Re: [OpenSIPStack] Cpmfort Noise Support
Brought to you by:
joegenbaclor
Menu
▾
▴
Re: [OpenSIPStack] Cpmfort Noise Support
|
From: <web...@dz...> - 2007-06-03 09:43:42
|
Hi, This is the log i'm getting -------------------------------------------------------------------------------------------------------------- ----------------16:09.322---------------- *** LISTENER STARTED *** 127.0.0.1:5060 ----------------16:09.513---------------- *** LISTENER STARTED *** 192.168.0.53:5060 [*** DEFAULT LISTENER ***] ----------------16:09.612---------------- SEND: enc=0 546 Bytes to 193.194.64.11:5060:UDP (REGISTER sip:193.194.64.11 SIP/2.0) Interface Address= REGISTER sip:193.194.64.11 SIP/2.0 From: 7000 <sip:7000@193.194.64.11>;tag=598b5f0dd6f8181099ef9f8b648defb9 To: sip:7000@193.194.64.11 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK74915f0dd6f8181099ef9f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 1 REGISTER Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060;transport=udp> User-Agent: OpenSIPStack-1.1.6-168 Expires: 3600 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:09.812---------------- RCV: enc=0 493 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) SIP/2.0 100 Trying From: 7000 <sip:7000@193.194.64.11>;tag=598b5f0dd6f8181099ef9f8b648defb9 To: sip:7000@193.194.64.11 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK74915f0dd6f8181099ef9f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 1 REGISTER Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9 Contact: <sip:7000@193.194.64.11> User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:09.835---------------- RCV: enc=0 553 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) SIP/2.0 401 Unauthorized From: 7000 <sip:7000@193.194.64.11>;tag=598b5f0dd6f8181099ef9f8b648defb9 To: sip:7000@193.194.64.11;tag=as186cc90c Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK74915f0dd6f8181099ef9f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 1 REGISTER Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9 User-Agent: Asterisk PBX WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="549f3188" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:09.899---------------- SEND: enc=0 708 Bytes to 193.194.64.11:5060:UDP (REGISTER sip:193.194.64.11 SIP/2.0) Interface Address= REGISTER sip:193.194.64.11 SIP/2.0 From: 7000 <sip:7000@193.194.64.11>;tag=51ff5f0dd6f8181099ef9f8b648defb9 To: sip:7000@193.194.64.11 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK51ff5f0dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 2 REGISTER Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060;transport=udp> User-Agent: OpenSIPStack-1.1.6-168 Expires: 3600 Max-Forwards: 10 Authorization: Digest username="7000", realm="asterisk", nonce="549f3188", uri="sip:193.194.64.11", response="ae2f9b140c8cd0c80f6f0522cf29364f", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:09.917---------------- RCV: enc=0 493 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) SIP/2.0 100 Trying From: 7000 <sip:7000@193.194.64.11>;tag=51ff5f0dd6f8181099ef9f8b648defb9 To: sip:7000@193.194.64.11 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK51ff5f0dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 2 REGISTER Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9 Contact: <sip:7000@193.194.64.11> User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:09.925---------------- RCV: enc=0 585 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: 7000 <sip:7000@193.194.64.11>;tag=51ff5f0dd6f8181099ef9f8b648defb9 To: sip:7000@193.194.64.11;tag=as186cc90c Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK51ff5f0dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 2 REGISTER Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9 Contact: <sip:7000@192.168.0.53:5060;transport=udp>;expires=3600 Date: Sun, 03 Jun 2007 09:31:56 GMT User-Agent: Asterisk PBX Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:22.831---------------- SEND: enc=0 754 Bytes to 193.194.64.11:5060:UDP (INVITE sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 INVITE sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 203 v=0 o=- 1180863134 1180863134 IN IP4 192.168.0.53 s=OSS RTP Session c=IN IP4 192.168.0.53 t=0 0 m=audio 5000 RTP/AVP 101 8 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 ----------------16:22.858---------------- RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:22.908---------------- SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:22.961---------------- SEND: enc=0 927 Bytes to 193.194.64.11:5060:UDP (INVITE sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 INVITE sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKdbed730dd6f8181099f19f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4712 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Proxy-Authorization: Digest username="7000", realm="asterisk", nonce="797c48aa", uri="sip:5000@193.194.64.11", response="98b951dc98b2ae41bfa445f6225af411", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 203 v=0 o=- 1180863134 1180863134 IN IP4 192.168.0.53 s=OSS RTP Session c=IN IP4 192.168.0.53 t=0 0 m=audio 5000 RTP/AVP 101 8 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 ----------------16:22.977---------------- RCV: enc=0 472 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 403 Forbidden) SIP/2.0 403 Forbidden From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKdbed730dd6f8181099f19f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4712 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:22.992---------------- SEND: enc=0 700 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKdbed730dd6f8181099f19f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4712 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Proxy-Authorization: Digest username="7000", realm="asterisk", nonce="797c48aa", uri="sip:5000@193.194.64.11", response="98b951dc98b2ae41bfa445f6225af411", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:23.858---------------- RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:23.888---------------- SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:24.859---------------- RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:24.889---------------- SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:26.860---------------- RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:26.893---------------- SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:30.859---------------- RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:30.895---------------- SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:34.860---------------- RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:34.893---------------- SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------16:38.862---------------- RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53 CSeq: 4711 INVITE Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 User-Agent: Asterisk PBX Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ----------------16:38.891---------------- SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53 ACK sip:5000@193.194.64.11 SIP/2.0 From: 7000 <sip:7000@192.168.0.53>;tag=c973730dd6f8181099f19f8b648defb9 To: sip:5000@193.194.64.11;tag=as318d4032 Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport CSeq: 4711 ACK Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9 Contact: "7000" <sip:7000@192.168.0.53:5060> User-Agent: OpenSIPStack-1.1.6-168 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 -------------------------------------------------------------------------------------------------------------------------------------------- Thanks On Sun, 03 Jun 2007 11:26:45 +0800, "Joegen E. Baclor" <jb...@so...> wrote: > Hi Yacine, > > Please send Ilian a level sip 5 log so he can determine the casue and > give you a fix. Thanks. > > Joegen > > > Yacine Auczone wrote: >> >> Hi, >> Thanks a lot for all your efforts. >> i have succesfully compiled ATLSIP with the new changes, but i have a >> little issue now. >> i'm not able to make calls since the updates, i'm getting a 403 >> Forbidden error code whene trying to make a call while i was able to >> make calls before. >> >> >> >> ------------------------------------------------------------------------ >> > Date: Wed, 30 May 2007 17:53:09 +0800 >> > From: ip...@so... >> > To: ope...@li... >> > Subject: Re: [OpenSIPStack] Cpmfort Noise Support >> > >> > Hi all, >> > >> > I have exposed the setting of silence detection mode and audio jitter >> > delay in ATLSIP and SoftPhoneInterface. >> > >> > Here are the methods: >> > >> > DisableSilenceDetection() >> > - Disables silence detection. Disables CNG as well. >> > >> > EnableFixedSilenceDetection( ULONG threshold ) >> > - Enables fixed silence detection. Any sound level below the threshold >> > is treated as silence (and CN is generated as a result). Don't use too >> > high threshold values or you'll only hear comfort noise. Try >> threshold=3 >> > as suggested by Whit in another thread. >> > >> > EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband, >> > ULONG silenceDeadband ) >> > - An extended version of the previous method. Don't tinker with this >> > unless you know what you're doing. For reference on how signalDeadband >> > and silenceDeadband are used, look in >> OpalSilenceDetector::ReceivedPacket(). >> > >> > EnableAdaptiveSilenceDetection( ULONG adaptivePeriod ) >> > - Enables an adaptive silence detection. Supposedly this enables the >> > threshold to *adapt* to the current sound level every adaptivePeriod >> > milliseconds. However, its silence detection doesn't seem to be very >> > effective (at least in my machine). I'll look into this further to see >> > what's wrong. This mode with adaptivePeriod=4800 is the default mode >> for >> > ATLSIP. >> > >> > EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG >> > signalDeadband, ULONG silenceDeadband ) >> > - An extended version of the previous method. Don't tinker with this >> > unless you know what you're doing. For reference on how signalDeadband >> > and silenceDeadband are used, look in >> OpalSilenceDetector::ReceivedPacket(). >> > >> > SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay ) >> > - Sets audio jitter delay settings. >> > >> > >> > Regards, >> > Ilian >> > >> > Ilian Jeri C. Pinzon wrote: >> > > Will prioritize this request. This should be available by tomorrow >> or on >> > > early Thursday tops. >> > > >> > > Regards, >> > > Ilian >> > > >> > > Joegen E. Baclor wrote: >> > > >> > >> Hi Ilian, >> > >> >> > >> Can you provide an ETC for exposing Jitter and Silent Detection >> params >> > >> in ATLSIP? Seems like a popular request. >> > >> >> > >> Joegen >> > >> >> > >> >> > >> Ilian Jeri C. Pinzon wrote: >> > >> >> > >> >> > >>> Hi, >> > >>> >> > >>> We haven't exposed this yet but we will soon. Please wait for >> updates >> > >>> in this list. >> > >>> >> > >>> For the meantime, please refer to the attached email on how this >> can >> > >>> be done. >> > >>> >> > >>> Thanks. >> > >>> >> > >>> Regards, >> > >>> Ilian >> > >>> >> > >>> Yacine Auczone wrote: >> > >>> >> > >>> >> > >>>> Hi All, >> > >>>> First, Thanks a lot for all the great job you are doing for >> > >>>> OpenSipStack and AtlSIP >> > >>>> I'm doing some devlopement test with the Softphone ActiveX, the >> > >>>> quality is very good and no bugs detected, the only thing is >> that the >> > >>>> softphone is doing by default some VAD and it is not >> transmiting the >> > >>>> silence, so there is no Comfort Noise generation sent whene the >> > >>>> calling party stop talking. i heard about a new ActiveX version >> which >> > >>>> will be available and gives the option to enable or disable >> CNG, is it >> > >>>> ready? if yes can i have it please? >> > >>>> Other Thing, on my Asterisk Server only G729 Work and not G729A >> > >>>> What's Wrong ? >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> ------------------------------------------------------------------------ >> > >>>> Avec Windows Live Spaces, publiez directement des messages >> > >>>> électroniques sur votre blog ou ajoutez-y des photos, des >> blagues et >> > >>>> d'autres infos. C'est gratuit ! >> > >>>> >> > <http://clk.atdmt.com/MSN/go/msnnksac0030000001msn/direct/01/?href=http://www.imagine-msn.com/spaces> > >> >> > >>>> >> > >>>> >> ------------------------------------------------------------------------ >> > >>>> >> > >>>> >> > ------------------------------------------------------------------------- >> > >>>> >> > >>>> This SF.net email is sponsored by DB2 Express >> > >>>> Download DB2 Express C - the FREE version of DB2 express and take >> > >>>> control of your XML. No limits. Just data. Click to get it now. >> > >>>> http://sourceforge.net/powerbar/db2/ >> > >>>> >> ------------------------------------------------------------------------ >> > >>>> >> > >>>> _______________________________________________ >> > >>>> opensipstack-devel mailing list >> > >>>> ope...@li... >> > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >>>> >> > >>>> >> ------------------------------------------------------------------------ >> > >>>> >> > >>>> No virus found in this incoming message. >> > >>>> Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: >> > >>>> 269.8.1/822 - Release Date: 5/28/2007 11:40 AM >> > >>>> >> > >>>> >> > >>>> >> > >>> >> ------------------------------------------------------------------------ >> > >>> >> > >>> Subject: >> > >>> [Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort > Noise] >> > >>> From: >> > >>> "Joegen E. Baclor" <joe...@gm...> >> > >>> Date: >> > >>> Tue, 29 May 2007 18:26:21 +0800 >> > >>> To: >> > >>> "Ilian Jeri C. Pinzon" <ip...@so...> >> > >>> >> > >>> To: >> > >>> "Ilian Jeri C. Pinzon" <ip...@so...> >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> ------------------------------------------------------------------------ >> > >>> >> > >>> Subject: >> > >>> Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise >> > >>> From: >> > >>> "Joegen E. Baclor" <joe...@gm...> >> > >>> Date: >> > >>> Thu, 12 Apr 2007 16:40:04 +0800 >> > >>> To: >> > >>> ope...@li... >> > >>> >> > >>> To: >> > >>> ope...@li... >> > >>> >> > >>> >> > >>> Whit, >> > >>> >> > >>> Good to hear you nailed it! Can't wait to see your contributions > if >> > >>> you get the chance to expose the other setters/accessors in > ATLSIP. >> > >>> >> > >>> >> > >>> >> > >>> Whit Thiele wrote: >> > >>> >> > >>> >> > >>>> Joegen, >> > >>>> >> > >>>> Thanks for the help. I thought I'd send the list an update on > what >> > >>>> solved my >> > >>>> problem. I changed the Silence Detector to Fixed with a >> threshold of >> > >>>> 3. This >> > >>>> eliminated all the problems! It seems that the adaptive silence >> > >>>> detector was >> > >>>> constantly incrementing and started affecting things about >> 10-15 seconds >> > >>>> into a conversation! >> > >>>> >> > >>>> I'll probably put in the ability to change the jitterbuffer and >> silence >> > >>>> detector into the ATLSIP library and send this in to the >> project in >> > >>>> the next >> > >>>> couple weeks... >> > >>>> >> > >>>> >> > >>>> Whit >> > >>>> >> > >>>> >> > >>>> >> > >>>> -----Original Message----- >> > >>>> From: ope...@li... >> > >>>> [mailto:ope...@li...] On >> Behalf Of >> > >>>> Joegen E. Baclor >> > >>>> Sent: Tuesday, April 10, 2007 9:36 PM >> > >>>> To: ope...@li... >> > >>>> Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort >> Noise >> > >>>> >> > >>>> Whit, >> > >>>> >> > >>>> It is probably best to ask this question to >> > >>>> ope...@li.... However, here's how to set >> > >>>> the silence detection in code. >> > >>>> >> > >>>> >> > >>>> >> > >>>> OpalSilenceDetector::Param param; >> > >>>> param.Mode = OpalSilenceDetector::NoSilenceDetection; >> > >>>> sfManager.SetSilenceDetectParams( params ); >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> Hope that helps. >> > >>>> >> > >>>> de...@wh... wrote: >> > >>>> >> > >>>> >> > >>>> >> > >>>>> Joegen, >> > >>>>> >> > >>>>> Thanks for the reply. I've been trying different jitterbuffer >> > >>>>> settings as >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> well >> > >>>> >> > >>>> >> > >>>> >> > >>>>> as changing the number soundChannelBuffers to a number of >> different >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> settings >> > >>>> >> > >>>> >> > >>>> >> > >>>>> which I came across in some online >> > >>>>> Opal documentation ( >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> >> > http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html > >> >> > >>>> >> > >>>> ) >> > >>>> >> > >>>> >> > >>>> >> > >>>>> I've tried setting the jitter buffer to minimums 25 through to >> 500 >> > >>>>> and the >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> depth >> > >>>> >> > >>>> >> > >>>> >> > >>>>> to as high as 15 but nothing is helping. As I described before, > I >> > >>>>> can get >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> about >> > >>>> >> > >>>> >> > >>>> >> > >>>>> 10-15 consecutive seconds of decent voice quality and then it >> gets very >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> choppy. >> > >>>> >> > >>>> >> > >>>>> Is anyone else experiencing this? >> > >>>>> >> > >>>>> I am wondering if it may have something to do with the Silence >> > >>>>> detection >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> portion >> > >>>> >> > >>>> >> > >>>> >> > >>>>> of Opal. I've noticed in the opal.log file that the Silence >> Threshold >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> creeps >> > >>>> >> > >>>> >> > >>>> >> > >>>>> upwards the longer the person talks. Is there a way to disable >> the >> > >>>>> silence >> > >>>>> detector? I could see that there are several Modes (Fixed, >> Adaptive, >> > >>>>> etc) >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> for >> > >>>> >> > >>>> >> > >>>> >> > >>>>> it but I can't figure out where this is initialized in the code. >> > >>>>> I may be on the wrong track but I can't figure out this strange >> > >>>>> behavior. >> > >>>>> Any help/ideas/suggestions would be greatly appreciated! >> > >>>>> >> > >>>>> Whit >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> -----Original Message----- >> > >>>>> From: ope...@li... >> > >>>>> [mailto:ope...@li...] On >> Behalf Of >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> Joegen >> > >>>> >> > >>>> >> > >>>> >> > >>>>> E. Baclor >> > >>>>> Sent: Monday, April 09, 2007 5:19 AM >> > >>>>> To: ope...@li... >> > >>>>> Subject: Re: [OpenSIPStack] Audio problems - Jitter and >> Comfort Noise >> > >>>>> >> > >>>>> de...@wh... wrote: >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>>> Members, >> > >>>>>> >> > >>>>>> I'm doing some testing with the ATLSIP and opensipstack >> libraries >> > >>>>>> and so >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> far >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> with pretty good success. I have written a softphone in C# >> using the >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> samples >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> provided, however I have a strange issue which I think is >> related to >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> jitter >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> and/or comfort noise: >> > >>>>>> >> > >>>>>> Setup: >> > >>>>>> C# Softphone ----> Asterisk ---> PRI -----> Telco >> > >>>>>> >> > >>>>>> Once I make a call, the system works fine except if the person >> > >>>>>> using the >> > >>>>>> softphone talks for more then about 10-15 seconds (in a row >> without >> > >>>>>> being >> > >>>>>> interupted). Then, the audio starts to break up and the >> person on the >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> telco >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> side can't make out what they are saying. Sometimes this >> situation is >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> reversed >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> and the person on the softphone can't make out the person on >> the telco >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> side. >> > >>>> >> > >>>> >> > >>>>>> By the way, there aren't any problems with the telco or > asterisk >> > >>>>>> setup as >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> I >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>> have >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>>> SIP hardphones using the system with no problems. >> > >>>>>> >> > >>>>>> So my question is: >> > >>>>>> >> > >>>>>> >> > >>>>>> 1. Can I send confort noise during silence breaks? >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>> CNG is a codec functionality and is not manually generated by > the >> > >>>>> stack. >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>>> 2. Where can I tweak the jitter-buffer or comfort noise >> settings? >> > >>>>>> Is this >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> done >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>> SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed >> an the >> > >>>>> ActiveX properties. Feel free to send in a patch if you get >> the chance >> > >>>>> to expose it. >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>>> in the code itself? >> > >>>>>> 3. Maybe I'm on the wrong track and any suggestions are > welcome! >> > >>>>>> >> > >>>>>> >> > >>>>>> Look forward to working more with everyone on this exciting >> project! >> > >>>>>> >> > >>>>>> Whit >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > ------------------------------------------------------------------------- >> > >>>>>> >> > >>>>>> Take Surveys. Earn Cash. Influence the Future of IT >> > >>>>>> Join SourceForge.net's Techsay panel and you'll get the >> chance to >> > >>>>>> share >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>> your >> > >>>> >> > >>>> >> > >>>> >> > >>>>>> opinions on IT & business topics through brief surveys-and >> earn cash >> > >>>>>> >> > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> > >>>>>> >> > >>>>>> _______________________________________________ >> > >>>>>> opensipstack-devel mailing list >> > >>>>>> ope...@li... >> > >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>>> >> > >>>>> >> > ------------------------------------------------------------------------- >> > >>>>> >> > >>>>> Take Surveys. Earn Cash. Influence the Future of IT >> > >>>>> Join SourceForge.net's Techsay panel and you'll get the chance >> to share >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> your >> > >>>> >> > >>>> >> > >>>> >> > >>>>> opinions on IT & business topics through brief surveys-and >> earn cash >> > >>>>> >> > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> > >>>>> >> > >>>>> _______________________________________________ >> > >>>>> opensipstack-devel mailing list >> > >>>>> ope...@li... >> > >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >>>>> >> > >>>>> >> > >>>>> >> > ------------------------------------------------------------------------- >> > >>>>> >> > >>>>> Take Surveys. Earn Cash. Influence the Future of IT >> > >>>>> Join SourceForge.net's Techsay panel and you'll get the chance >> to share >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> your >> > >>>> >> > >>>> >> > >>>> >> > >>>>> opinions on IT & business topics through brief surveys-and >> earn cash >> > >>>>> >> > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> > >>>>> >> > >>>>> _______________________________________________ >> > >>>>> opensipstack-devel mailing list >> > >>>>> ope...@li... >> > >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>>> >> > >>>> >> > ------------------------------------------------------------------------- >> > >>>> >> > >>>> Take Surveys. Earn Cash. Influence the Future of IT >> > >>>> Join SourceForge.net's Techsay panel and you'll get the chance to >> > >>>> share your >> > >>>> opinions on IT & business topics through brief surveys-and earn >> cash >> > >>>> >> > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> > >>>> >> > >>>> _______________________________________________ >> > >>>> opensipstack-devel mailing list >> > >>>> ope...@li... >> > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > ------------------------------------------------------------------------- >> > >>>> >> > >>>> Take Surveys. Earn Cash. Influence the Future of IT >> > >>>> Join SourceForge.net's Techsay panel and you'll get the chance to >> > >>>> share your >> > >>>> opinions on IT & business topics through brief surveys-and earn >> cash >> > >>>> >> > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> > >>>> >> > >>>> _______________________________________________ >> > >>>> opensipstack-devel mailing list >> > >>>> ope...@li... >> > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>>> >> > >>> >> ------------------------------------------------------------------------ >> > >>> >> > >>> No virus found in this incoming message. >> > >>> Checked by AVG Free Edition. >> > >>> Version: 7.5.472 / Virus Database: 269.8.1/822 - Release Date: >> 5/28/2007 11:40 AM >> > >>> >> > >>> >> ------------------------------------------------------------------------ >> > >>> >> > >>> >> > ------------------------------------------------------------------------- >> > >>> This SF.net email is sponsored by DB2 Express >> > >>> Download DB2 Express C - the FREE version of DB2 express and take >> > >>> control of your XML. No limits. Just data. Click to get it now. >> > >>> http://sourceforge.net/powerbar/db2/ >> > >>> >> ------------------------------------------------------------------------ >> > >>> >> > >>> _______________________________________________ >> > >>> opensipstack-devel mailing list >> > >>> ope...@li... >> > >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >>> >> > >>> >> > >>> >> > >> >> > ------------------------------------------------------------------------- >> > >> This SF.net email is sponsored by DB2 Express >> > >> Download DB2 Express C - the FREE version of DB2 express and take >> > >> control of your XML. No limits. Just data. Click to get it now. >> > >> http://sourceforge.net/powerbar/db2/ >> > >> _______________________________________________ >> > >> opensipstack-devel mailing list >> > >> ope...@li... >> > >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > >> >> > >> >> > >> >> > >> >> > > >> > > >> > > >> > ------------------------------------------------------------------------- >> > > This SF.net email is sponsored by DB2 Express >> > > Download DB2 Express C - the FREE version of DB2 express and take >> > > control of your XML. No limits. Just data. Click to get it now. >> > > http://sourceforge.net/powerbar/db2/ >> > > _______________________________________________ >> > > opensipstack-devel mailing list >> > > ope...@li... >> > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > >> > > >> > > >> > >> > >> > >> > ------------------------------------------------------------------------- >> > This SF.net email is sponsored by DB2 Express >> > Download DB2 Express C - the FREE version of DB2 express and take >> > control of your XML. No limits. Just data. Click to get it now. >> > http://sourceforge.net/powerbar/db2/ >> > _______________________________________________ >> > opensipstack-devel mailing list >> > ope...@li... >> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> ------------------------------------------------------------------------ >> Soyez parmi les premiers à essayer Windows Live Mail. Windows Live >> Mail. >> > <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d> > >> >> ------------------------------------------------------------------------ >> >> > ------------------------------------------------------------------------- >> This SF.net email is sponsored by DB2 Express >> Download DB2 Express C - the FREE version of DB2 express and take >> control of your XML. No limits. Just data. Click to get it now. >> http://sourceforge.net/powerbar/db2/ >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > control of your XML. No limits. Just data. Click to get it now. > http://sourceforge.net/powerbar/db2/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
