modparam("mi_fifo", "fifo_mode", 0660) 91 modparam("mi_fifo", "fifo_group", "openser") 92 modparam("mi_fifo", "fifo_user", "openser") 93 modparam("mi_fifo", "reply_dir", "/tmp/") 94 modparam("mi_fifo", "reply_indent", "\t") Am I missing something somewhere? Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

3 3

[Users] loose_route problem with 1.2
by jh＠tutpro.com 20 Mar '07

20 Mar '07

i have in openser.cfg file listen=x.x.x.x port=5060 alias="sip.test.fi:5060" tcp_accept_aliases=yes out-of-dialog request comes in with pre-loaded route header: Route: <sip:sip.test.fi;lr;transport=TCP> loose_route() returns true although according to rr/readme it should return false: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. this is with openser 1.2. i don't remember having this problems with openser 1.1.1. has something changed in config or what it is that i'm missing? -- juha

2 1

[Serusers] Find an issue with FIFO after crash
by inge 20 Mar '07

20 Mar '07

Hi all, After a crash, the server runs normally SER process. In /tmp/ I've my ser_fifo with same rights than an other server. But my SERWeb says : Warning: filetype(): Lstat failed for (null) (errno=13 - Permission denied) in /var/www/client/functions.php on line 332 FIFO not running or bad path to it, file: /var/www/client/functions.php:335 I've make a test by extracting the if bloc in functions.php near line 332. I put this in an other file. When I execute "php test.php", filetype is Okay. But when I access by http://.../test.php I've got the same error than serweb. Thanks for your support. Adrien

2 4

[Serusers] Unable to compile the xmlrpc module
by Fabian Dedu 20 Mar '07

20 Mar '07

Hi all, I'm trying to compile the ser-2.0.0+cvs20070315_src.tar.gz version and I'm obtaining the following error when trying to compile with xmlrpc support: Input: make include_modules="mysql xmlrpc" all ------------------------------------------------------------------------------------------------------------ make[1]: Entering directory `/downloads/ser2.0/ser/modules/mysql' make[1]: `mysql.so' is up to date. make[1]: Leaving directory `/downloads/ser2.0/ser/modules/mysql' make[1]: Entering directory `/downloads/ser2.0/ser/modules/xmlrpc' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -mcpu=athlon -Wall -DNAME='"ser"' -DVERSION='" 2.0.0+cvs20070309"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 3.2.3"' -D__CPU_i386 -D__OS_linux -DSER_VER=2000000 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -c xmlrpc.c -o xmlrpc.o xmlrpc.c: In function `open_doc': xmlrpc.c:1159: warning: implicit declaration of function `xmlReadMemory' xmlrpc.c:1160: `XML_PARSE_NOBLANKS' undeclared (first use in this function) xmlrpc.c:1160: (Each undeclared identifier is reported only once xmlrpc.c:1160: for each function it appears in.) xmlrpc.c:1161: `XML_PARSE_NONET' undeclared (first use in this function) xmlrpc.c:1162: `XML_PARSE_NOCDATA' undeclared (first use in this function) xmlrpc.c:1162: warning: assignment makes pointer from integer without a cast xmlrpc.c: In function `select_method': xmlrpc.c:1505: `XML_PARSE_NOBLANKS' undeclared (first use in this function) xmlrpc.c :1505: `XML_PARSE_NONET' undeclared (first use in this function) xmlrpc.c:1505: `XML_PARSE_NOCDATA' undeclared (first use in this function) xmlrpc.c:1505: warning: assignment makes pointer from integer without a cast make[1]: *** [xmlrpc.o] Error 1 make[1]: Leaving directory `/downloads/ser2.0/ser/modules/xmlrpc' make[1]: Entering directory `/downloads/ser2.0/ser/modules/acc_db' make[1]: `acc_db.so' is up to date. make[1]: Leaving directory `/downloads/ser2.0/ser/modules/acc_db' etc.... all the other modules are compiling okay ------------------------------------------------------------------------------------------------------------ I'm using Red Hat Enterprise AS 3.0 with all updates included. Did anyone experiencing the above compile issue regarding this new xmlrpc module ? Thank you, Fabian.

2 4

[Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Fabio Macchi 20 Mar '07

20 Mar '07

Hi all, I'm running the following schema: UA ( possibly natted ) -> SER -> PSTN Gateway I have a problem with UA belonging to a particular network with private address not RFC1918 compliant ( class 1.x.x.x ), SER and PSTN Gateway have pubblic address. The problem is that, after a succesfull call, if the PSTN gateway send a BYE to SER, then SER forward BYE to the private address of UA instead of pubblic one. I don't understand which is the section that handle BYE messages and how can I solve this problem: anyone may help ? Second, another question: with this particular network I had problem with INVITE too, because SER was sending "proxy authorization request" to the wrong TCP port. To solve this, I've moved the nat handling ( with force_rport ) before the proxy_authorize block and it's working, but why this is not necessary on standard RFC1918 compliant natted address ? Many thanks for any explanation Fabio

3 3

[Serusers] Heartbeat with ser and asterisk
by aespinoza＠vivophone.com 20 Mar '07

20 Mar '07

Hello people I have a heartbeat cluster that manages a ser 0.9.3 running on one machine and asterisk1.2.3 running on another in an Active/Active two IP address Configuration with failover support. I have SUSE 10.1 installed on both machines. My ha.cf files look like this ############################################### logfile /var/log/ha-log logfacility local0 keepalive 2 deadtime 10 warntime 10 initdead 20 udpport 694 baud 19200 bcast eth1 ping xxx.xxx.x.x auto_failback on node linux-xczz node prueba2 respawn hacluster /usr/local/lib/heartbeat/ipfail ################################################# My haresources files look like this ######################################## prueba2 xxx.xxx.x.125/24 safe_asterisk linux-xczz xxx.xxx.x.124/24 serctl ######################################## so "linux-xczz" is the master when running ser and "prueba2" is the master when running asterisk the authkeys files are the same on both machines too with the right permissions (mod 600). The /etc/hosts files look like this ########################## 10.10.10.1 linux-xczz 10.10.10.2 prueba2 ########################## The First time I try to run heartbeat on one machine (prueba2) with /etc/init.d/heartbeat start, both my services run good. But when I try to run heartbeat on the other machine (linux-xczz) so that it takes over the ser service, the system goes crazy and once linux-xczz takes over the ser service, prueba2 gives up the other resource (asterisk) which it should not do, and it appears on linux-xczz, only to disappear seconds later along with ser, leaving my cluster-ha a complete wreck with no service running on either machine. The error log I get from prueba2 is this: heartbeat[12609]: 2007/03/15_11:41:18 info: Link linux-xczz:eth1 up. heartbeat[12609]: 2007/03/15_11:41:18 info: Status update for node linux-xczz: status init heartbeat[12609]: 2007/03/15_11:41:18 info: Status update for node linux-xczz: status up harc[13730]: 2007/03/15_11:41:18 info: Running /etc/ha.d/rc.d/status status harc[13741]: 2007/03/15_11:41:18 info: Running /etc/ha.d/rc.d/status status heartbeat[12609]: 2007/03/15_11:41:19 info: Status update for node linux-xczz: status active harc[13754]: 2007/03/15_11:41:19 info: Running /etc/ha.d/rc.d/status status heartbeat[12609]: 2007/03/15_11:41:19 info: remote resource transition completed. heartbeat[12609]: 2007/03/15_11:41:19 info: prueba2 wants to go standby [foreign] heartbeat[12609]: 2007/03/15_11:41:20 info: standby: linux-xczz can take our foreign resources heartbeat[13767]: 2007/03/15_11:41:20 info: give up foreign HA resources (standby). ResourceManager[13777]: 2007/03/15_11:41:20 info: Releasing resource group: linux-xczz xxx.xxx.x.124/24 serctl ResourceManager[13777]: 2007/03/15_11:41:20 info: Running /etc/init.d/serctl stop ResourceManager[13777]: 2007/03/15_11:41:20 info: Running /etc/ha.d/resource.d/IPaddr xxx.xxx.x.124/24 stop IPaddr[13915]: 2007/03/15_11:41:20 INFO: /sbin/route -n del -host xxx.xxx.x.124 IPaddr[13915]: 2007/03/15_11:41:20 INFO: /sbin/ifconfig eth0:0 xxx.xxx.x.124 down IPaddr[13915]: 2007/03/15_11:41:20 INFO: IP Address xxx.xxx.x.124 released IPaddr[13836]: 2007/03/15_11:41:20 INFO: IPaddr Success heartbeat[13767]: 2007/03/15_11:41:20 info: foreign HA resource release completed (standby). heartbeat[12609]: 2007/03/15_11:41:20 info: Local standby process completed [foreign]. heartbeat[12609]: 2007/03/15_11:41:23 WARN: 1 lost packet(s) for [linux-xczz] [13:15] heartbeat[12609]: 2007/03/15_11:41:23 info: remote resource transition completed. heartbeat[12609]: 2007/03/15_11:41:23 info: No pkts missing from linux-xczz! heartbeat[12609]: 2007/03/15_11:41:23 info: Other node completed standby takeover of foreign resources. heartbeat[12609]: 2007/03/15_11:41:35 info: linux-xczz wants to go standby [foreign] heartbeat[12609]: 2007/03/15_11:41:36 info: standby: acquire [foreign] resources from linux-xczz heartbeat[14011]: 2007/03/15_11:41:36 info: acquire local HA resources (standby). ResourceManager[14021]: 2007/03/15_11:41:36 info: Acquiring resource group: prueba2 xxx.xxx.x.125/24 asterisk-rosa IPaddr[14048]: 2007/03/15_11:41:36 INFO: IPaddr Running OK ResourceManager[14021]: 2007/03/15_11:41:36 info: Running /etc/init.d/safe_asterisk start ResourceManager[14021]: 2007/03/15_11:41:36 ERROR: Return code 1 from /etc/init.d/safe_asterisk ResourceManager[14021]: 2007/03/15_11:41:36 CRIT: Giving up resources due to failure of safe_asterisk ResourceManager[14021]: 2007/03/15_11:41:36 info: Releasing resource group: prueba2 xxx.xxx.x.125/24 asterisk-rosa ResourceManager[14021]: 2007/03/15_11:41:xxz.xxz.x.xxz36 info: Running /etc/init.d/safe_asterisk stop ResourceManager[14021]: 2007/03/15_11:41:37 info: Running /etc/ha.d/resource.d/IPaddr xxx.xxx.x.125/24 stop IPaddr[14310]: 2007/03/15_11:41:37 INFO: /sbin/route -n del -host xxx.xxx.x.125 IPaddr[14310]: 2007/03/15_11:41:37 INFO: /sbin/ifconfig eth0:2 xxx.xxx.x.125 down IPaddr[14310]: 2007/03/15_11:41:37 INFO: IP Address xxx.xxx.x.125 released IPaddr[14231]: 2007/03/15_11:41:37 INFO: IPaddr Success heartbeat[14011]: 2007/03/15_11:41:37 info: local HA resource acquisition completed (standby). heartbeat[12609]: 2007/03/15_11:41:37 info: Standby resource acquisition done [foreign]. heartbeat[12609]: 2007/03/15_11:41:37 info: remote resource transition completed. heartbeat[12609]: 2007/03/15_11:41:38 WARN: G_CH_dispatch_int: Dispatch function for read child took too long to execute: 520 ms (> 50 ms) (GSource: 0x80fbe00) hb_standby[14375]: 2007/03/15_11:42:07 Going standby [foreign]. heartbeat[12609]: 2007/03/15_11:42:07 info: prueba2 wants to go standby [foreign] heartbeat[12609]: 2007/03/15_11:42:08 info: standby: linux-xczz can take our foreign resources heartbeat[14385]: 2007/03/15_11:42:08 info: give up foreign HA resources (standby). ResourceManager[14395]: 2007/03/15_11:42:08 info: Releasing resource group: linux-xczz xxx.xxx.x.124/24 serctl ResourceManager[14395]: 2007/03/15_11:42:08 info: Running /etc/init.d/serctl stop ResourceManager[14395]: 2007/03/15_11:42:08 ERROR: Return code 1 from /etc/init.d/serctl The error log I get from linux-xczz when I run heartbeat is this: heartbeat[1063]: 2007/03/15_14:26:12 WARN: Core dumps could be lost if multiple dumps occur heartbeat[1063]: 2007/03/15_14:26:12 WARN: Consider setting /proc/sys/kernel/core_uses_pid (or equivalent) to 1 for maximum supportability heartbeat[1063]: 2007/03/15_14:26:12 WARN: Logging daemon is disabled --enabling logging daemon is recommended heartbeat[1063]: 2007/03/15_14:26:12 info: ************************** heartbeat[1063]: 2007/03/15_14:26:12 info: Configuration validated. Starting heartbeat 2.0.7 heartbeat[1064]: 2007/03/15_14:26:12 info: heartbeat: version 2.0.7 heartbeat[1064]: 2007/03/15_14:26:12 info: Heartbeat generation: 130 heartbeat[1064]: 2007/03/15_14:26:12 info: G_main_add_TriggerHandler: Added signal manual handler heartbeat[1064]: 2007/03/15_14:26:12 info: G_main_add_TriggerHandler: Added signal manual handler heartbeat[1064]: 2007/03/15_14:26:12 info: Removing /usr/local/var/run/heartbeat/rsctmp failed, recreating. heartbeat[1064]: 2007/03/15_14:26:12 info: glib: UDP Broadcast heartbeat started on port 694 (694) interface eth1 heartbeat[1064]: 2007/03/15_14:26:12 info: glib: UDP Broadcast heartbeat closed on port 694 interface eth1 - Status: 1 heartbeat[1064]: 2007/03/15_14:26:12 info: glib: ping heartbeat started. heartbeat[1064]: 2007/03/15_14:26:12 info: G_main_add_SignalHandler: Added signal handler for signal 17 heartbeat[1064]: 2007/03/15_14:26:12 info: Local status now set to: 'up' heartbeat[1064]: 2007/03/15_14:26:13 info: Link linux-xczz:eth1 up. heartbeat[1064]: 2007/03/15_14:26:13 info: Link prueba2:eth1 up. heartbeat[1064]: 2007/03/15_14:26:13 info: Status update for node prueba2: status active heartbeat[1064]: 2007/03/15_14:26:13 info: Link xxx.xxx.x.x:xxx.xxx.x.x up. heartbeat[1064]: 2007/03/15_14:26:13 info: Status update for node xxx.xxx.x.x: status ping harc[1073]: 2007/03/15_14:26:13 info: Running /usr/local/etc/ha.d/rc.d/status status heartbeat[1064]: 2007/03/15_14:26:14 info: Comm_now_up(): updating status to active heartbeat[1064]: 2007/03/15_14:26:14 info: Local status now set to: 'active' heartbeat[1064]: 2007/03/15_14:26:14 info: Starting child client "/usr/local/lib/heartbeat/ipfail" (1001,100) heartbeat[1084]: 2007/03/15_14:26:14 info: Starting "/usr/local/lib/heartbeat/ipfail" as uid 1001 gid 100 (pid 1084) heartbeat[1064]: 2007/03/15_14:26:14 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:14 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:14 info: Local Resource acquisition completed. (none) heartbeat[1064]: 2007/03/15_14:26:15 info: prueba2 wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:26:15 info: standby: acquire [foreign] resources from prueba2 heartbeat[1088]: 2007/03/15_14:26:15 info: acquire local HA resources (standby). ResourceManager[1098]: 2007/03/15_14:26:15 info: Acquiring resource group: linux-xczz xxx.xxx.x.124/24 serctl IPaddr[1122]: 2007/03/15_14:26:16 INFO: IPaddr Resource is stopped ResourceManager[1098]: 2007/03/15_14:26:16 info: Running /usr/local/etc/ha.d/resource.d/IPaddr 192.168.1.124/24 start IPaddr[1321]: 2007/03/15_14:26:16 INFO: eval /sbin/ifconfig eth0:0 xxx.xxx.x.124 netmask 255.255.255.0 broadcast xxx.xxx.x.255 IPaddr[1321]: 2007/03/15_14:26:16 INFO: Sending Gratuitous Arp for xxx.xxx.x.124 on eth0:0 [eth0] IPaddr[1321]: 2007/03/15_14:26:16 INFO: /usr/local/lib/heartbeat/send_arp -i 500 -r 10 -p /usr/local/var/run/heartbeat/rsctmp/send_arp/send_arp-xxx.xxx.x.124 eth0 xxx.xxx.x.124 auto xxx.xxx.x.124 ffffffffffff IPaddr[1241]: 2007/03/15_14:26:16 INFO: IPaddr Success ResourceManager[1098]: 2007/03/15_14:26:16 info: Running /etc/init.d/serctl start heartbeat[1088]: 2007/03/15_14:26:17 info: local HA resource acquisition completed (standby). heartbeat[1064]: 2007/03/15_14:26:17 info: Standby resource acquisition done [foreign]. heartbeat[1064]: 2007/03/15_14:26:17 info: Initial resource acquisition complete (auto_failback) heartbeat[1064]: 2007/03/15_14:26:23 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:28 info: linux-xczz wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:26:28 info: standby: prueba2 can take our foreign resources heartbeat[1492]: 2007/03/15_14:26:28 info: give up foreign HA resources (standby). ResourceManager[1502]: 2007/03/15_14:26:28 info: Releasing resource group: prueba2 xxx.xxx.x.125/24 safe_asterisk ResourceManager[1502]: 2007/03/15_14:26:28 info: Running /etc/init.d/safe_asterisk stop ResourceManager[1502]: 2007/03/15_14:26:28 info: Running /usr/local/etc/ha.d/resource.d/IPaddr xxx.xxx.x.125/24 stop IPaddr[1561]: 2007/03/15_14:26:29 INFO: IPaddr Success heartbeat[1492]: 2007/03/15_14:26:29 info: foreign HA resource release completed (standby). heartbeat[1064]: 2007/03/15_14:26:29 info: Local standby process completed [foreign]. heartbeat[1064]: 2007/03/15_14:26:30 WARN: 1 lost packet(s) for [prueba2] [68:70] heartbeat[1064]: 2007/03/15_14:26:30 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:26:30 info: No pkts missing from prueba2! heartbeat[1064]: 2007/03/15_14:26:30 info: Other node completed standby takeover of foreign resources. heartbeat[1064]: 2007/03/15_14:27:00 info: prueba2 wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:27:11 info: standby: acquire [foreign] resources from prueba2 heartbeat[1784]: 2007/03/15_14:27:11 info: acquire local HA resources (standby). ResourceManager[1794]: 2007/03/15_14:27:11 info: Acquiring resource group: linux-xczz xxx.xxx.x.124/24 serctl IPaddr[1818]: 2007/03/15_14:27:11 INFO: IPaddr Running OK ResourceManager[1794]: 2007/03/15_14:27:11 info: Running /etc/init.d/serctl start ResourceManager[1794]: 2007/03/15_14:27:11 ERROR: Return code 1 from /etc/init.d/serctl ResourceManager[1794]: 2007/03/15_14:27:11 CRIT: Giving up resources due to failure of serctl ResourceManager[1794]: 2007/03/15_14:27:11 info: Releasing resource group: linux-xczz xxx.xxx.x.124/24 serctl ResourceManager[1794]: 2007/03/15_14:27:11 info: Running /etc/init.d/serctl stop ResourceManager[1794]: 2007/03/15_14:27:11 info: Running /usr/local/etc/ha.d/resource.d/IPaddr xxx.xxx.x.124/24 stop IPaddr[2090]: 2007/03/15_14:27:12 INFO: /sbin/route -n del -host xxx.xxx.x.124 IPaddr[2090]: 2007/03/15_14:27:12 INFO: /sbin/ifconfig eth0:0 xxx.xxx.x.124 down IPaddr[2090]: 2007/03/15_14:27:12 INFO: IP Address xxx.xxx.x.124 released IPaddr[2006]: 2007/03/15_14:27:12 INFO: IPaddr Success heartbeat[1784]: 2007/03/15_14:27:12 info: local HA resource acquisition completed (standby). heartbeat[1064]: 2007/03/15_14:27:12 info: Standby resource acquisition done [foreign]. heartbeat[1064]: 2007/03/15_14:27:12 info: remote resource transition completed. hb_standby[2228]: 2007/03/15_14:27:42 Going standby [foreign]. heartbeat[1064]: 2007/03/15_14:27:42 info: linux-xczz wants to go standby [foreign] heartbeat[1064]: 2007/03/15_14:27:42 info: standby: prueba2 can take our foreign resources heartbeat[2238]: 2007/03/15_14:27:42 info: give up foreign HA resources (standby). ResourceManager[2248]: 2007/03/15_14:27:43 info: Releasing resource group: prueba2 xxx.xxx.x.125/24 safe_asterisk ResourceManager[2248]: 2007/03/15_14:27:43 info: Running /etc/init.d/safe_asterisk stop ResourceManager[2248]: 2007/03/15_14:27:43 info: Running /usr/local/etc/ha.d/resource.d/IPaddr xxx.xxx.x.125/24 stop IPaddr[2310]: 2007/03/15_14:27:43 INFO: IPaddr Success heartbeat[2238]: 2007/03/15_14:27:43 info: foreign HA resource release completed (standby). heartbeat[1064]: 2007/03/15_14:27:43 info: Local standby process completed [foreign]. heartbeat[1064]: 2007/03/15_14:27:44 WARN: 1 lost packet(s) for [prueba2] [114:116] heartbeat[1064]: 2007/03/15_14:27:44 info: remote resource transition completed. heartbeat[1064]: 2007/03/15_14:27:44 info: No pkts missing from prueba2! heartbeat[1064]: 2007/03/15_14:27:44 info: Other node completed standby takeover of foreign resources. What I am trying to have is ser running on linux-xczz and asterisk running on prueba2 with failover configured on both machines but apparently the failover crashes and I lose both my services if both heartbeats are running. Any idea why this happens or what I'm doing wrong?. Can ser and asterisk be run by heartbeat with failover support? thanxs in advance

2 1

[Serusers] Question about Radius Leg
by Gustavo Passos Tourinho 20 Mar '07

20 Mar '07

Hello I saw a Stop-Ack LEG and I would like to know what kind of signaling message could generate it. I mean, that 183 is Session Progress I know but why it generate Stop Ack leg? Shouldnt I have only start-ack (ACK send after 200 OK was received)? This is a quote of my radius log Thu Mar 15 13:49:18 2007 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 183 Error-Cause = Ack Thanks, Gustavo

2 1

[Serusers] compiling the most recent unstable snapshots
by Eliott Spencer 20 Mar '07

20 Mar '07

I am having difficulty compiling the most recent unstable snapshots on Debian (Ubuntu Feisty Fawn) starting with the version released March 3rd. Previous versions compile well so I have been using the March 1st release and am learning a lot about the new SER (I really like @selects). My compile gets stuck at the eval module but when I skip that module, it gets stuck at the unixsock module. If I skip that module it gets stuck at another module and so on until when I finish compiling it won't start. Am I missing a key step or command? Thanks!

2 1

[Serusers] Problem loading mysql module
by POON Ben 20 Mar '07

20 Mar '07

Hi, I installed SER 0.9.3 for Solaris 10 (is there a newer version pre-compiled for solaris 10?). I'm using the sample config it comes with, except I uncommented the line: loadmodule "/usr/local/lib/ser/modules/mysql.so" Having done that, I get this error: 0(23162) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: ld.so.1: ser: fatal: libmysqlclient.so.10: open failed: No such file or directory I have installed MySQL already, and I checked that mysql.so is where it should be. Is there something else I have to configure to get MySQL running with SER? Thanks, Ben

4 3

[Serusers] Ser2 and radius - Accounting request doesn't send User-Name Attribute
by tzieleniewski 20 Mar '07

20 Mar '07

Hi, I am using radius for accounting and I have the following problem that whenever SER sends an Accounting message to Radius which contains the Acct-Status-Type = Start it doesn't send the User-Name attribute. Strange is that the User-Name is send with the massage with Acct-Status-Type = Stop. This disables radius to perform the correct accounting statistics because radius is unable to map the start and end of the session. Can it be somehow enabled in ser to send User-Name parameter in the first message? Below is an example of such two corresponding accounting messages: Mon Mar 12 14:46:17 2007 SER-Attr = "" Acct-Session-Id = "9e6a674c-0fcf-db11-94ea-00148539376e@sen" Sip-To-Tag = "36974326" SER-From = "\"Sen Server\" <sip:tzl@voip.touk.pl>;tag=406e674c-0fcf-db11-94ea-00148539376e" SER-Flags = 8 SER-Original-Request-ID = "sip:110@voip.touk.pl" Sip-Method = "INVITE" Sip-Cseq = 2 Sip-Translated-Request-ID = "sip:hellboy@192.168.0.105:7060" Sip-Source-IP-Address = 192.168.0.165 Sip-From-Tag = "406e674c-0fcf-db11-94ea-00148539376e" SER-To = "<sip:110@voip.touk.pl>;tag=36974326" SER-Digest-Username = "tzl" SER-Request-Timestamp = 1173707170 Calling-Station-Id = "sip:tzl@voip.touk.pl" Sip-Source-Port = 5067 SER-Digest-Realm = "voip.touk.pl" Sip-Response-Code = 200 Called-Station-Id = "sip:110@voip.touk.pl" SER-Response-Timestamp = 1173707177 Acct-Status-Type = Start Service-Type = IAPP-Register NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "16bd1ddc2a514fa0" Timestamp = 1173707177 Mon Mar 12 14:47:10 2007 SER-Attr = "" Acct-Session-Id = "9e6a674c-0fcf-db11-94ea-00148539376e@sen" Sip-To-Tag = "36974326" SER-From = "\"Sen Server\" <sip:tzl@voip.touk.pl>;tag=406e674c-0fcf-db11-94ea-00148539376e" SER-Flags = 8 SER-Original-Request-ID = "sip:hellboy@192.168.0.105:7060" Sip-Method = "BYE" Sip-Cseq = 4 Sip-Translated-Request-ID = "sip:hellboy@192.168.0.105:7060" Sip-Source-IP-Address = 192.168.0.165 Sip-From-Tag = "406e674c-0fcf-db11-94ea-00148539376e" SER-To = "<sip:110@voip.touk.pl>;tag=36974326" SER-Request-Timestamp = 1173707230 Calling-Station-Id = "sip:tzl@voip.touk.pl" Sip-Source-Port = 5067 Sip-Response-Code = 200 Called-Station-Id = "sip:110@voip.touk.pl" SER-Response-Timestamp = 1173707230 Acct-Status-Type = Stop Service-Type = IAPP-Register User-Name = "tzl(a)voip.touk.pl" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "4a9f1334616db4a4" Stripped-User-Name = "tzl" Realm = "voip.touk.pl" Timestamp = 1173707230

2 1

Re: [Serusers] SIP Accounting Module problem
by Greger V. Teigre 20 Mar '07

20 Mar '07

Copy serusers. Private emails that should go on SER end up in a different inbox and get lower priority... I'm not a solaris person, but ser logs to syslog. g-) POON Ben wrote: > Hi Greger, > > Thanks for the tip. I'm using Solaris 10 and whatever Ser version > it's bundled with, everything came preinstalled. I went to /var/log, > but there's no messages folder, all I have a syslog.# files. How can > I find out where the log messages are kept, if there are any at all? > Is that something I can set in the ser.cfg file? > > Thanks, > Ben > > ------------------------------------------------------------------------ > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Friday, March 09, 2007 1:56 AM > *To:* POON Ben > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] SIP Accounting Module problem > > Ben, > the cat error just tells you that ser didn't start. Find the command > in your ser service script that actually starts ser and start it > manually. You may also add -E -D -D -D -D -D > to get lots of debug info to stderr (or just look in your > /var/log/messages) > g-) > > POON Ben wrote: >> Hi folks, >> >> I'm trying to get my accounting services working, but am having >> trouble doing so... I am using the sample ser.cfg file, and am >> having a problem loading the acc.so module. >> >> In the ser.cfg file, I try added the line: loadmodule >> "/usr/sfw/lib/ser/modules/acc.so" to the bottom of the loadmodule >> section. I haven't done any modparam or scripting yet, even so, when >> I restart the ser service, I get: Starting SER : cat: cannot open >> /var/run/ser.pid. >> >> If I comment or remove the loadmodule line, ser will load again. Is >> there anything else I need to add or configure before loading this >> module? I went to /usr/sfw/lib/ser/modules to make sure acc.so is >> there, and it is. >> >> Any help is greatly appreciated. Thank you very much. >> Ben >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >>

1 0

[Serusers] Change URI before accounting ?
by inge 20 Mar '07

20 Mar '07

Hi all, When I modify an URI by stripping some words, in ACC table I've got the old URI, before the stripping. So if INVITE contains sip:7000123456789, after strip he contains sip:0123456789, in acc I've INVITE ACK and BYE with sip:7000123456789. How can I solve it ? Thanks for your help. Regards, Adrien

3 11

[Serusers] SER home proxy with Asterisk voicemail
by Meidiana Ten 20 Mar '07

20 Mar '07

Hi guys, I'm give up with SEMS as media server for voicemail cause it can't successly run although I've followed how to install SEMS at http://ftp.iptel.org/pub/sems/doc/current/Configure-Sems-Ser-HOWTO.html and there is no answer for the problem So I decide to try Asterisk as the voicemail server. But I never use Asterisk before. I really need help in configuring Asterisk. What should I have (the softwares) before compiling and configuring Asterisk as voicemail server. Thanx for the attention. Regards, Meidiana ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

5 9

[Serusers] Call forwarding
by Jai Rangi 20 Mar '07

20 Mar '07

Hello, I have a ser and asterisk setup. Users register on ser and all the national and international calls go through asterisk. Asterisk handles the billing part for long distance calls. Now I got an issue with call forwarding. Here is the situation. Useragent A calls Usergent B. User B has setup the call the call forwarding to his cell phone. So when user A calls user B, ser check that user B has set the forwarding and send the call to Asterisk box. Asterisk box sees user A in the callerID and bills user A for that call. We want user B to be billed for that call, cause he is forwarding the calls to his cell phone (long distance number). How can I achieve this. Thank you, -Jai

1 0

[Users] permissions module and avp definition
by Andreas Granig 19 Mar '07

19 Mar '07

Hi, It seems like most (all?) of the module params defining an AVP have changed from just the name (like "i:100") to a full avp spec (like "$avp(i:100)"). Does this also hold true for peer_tag_avp in permissions module? I ask because it's still "i:707" in the example of http://openser.org/docs/modules/1.2.x/permissions.html#AEN343 and it's not mentioned in http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x Regards, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

2 1

[Users] t_uac_dlg disconnect example - WILLING TO PAY
by Daryl Sanders 19 Mar '07

19 Mar '07

I need a complete example (with proper syntax) of how to disconnect a call (send BYE) using "t_uac_dlg". This can be either via bash script or openserctl command. I have been fighting with this for some time and always seem to get "Too few or to many parameters" errors. I have been through the ctd.sh example and searched the mailing list but, I'm still having no luck. I have read Jan's "SER FIFI Interface" doc, and I believe my fifo files are working properly, however I really don't know the exact syntax, or what elements are required to get this to work. I have access to the "Call-ID" as it is stored in MySQL via acc. I am willing to pay for this example. Just let me know a fair price. FYI: I am using OpenSER 1.2. Thanks, Daryl

2 1

[Users] Permissions module w/DBTEXT database doesn't work in 1.2 ???
by Daryl Sanders 19 Mar '07

19 Mar '07

Hi Everyone, I cannot seem to get the permissions module with dbtext to load in version 1.2. I am pretty sure the "trusted" table format is correct. This is what I have. src_ip(str) proto(str) from_pattern(str) tag(str,null) 192.168.0.10:any:^sip\:.*$: I get the following error when I start OpenSER... ERROR: permissions: reload_address_table(): Error while querying database Regards, Daryl

3 5

[Users] using avp's as lists
by Kro Kamil 19 Mar '07

19 Mar '07

Hi, can i use avp's as a list holder? what i want to do is using avp_db_query function to get the list of routes which will be used for forwarding calls. for example avp_db_query("select route_ip from tbl_route","$avp(s:route_list)"); then rewritehost("$avp(s:route_list)[1]"); on failure rewritehost("$avp(s:route_list)[2]"); thanks in advance _________________________________________________________________ Live.com'u deneyin: çevrimiçi dünyanızı bir araya getirin; haberler, spor, hava durumu ve çok daha fazlası. http://www.live.com/getstarted

2 1

[Users] Manipulate Refer To
by Sam Lee 19 Mar '07

19 Mar '07

Hey guys, How do you manipulate the 'Refer to' header in the REFER Method ? I need to do some string manipulation to change it to something that i want. Thanks ahead! Regards, Sam

3 8

[Users] Server call forwarding
by Sam Lee 19 Mar '07

19 Mar '07

Hey guys, I'm thinking of try out a server side forwarding where forwarded number is stored in a database. I just want to ask if anyone has done it before and how was the call routing done . Been thinking alot but couldn't figure out a way to do it. Any pointers ? Regards, Sam

3 2

[Users] WeSIP session question
by tele 19 Mar '07

19 Mar '07

Hi, I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx correctly terminated, when i try to send another call to WeSIP i recive a "473 Filtered destination" then for send another call i've to wait WeSIP complete some management with session: 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: Recycling processor SipProcessor[3] 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e6006f] in state [3] with lifetime of :74490 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 14:10:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e68616] in state [3] with lifetime of :40281 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e6006f] in state [3] with lifetime of :134500 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 14:11:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- SipSession [z9hG4bK69e68616] in state [3] with lifetime of :100291 14:12:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- AppSession Id [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 14:12:56 19Mar2007 DEBUG StandardAppSessionManager [StandardAppSessionManager[/inapp]]- Remove AppSession [B4D6C9C4288784A68E032D20AB78BD8E] When i see Remove AppSession i'm able to send another call... I've read the sip servlet spec about that and i'm trying to invalidate() or setExpires() to SiApplicationSession in case of failed call. but it's not clear how to do yet. i can provide the full debug if needed. thank you! regards :tele

1 0

[Users] WeSIP with OpenSER 1.2.0
by tele 19 Mar '07

19 Mar '07

Hi, I've upgraded openser from 1.2.0-pre6 to 1.2.0 with the seas module loaded. try to send a call and this is the output: 5(24664) SIP Request: 5(24664) method: <INVITE> 5(24664) uri: <sip:390104491079@82.215.163.67> 5(24664) version: <SIP/2.0> 5(24664) parse_headers: flags=2 5(24664) Found param type 232, <branch> = <z9hG4bK5d297af7>; state=16 5(24664) end of header reached, state=5 5(24664) parse_headers: Via found, flags=2 5(24664) parse_headers: this is the first via 5(24664) After parse_msg... 5(24664) preparing to run routing scripts... 5(24664) parse_headers: flags=100 5(24664) DEBUG:maxfwd:is_maxfwd_present: value = 70 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) DEBUG:parse_to:end of header reached, state=9 5(24664) DBUG:parse_to: display={}, ruri={sip:390104491079@82.215.163.67} 5(24664) DEBUG: get_hdr_field: <To> [32]; uri=[sip:390104491079@82.215.163.67] 5(24664) DEBUG: to body [sip:390104491079@82.215.163.67 ] 5(24664) get_hdr_field: cseq <CSeq>: <1> <INVITE> 5(24664) DEBUG: get_hdr_body : content_length=213 5(24664) found end of header 5(24664) DEBUG: t_newtran: T on entrance=0xffffffff 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) parse_headers: flags=78 5(24664) t_lookup_request: start searching: hash=4873, isACK=0 5(24664) DEBUG: RFC3261 transaction matching failed 5(24664) DEBUG: t_lookup_request: no transaction found 5(24664) DEBUG: add_param: tag=5d297af7 5(24664) DEBUG:parse_to:end of header reached, state=29 5(24664) DBUG:parse_to: display={}, ruri={sip:3405300695@82.215.163.5} 5(24664) new INVITE 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) check_via_address(82.215.163.5, 82.215.163.5, 0) 5(24664) WARNING:vqm_resize: resize(0) called 5(24664) DEBUG:tm:_reply_light: reply sent out. buf=0x814fa58: SIP/2.0 1..., shmem=0xb60a0680: SIP/2.0 1 5(24664) DEBUG:tm:_reply_light: finished 5(24664) as found ! (app_server_one) processor id = 1 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) parse_headers: flags=ffffffffffffffff 5(24664) msglen = 561,msg starts at 167 5(24664) DEBUG:tm:UNREF_UNSAFE: after is 0 5(24664) DEBUG:destroy_avp_list: destroying list (nil) 5(24664) receive_msg: cleaning up -1(0) Event relaied to app_server_one AS -1(0) Event type FINAL_REPLY -1(0) Scanning data from 1 AS -1(0) Scanning data from 0 uncomplete AS -1(0) polling [2 ServSock] [1 pipe] [1 App Servers] [0 Uncomplete AS] java.lang.StringIndexOutOfBoundsException: String index out of range: -19 at java.lang.String.checkBounds(String.java:368) at java.lang.String.<init>(String.java:404) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:30) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPRequest.<init>(SIPRequest.java:154) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:355) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:135) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:141) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595)

2 2

[Users] Failure routes
by Sam Lee 19 Mar '07

19 Mar '07

Hey guys , I noticed that only 1 failure routes is valid for a t_relay . Is that true or i've done wrong with my config. What i did was : If condition A { t_on_failure("1") } If condition B { T_on_failure("2") } t_relay If it can't be done , how am i suppose to do condition A or B ? Regards, Sam

2 3

[Users] error adding a new subscriber
by rkmr.em＠gmail.com 19 Mar '07

19 Mar '07

Hi I am adding a subscriber and I get this error below: /usr/local/sbin/openserctl add 100 100 100@localhost database engine 'PGSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter= ERROR: user '100' already exists as alias I am sure this user does not exist as I checked the table subscriber and there is only one row with user 'admin' Thanks mark

2 4

[Users] RTPProxy problem with music-on-hold
by Federico Giannici 18 Mar '07

18 Mar '07

I have a problem with RTPProxy: often when I put a call on hold the callee hear no music. No problem if I don't pass through RTPPropxy. The strangest thing is that, if I do a lot of tries, I eventually succeed to make the callee to hear the music! I put rtpproxy in debug mode and found that when I finally succeed the usual line: callee's address filled in: 83.211.227.15:63829 (RTCP) is replaced by: callee's address filled in: 83.211.227.15:63828 (RTP) guessing RTCP port for callee to be 63829 I don't know RTP enough to understand if all these is normal or some kind of bug. Somebody have a clue? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

1 0

[Users] openser + postgres starting problem
by rkmr.em＠gmail.com 18 Mar '07

18 Mar '07

Hi If I compiled openser with postgres module and I get the following errors: [mark@localhost|~|16:34:34] /usr/local/sbin/openserctl moni /usr/local/lib/openser/openserctl/openserctl.pgsql: line 57: unexpected EOF while looking for matching `'' /usr/local/lib/openser/openserctl/openserctl.pgsql: line 70: syntax error: unexpected end of file database engine 'PGSQL' loaded Control engine 'FIFO' loaded ERROR: Error opening OpenSER's FIFO FIFO ERROR: Make sure you have line 'fifo=FIFO' in your config The line 57 3rd line in here: # input: sql query, optional pgsql command-line params pgsql_ro_query() { mdbg "pgsql_ro_query: $PGSQL $2 -h $DBHOST -U $DBROUSER $DBNAME -c '$1'" PGPASSWORD="$DBROPW" $PGSQL $2 \ -h $DBHOST \ -U $DBROUSER \ $DBNAME \ -c "$1" } How to fix this? thanks

3 9

Re: Fw: [Users] TM : retransmission timers
by Rao Ramaratnamma 18 Mar '07

18 Mar '07

Hi Weiter, Yeah, I have been trying to limit myself to technical observations too, but the governance aspect is somewhat interesting too as a hint for future development, even though I guess even this is much more confusing than the technical ones. I have investigated, both projects have their firms with them that pursue their commercial interests which creates a risk of possibly departing from the public interest, like with redhat. From this angle they look quite similar. But if any worries me just a little bit more than openser. Appearance at commercial shows on the "open" side versus technical event on the "net" side if I take your BSD parallel, marketing "open" webpage accusing "net" version bad, hiding root commerical sponsors on the "open" webpage, this could be signs for a redhat-like doubleedged sword. Hopefully I am oversensing because I mean it is natural that everybody has SOME interest, but indisputably folks on both sides have done good work, but same indisputably more TRANSPARENCY would be helpful for both projects so that users can be less investigative. But I agree the technical comparison you suggest will be very useful if not most useful. This is what I am eventually upto. Anything folks have to tell in this topic is most welcome like the retransmission timers in subject or user loading. rr disconcerted by the fact that the more I know the more I am confused and determined to get over the learning curve quickly. also excuse the abuse I crossposted again but I think cross interrogation is a bit painful but the more effective :-) ----- Original Message ---- From: Weiter Leiter <bp4mls(a)googlemail.com> To: Kim Il <kim_il_s(a)yahoo.com> Cc: users(a)openser.org Sent: Thursday, November 9, 2006 1:42:29 PM Subject: Re: Fw: [Users] TM : retransmission timers Common user barely has time to meet his boss requirements, rather than playing around with different scenarios, platforms, environments. I only read one email where Daniel stated that OpenSER now performs a whole much better while loading users from database. SER guys put no figure out yet, neither bare numbers nor comparisons. I'm just really curious to see how both servers perform, that's all. Even though I must maintain my SER, I kinda like OpenSER's faster releases and developers' responsiveness (that I shamelessly exploit for the common code left there :-), which is pretty much nonexistent with iptel (at least this is the general belief here at OpenSER). But about this I'll probably have to fight on SER's mailing list. I still wish that one day I won't have to compare features; heck, NetSER and FreeSER are still available ;-). WL. PS. Maybe regretfully, I haven't seen any iptel booth at von this year, while OpenSER guys put up a nice show. My congrats. On 11/9/06, Kim Il <kim_il_s(a)yahoo.com> wrote: I can see what you are hinting at, but I guess that the users are the unbiased party that should do the judgment and not the parties who have something to gain. cheers Weiter Leiter <bp4mls(a)googlemail.com> wrote: This features comparisons are not to last for too long, some performance comparisons would also be nice. After all, there are plenty of UA-level stacks out there. At least now that both projects get to have stable releases after forking and some core functionality remained shared. I wonder what "unbiased" organization will take up the challenge. :-) On 11/8/06, Kim Il < kim_il_s(a)yahoo.com > wrote:Mike, this is a really good start and we should collect these things so as to help the community to take the right choice. I would also suggest that what ever ground breaking issues we list we stay at the functional level (I do not think anyone is helped by using a description containing "allowing carrier grade platforms" and similar marketing phrases). cheers {truncated because too large} Sponsored Link Talk more and pay less. Vonage can save you up to $300 a year on your phone bill. Sign up now. _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

19 63

[Users] Openser architectural question
by vallimamod abdullah 17 Mar '07

17 Mar '07

Hello List, I have currently one openser proxy running on an Intel Xeon 2.8GHz and which is offering different sip customer services. It works fine, the only problem is the complexity of the configuration. For better security and easier maintenance, I have imagined to have one openser instance per service running on a virtual server (linux vserver). I would like to hear from the community opinion about the kind of problems I may encounter with this schema (performance limitations over all). If somebody have already built such architecture, I'll be happy to learn from his experience ! Thanks. -- Regards, -vma .

2 1

[Serusers] SER doesn't load but system thinks it is
by POON Ben 16 Mar '07

16 Mar '07

Hi all, I am trying to load ser, and it says it does. But when I do a ps -el | grep ser, it doesn't show up... I am running Solaris 10, ser 0.9.3. Can anyone please give me a hand? This is what I did: ************************************************************************ ****************** bash # ./serctl start Starting SER : started pid(532) bash # ps -el | grep ser 0 S 0 421 402 0 40 20 ? 695 ? ? 0:00 htt_serv 0 S 0 384 1 0 40 20 ? 253 ? ? 0:00 dpkeyser 0 S 0 390 1 0 40 20 ? 232 ? ? 0:00 jserver 0 S 0 391 390 0 40 20 ? 328 ? ? 0:00 jserver_ bash # ./serctl restart Stopping SER : ./serctl: line 812: kill: (532) - No such process stopped Starting SER : PID file exists! (/var/run/ser.pid) already running? ************************************************************************ ****************** The system thinks it's already running, but there's no actual process started... Any help is greatly appreciated. Thank you very much, Ben

2 1

[Users] OT: Troubleshooting DNS SRV records
by kjcsb 16 Mar '07

16 Mar '07

I'm trying to get SRV records working and am having difficulties. I wonder if anyone could point me in the direction of some useful information on this topic to assist in troubleshooting. I've read a lot of stuff about how to set it up and check the records, but since I'm still having problems, there's something else wrong and I've run out of ideas. nslookup set type=srv _sip._tcp.domain.co.nz Server: 64.38.5.242 Address: 64.38.5.242#53 Non-authoritative answer: _sip._tcp.domain.co.nz service = 10 10 5060 domain.co.nz. _sip._tcp.domain.co.nz service = 20 10 5060 beta.domain.co.nz. Authoritative answers can be found from: domain.co.nz nameserver = ns1.widge.net. domain.co.nz nameserver = ns2.widge.net. > _sip._udp.domain.co.nz Server: 64.38.5.242 Address: 64.38.5.242#53 Non-authoritative answer: _sip._udp.domain.co.nz service = 10 10 5060 domain.co.nz. _sip._udp.domain.co.nz service = 20 10 5060 beta.domain.co.nz. Authoritative answers can be found from: domain.co.nz nameserver = ns2.widge.net. domain.co.nz nameserver = ns1.widge.net. One thing I notice from this is that no IP addresses are returned. Is that normal? The problems I am experiencing are that neither my Grandstream BT102 or SPA3000 can resolve the proxy name using SRV (when use SRV is set to on in those devices). In the case of the BT102 it won't register (switching off use SRV allows it to register). In the case of the SPA3000 when I view the syslog output I can see it failing over from the SRV record to the A record which then resolves. Any help appreciated. Cameron

2 1

[Users] Re: help in uac config
by vallimamod abdullah 16 Mar '07

16 Mar '07

On 3/16/07, Jobson Andrade <jandrade(a)obelisknet.com.br> wrote: > I made this configuration but not work, i need send International calls for > my gateway using one identification (username and password) only this more > not work my configuration. I though you just wanted to change the From header. The 'uac_replace_from()' just changes this header in the sip message. To make client authentification, you need to configure a failure route and use 'uac_auth()' in it. There is an example showing exactly what you are looking for at: http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth -- Regards, -vma .

1 0

[Users] Re: help in uac config
by vallimamod abdullah 16 Mar '07

16 Mar '07

Hello Jobson, On 3/16/07, "Jobson Andrade" <jandrade(a)obelisknet.com.br> wrote: > Hello, > > I need one help in my configuration using authentication > > Bellow is my configuration more not work, i need one help for work this > configuration > > modparam("uac","credential","username:itsp.com.br:password") > > ##Send Call For DDI > if (uri=~"^sip:00[1-9][1-9][0-9]*@") { > route(3); > exit; > } > > route[3]{ > #enviando route to hitconferencing > strip(2); > uac_replace_from("credential"); > route(1); > exit; > } > > This is correct? IMHO, it shoud be, according to your example: uac_replace_from(" sip:username@itsp.com.br"); Here are 2 pointers to interesting documentation and example: http://www.openser.org/docs/modules/1.2.x/uac.html#AEN174 http://www.voice-system.ro/docs/uac/ar01s06.html Hope this will help. -- Regards, -vma .

2 1

[Users] Openser = 1.1.1 init_tls: null openssl compression methods
by Bodin Bruno 16 Mar '07

16 Mar '07

I use Fedora Core 5 and when I launch openser in TLS mode : 0(13701) init_tls: Entered 0(13701) WARNING:init_tls: disabling compression due ZLIB problems 0(13701) ERRRO:init_tls: null openssl compression methods 0(13701) could not initialize tls, exiting... How can I fix this ? Thank :)

1 0

R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem
by Fabio Macchi 16 Mar '07

16 Mar '07

Hello, I'm using Nathelper + RTPproxy ( can't use Nathelper-only because my gateway doesn't support passive mode ) : route part of ser.cfg attached. In the invite block I use force_rport before authorize block ( as previous discussed ) and the NAT block after: route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } The original BYE comes from Gateway with the wrong request URI ( private UA ip ) Request-Line: BYE sip:xxxxxx@1.255.17.9 SIP/2.0 Method: BYE [Resent Packet: False] Fabio -----Messaggio originale----- Da: Kostas Marneris [mailto:K.Marneris@otenet.gr] Inviato: venerdì 16 marzo 2007 13.19 A: Fabio Macchi Cc: Kostas Marneris Oggetto: Re: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem Hello, Which 'NAT Traversal mechanism' do you use ? Nathelper only or Mediaproxy ? Do you change the Contact address somewhere in INVITE block ? force_rport(); fix_nated_contact(); fix_nated_sdp("3"); Check (with a tcpdump at SER) how the original BYE comes from GW. What is the R-URI at the original BYE request (comes from GW) ? Kostas Fabio Macchi wrote: > First, thanks for answer. > > I've tryed your trik and in effect this solve the problem of the '200 ok' > forwarded to the UA, but my problem still remain alive: when BYE is sent > from Gateway, it reaches correctly SER, but it still forward it to the > private UA address. I was wondering about the nat_uac_test in this case, as > the source of the BYE message is the gateway ( not natted ) and not the UA. > > Have any idea about this ? > > Fabio > > -----Messaggio originale----- > Da: Kostas Marneris [mailto:K.Marneris@otenet.gr] > Inviato: giovedì 15 marzo 2007 20.39 > A: Fabio Macchi > Cc: serusers(a)lists.iptel.org > Oggetto: Re: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem > > Hello, > I was working on about the same problem today either with 'Mediaproxy > solution' > or with 'SER's Nathelper only solution' . > > The NAT issue is a nightmare, not because of SER but because of > different implementations on NAT boxes. > > Actually my problem was : > if the NATed UA send a BYE to SER, SER forward it to PSTN-GW, > then the '200 Ok' Response from PSTN-GW is forwarded by SER to UA > to the wrong port (Contact or Via header port). > > I used the following block on Loose Route section, > (because BYE is loose_routed if you use Record-Route), > and it seems to work. > > # --------------------------------------- > # Loose Route Section > # --------------------------------------- > if (loose_route()) { > # mark routing logic in request > if (method == "BYE") { > if (nat_uac_test("22")) { > xlog("L_NOTICE", "*** LR -> NATed BYE - Use > force_rport()"); > force_rport(); > }; > }; > route(1); > break; > }; > > > > > I faced up your second problem too. > The solution was to move the NAT handling block before proxy_authorize > block. > > I think that the different behaviour does not come with the 'standard > RFC1918 addresses', > but with the different NAT type. > > I realize that the provisional mesgs '100 Trying' and '407 Proxy > Authentication Required' > are relayed back to the real IP addr of NATed UA (this is correct), > but to the WRONG port (that of Contact/Via header and not the signalling > received port). > It seems that these mesgs use the IP address part of 'Received' field of > Location DB > but not the port. > > It happens to work if NAT box use the SAME port (eg. 5060) on NAT > translation > (10.10.10.1:5060 --> Real_IP:5060) (eg. with a SAGEM1500 Router) > But it does not work if NAT box doesn't use the same port > (10.10.10.1:5060 --> Real_IP:38181) > > > I think that this has to be verified by SER developers or SER experts. > > > > Kostas > > --- > K.Marneris(a)otenet.gr > > ----- Original Message ----- > From: "Fabio Macchi" <f.macchi(a)keeptelecom.com> > To: <serusers(a)lists.iptel.org> > Sent: 15 March 2007 19:34 > Subject: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem > > >> Hi all, >> >> >> >> I'm running the following schema: >> >> >> >> UA ( possibly natted ) -> SER -> PSTN Gateway >> >> >> >> I have a problem with UA belonging to a particular network with private >> address not RFC1918 compliant ( class 1.x.x.x ), SER and PSTN Gateway have >> pubblic address. >> >> >> >> The problem is that, after a succesfull call, if the PSTN gateway send a > BYE >> to SER, then SER forward BYE to the private address of UA instead of > pubblic >> one. >> >> >> >> I don't understand which is the section that handle BYE messages and how > can >> I solve this problem: anyone may help ? >> >> >> >> >> >> >> >> Second, another question: with this particular network I had problem with >> INVITE too, because SER was sending "proxy authorization request" to the >> wrong TCP port. To solve this, I've moved the nat handling ( with >> force_rport ) before the proxy_authorize block and it's working, but why >> this is not necessary on standard RFC1918 compliant natted address ? >> >> >> >> Many thanks for any explanation >> >> >> >> Fabio >> >> > > > > -- Kostas Marneris e-mail: K.Marneris(a)otenet.gr Tel : +30-210-6151886

1 0

[Users] Problem with dialog module
by Leonid Fainshtein 16 Mar '07

16 Mar '07

Hi, I see that when Openser 1.1 receives re-INVITE it removes parameter "did=" from header field Route before sending it to the second call party. Any idea? Thank you in advance, Leonid ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

2 1

[Users] AVPOPS: question about avp_db_query
by Papadopoulos Georgios 16 Mar '07

16 Mar '07

Hello, Is there a way to load an integer value into an integer avp using avp_db_query? For example if in table usr_preferences I have value='15' and type=1, then using avp_db_load I get an avp with an int value of 15. Then I can use this int avp to check whether it is greater or less than some value. I am trying to do the same thing with avp_db_query but the avp that I get is a string. thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

3 3

[Users] TLS between OPENSer and FreeRadius ?
by Bodin Bruno 16 Mar '07

16 Mar '07

Hello, Is it possible to use a TLS tunnel between OpenSer and Radius ? thank

1 0

[Serusers] make all error: avp_radius module compilation error - parse_uri.h:50
by tzieleniewski 16 Mar '07

16 Mar '07

Hi, I have just downloaded the cvs version of ser 2.1.0-dev1 and after I run #make group_include="radius" all (like usually) I got the following error: make[1]: Entering the directory `/home/ser/src/sip_router/modules/avp_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev1"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -I/usr/local/include -c avp_radius.c -o avp_radius.o avp_radius.c:244: error: static declaration of ‘uri_type_to_str’ follows non-static declaration ../../parser/parse_uri.h:50: error: previous declaration of ‘uri_type_to_str’ was here make[1]: *** [avp_radius.o] Error 1 make[1]: Opuszczenie katalogu `/home/ser/src/sip_router/modules/avp_radius' make: *** [modules] Error 1 Cheers Tomasz

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