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From: tomach <to...@dg...> - 2007-06-05 09:05:15
|
Hello!<br /><br />I am using atlsip activex.<br />I noticed that when I try to create connection, even call isnt established yet I recieve OnOutgoingCallConnected event. Also when connection is esablished I recieved it second time. Did anyone have the same kind of problem? <br /><br />Next thing is when i run softphone based on atlsip activex it does not send any events. But when I run it without visual studio it works fine. Was that bug reported before?<br /><br />Last thing I want to ask is: how can I know when the line is busy? I tried OnLogSIPMessage but then I have to parse it etc... I thought more about somethign like OnLineBusy event, does this kind of solution exist allready?<br /><br />Best Regards!<br /> |
|
From: Joegen E. B. <jb...@so...> - 2007-06-05 05:04:42
|
Hi, I took time to look at your log. One thing I have noticed is that your asterisk box keeps on retransmitting the 407 even after it has been ACKed several times. Would you be able to post your logs to the asterisk mailing list? From what I can guess, asterisk hates something that opensipstack sends in ACK. I might have missed something but I'm sure our ACK is 100% compliant. OpenSIPStack is also sending the Proxy-Authorization response header in response to the 407. Joegen web...@dz... wrote: > 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 >> > > > ------------------------------------------------------------------------- > 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 > |
|
From: Yacine A. <yac...@ms...> - 2007-06-04 18:52:58
|
Thanks very much Ilian,i was able to do all that, but since the update of C= NG i'm not able to make calls, i'm always getting 403 forbidden error while= i'm able to receive calls.Thanks> Date: Mon, 4 Jun 2007 20:40:31 +0800> Fr= om: ip...@so...> To: ope...@li...= > Subject: Re: [OpenSIPStack] ATLSIP compilation steps> > One more thing. I= think you have to rename the g729 library files to > va_g729a.h and va_g72= 9a.lib.> > - Ilian> > Ilian Jeri C. Pinzon wrote:> > Hi,> >> > Some codecs = are not enabled by default because of licensing issues. But > > you can ena= ble them by downloading the codec libraries into > > opensipstack/external/= codecs/ and then recompile.> >> > A G.723.1 codec can be downloaded from th= e MyPhone project at > > Sourceforge (http://myphone.sourceforge.net). A G.= 729 codec can be > > downloaded from the VoiceAge site > > (http://www.acel= p.net/openinit_g729.php) . Take note, these codecs are > > bound by each of= their own license restrictions so you will not be able > > to use them leg= ally in certain scenarios. Use these at your own risk.> >> > Regards,> > Il= ian> >> > web...@dz... wrote:> > > >> Hello,> >>> >> Thank you ve= ry much Joegen, the problem was that i was trying to compile it on a WIndow= s XP home edition computer, i succeed now on windows xp pro, i just noticed= that some codecs are not available now whene i compiled ossphone, do i hav= e to enable theme somewhere before starting compilation ?> >>> >> thanks> >= >> >>> >>> >> On Sat, 02 Jun 2007 09:49:21 +0800, "Joegen E. Baclor" <jbacl= or...@so...> wrote:> >>> >> > >> > >>> Hi,> >>> > >>> = > >> > >>> >> > >> > >>> Can you copy and paste the error from = the MSVC output window? What is> >>> > >>> > >> > >> > >>= > your OS version?> >>> > >>> > >> > >>> >> > >> > >>> Jo= egen> >>> > >>> > >> > >>> >> > >> > >>> webmaster@dzmeet= .com wrote:> >>> > >>> > >> > >> > >>>> Hello,> >>>> = > >>>> > >> > >>> >> > >> > >>>> I have the latest cvs open= sipstack and atlsip, i'm using microsoft> >>>> > >>>> > >> = > >> > >>> visual c++ 2005 (not express edition), i folowed the steps o= n how to> >>> > >>> > >> > >> > >>> compile opensipstack fr= om the wiki, using release build first but it> >>> > >>> > >> >= >> > >>> faild, is there any requirements before starting the compilat= ion process,> >>> > >>> > >> > >> > >>> can i know what are= the steps of compilation to have atlsip compiled.> >>> > >>> > >= > > >>> >> > >> > >>>> Thanks for everyone.> >>>> > >>>> = > >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > >> = > >>> ---------------------------------------------------------------------= ----> >>> > >>> > >> > >> > >>> This SF.net email is sponso= red 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 ge= t it now.> >>> > >>> > >> > >> > >>> http://sourceforge.net= /powerbar/db2/> >>> > >>> > >> > >> > >>> _________________= ______________________________> >>> > >>> > >> > >> > >>> o= pensipstack-devel mailing list> >>> > >>> > >> > >> > >>> o= pen...@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> >> cont= rol of your XML. No limits. Just data. Click to get it now.> >> http://sour= ceforge.net/powerbar/db2/> >> _____________________________________________= __> >> opensipstack-devel mailing list> >> ope...@li...urcef= orge.net> >> https://lists.sourceforge.net/lists/listinfo/opensipstack-deve= l> >>> >> > >> > >> >> > --------------------------------------------= -----------------------------> > This SF.net email is sponsored by DB2 Expr= ess> > 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...urce= forge.net> > https://lists.sourceforge.net/lists/listinfo/opensipstack-deve= l> >> >> > > > > --------------------------------------------------------= -----------------> This SF.net email is sponsored by DB2 Express> Download = DB2 Express C - the FREE version of DB2 express and take> control of your X= ML. No limits. Just data. Click to get it now.> http://sourceforge.net/powe= rbar/db2/> _______________________________________________> opensipstack-de= vel mailing list> ope...@li...> https://lists.s= ourceforge.net/lists/listinfo/opensipstack-devel _________________________________________________________________ Essayez Live.com et cr=E9ez l'Internet qui vous ressemble : infos, sports, = m=E9t=E9o et bien plus encore ! http://www.live.com/getstarted= |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-04 12:40:27
|
One more thing. I think you have to rename the g729 library files to va_g729a.h and va_g729a.lib. - Ilian Ilian Jeri C. Pinzon wrote: > Hi, > > Some codecs are not enabled by default because of licensing issues. But > you can enable them by downloading the codec libraries into > opensipstack/external/codecs/ and then recompile. > > A G.723.1 codec can be downloaded from the MyPhone project at > Sourceforge (http://myphone.sourceforge.net). A G.729 codec can be > downloaded from the VoiceAge site > (http://www.acelp.net/openinit_g729.php) . Take note, these codecs are > bound by each of their own license restrictions so you will not be able > to use them legally in certain scenarios. Use these at your own risk. > > Regards, > Ilian > > web...@dz... wrote: > >> Hello, >> >> Thank you very much Joegen, the problem was that i was trying to compile it on a WIndows XP home edition computer, i succeed now on windows xp pro, i just noticed that some codecs are not available now whene i compiled ossphone, do i have to enable theme somewhere before starting compilation ? >> >> thanks >> >> >> >> On Sat, 02 Jun 2007 09:49:21 +0800, "Joegen E. Baclor" <jb...@so...> wrote: >> >> >> >>> Hi, >>> >>> >> >> >> >> >>> Can you copy and paste the error from the MSVC output window? What is >>> >>> >> >> >>> your OS version? >>> >>> >> >> >> >> >>> Joegen >>> >>> >> >> >> >> >>> web...@dz... wrote: >>> >>> >> >> >>>> Hello, >>>> >>>> >> >> >> >> >>>> I have the latest cvs opensipstack and atlsip, i'm using microsoft >>>> >>>> >> >> >>> visual c++ 2005 (not express edition), i folowed the steps on how to >>> >>> >> >> >>> compile opensipstack from the wiki, using release build first but it >>> >>> >> >> >>> faild, is there any requirements before starting the compilation process, >>> >>> >> >> >>> can i know what are the steps of compilation to have atlsip compiled. >>> >>> >> >> >> >> >>>> Thanks for everyone. >>>> >>>> >> >> >> >> >> >> >> >> >> >> >> >> >>> ------------------------------------------------------------------------- >>> >>> >> >> >>> 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 > > > |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-04 12:37:24
|
Hi, Some codecs are not enabled by default because of licensing issues. But you can enable them by downloading the codec libraries into opensipstack/external/codecs/ and then recompile. A G.723.1 codec can be downloaded from the MyPhone project at Sourceforge (http://myphone.sourceforge.net). A G.729 codec can be downloaded from the VoiceAge site (http://www.acelp.net/openinit_g729.php) . Take note, these codecs are bound by each of their own license restrictions so you will not be able to use them legally in certain scenarios. Use these at your own risk. Regards, Ilian web...@dz... wrote: > Hello, > > Thank you very much Joegen, the problem was that i was trying to compile it on a WIndows XP home edition computer, i succeed now on windows xp pro, i just noticed that some codecs are not available now whene i compiled ossphone, do i have to enable theme somewhere before starting compilation ? > > thanks > > > > On Sat, 02 Jun 2007 09:49:21 +0800, "Joegen E. Baclor" <jb...@so...> wrote: > > >> Hi, >> > > > > >> Can you copy and paste the error from the MSVC output window? What is >> > > >> your OS version? >> > > > > >> Joegen >> > > > > >> web...@dz... wrote: >> > > >>> Hello, >>> > > > > >>> I have the latest cvs opensipstack and atlsip, i'm using microsoft >>> > > >> visual c++ 2005 (not express edition), i folowed the steps on how to >> > > >> compile opensipstack from the wiki, using release build first but it >> > > >> faild, is there any requirements before starting the compilation process, >> > > >> can i know what are the steps of compilation to have atlsip compiled. >> > > > > >>> Thanks for everyone. >>> > > > > > > > > > > > > >> ------------------------------------------------------------------------- >> > > >> 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 > > |
|
From: rnones <rn...@so...> - 2007-06-04 12:25:09
|
test <br /><br /><br /><br />post<br /><br /> |
|
From: tomach <to...@dg...> - 2007-06-04 12:11:54
|
hello!<br /><br />I have this problem:<br /><br />I used altsip activex for .net. Everything works fine, but I can not recieve events like: onringing, onconected, ondisconetced etc... the only event i recieve is OnLogSIPMessage. Even I create whole connection can hear my voice in phone etc...but no event is processe :(<br /><br />Does anyone have any idea?<br /> |
|
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 |
|
From: Joegen E. B. <jb...@so...> - 2007-06-03 03:28:57
|
Leo, I agree, the softphone should be able to choose the error code when rejecting calls. I will personally find time to patch this up within the next couple of days. I will let the list know. Joegen le...@da... wrote: > Hi all, > > I'm using the SoftphoneInterface class in my SIP client. I noticed > that if I reject a call manually, the client will send 403. The server > we're connecting to prefers us to send 480. Is there anyway to force > the response code? I've tried calling ClearCall directly with various > reasons (EndedByNoAnswer, EndedByLocalBusy and EndedByAnswerDenied), > but it always sends 403. Is there a mapping table that shows which > response code maps to which reason. > > Regards and TIA. > > Leo > > > ------------------------------------------------------------------------- > 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 > > > |
|
From: Joegen E. B. <jb...@so...> - 2007-06-03 03:27:11
|
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. 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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 > |
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From: <web...@dz...> - 2007-06-02 13:37:43
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Hello, Thank you very much Joegen, the problem was that i was trying to compile it on a WIndows XP home edition computer, i succeed now on windows xp pro, i just noticed that some codecs are not available now whene i compiled ossphone, do i have to enable theme somewhere before starting compilation ? thanks On Sat, 02 Jun 2007 09:49:21 +0800, "Joegen E. Baclor" <jb...@so...> wrote: > Hi, > > Can you copy and paste the error from the MSVC output window? What is > your OS version? > > Joegen > > web...@dz... wrote: >> Hello, >> >> I have the latest cvs opensipstack and atlsip, i'm using microsoft > visual c++ 2005 (not express edition), i folowed the steps on how to > compile opensipstack from the wiki, using release build first but it > faild, is there any requirements before starting the compilation process, > can i know what are the steps of compilation to have atlsip compiled. >> >> Thanks for everyone. >> >> >> > > > ------------------------------------------------------------------------- > 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 |
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From: Yacine A. <yac...@ms...> - 2007-06-02 13:36:00
|
Hi,Thanks a lot for all your efforts.i have succesfully compiled ATLSIP wit= h 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: opensipstack-devel@lis= ts.sourceforge.net> Subject: Re: [OpenSIPStack] Cpmfort Noise Support> > Hi= all,> > I have exposed the setting of silence detection mode and audio jit= ter > delay in ATLSIP and SoftPhoneInterface.> > Here are the methods:> > D= isableSilenceDetection()> - Disables silence detection. Disables CNG as wel= l.> > EnableFixedSilenceDetection( ULONG threshold )> - Enables fixed silen= ce 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=3D3 > as suggested by Whit in= another thread.> > EnableFixedSilenceDetectionEx( ULONG threshold , ULONG = signalDeadband, > ULONG silenceDeadband )> - An extended version of the pr= evious 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 a= daptivePeriod > milliseconds. However, its silence detection doesn't seem t= o be very > effective (at least in my machine). I'll look into this further= to see > what's wrong. This mode with adaptivePeriod=3D4800 is the default= mode for > ATLSIP.> > EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeri= od, 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 use= d, look in OpalSilenceDetector::ReceivedPacket().> > SetAudioJitterDelay( U= LONG minDelay, ULONG maxDelay )> - Sets audio jitter delay settings.> > > R= egards,> Ilian> > Ilian Jeri C. Pinzon wrote:> > Will prioritize this reque= st. 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 e= xposed this yet but we will soon. Please wait for updates > >>> in this lis= t.> >>>> >>> For the meantime, please refer to the attached email on how th= is 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 an= d 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 whic= h> >>>> 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 Aste= risk Server only G729 Work and not G729A> >>>> What's Wrong ?> >>>>> >>>>> = >>>>> >>>>> >>>> ----------------------------------------------------------= --------------> >>>> Avec Windows Live Spaces, publiez directement des mess= ages > >>>> =E9lectroniques sur votre blog ou ajoutez-y des photos, des bla= gues et > >>>> d'autres infos. C'est gratuit ! > >>>> <http://clk.atdmt.com= /MSN/go/msnnksac0030000001msn/direct/01/?href=3Dhttp://www.imagine-msn.com/= spaces> > >>>>> >>>> ------------------------------------------------------= ------------------> >>>>> >>>> --------------------------------------------= ----------------------------- > >>>>> >>>> This SF.net email is sponsored b= y DB2 Express> >>>> Download DB2 Express C - the FREE version of DB2 expres= s 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.sourcef= orge.net/lists/listinfo/opensipstack-devel> >>>> > >>>> -----------------= -------------------------------------------------------> >>>>> >>>> No viru= s found in this incoming message.> >>>> Checked by AVG Free Edition. Versio= n: 7.5.472 / Virus Database: > >>>> 269.8.1/822 - Release Date: 5/28/2007 1= 1:40 AM> >>>> > >>>> > >>>> > >>> -------------------------= -----------------------------------------------> >>>> >>> Subject:> >>> [Fw= d: 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" <ipinzon@soleg= ysystems.com>> >>>> >>> To:> >>> "Ilian Jeri C. Pinzon" <ipinzon@solegysyst= ems.com>> >>>> >>>> >>>> >>>> >>> -----------------------------------------= -------------------------------> >>>> >>> 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,> >>>> >>> Goo= d to hear you nailed it! Can't wait to see your contributions if > >>> yo= u 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 t= hreshold 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 s= ilence> >>>> detector into the ATLSIP library and send this in to the proje= ct in > >>>> the next> >>>> couple weeks...> >>>>> >>>>> >>>> Whit> >>>>> >= >>>> >>>>> >>>> -----Original Message-----> >>>> From: opensipstack-devel-b= ou...@li...> >>>> [mailto:opensipstack-devel-bounces@lists= .sourceforge.net] 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.> >>>>> >>>>> >>>>> >>>> OpalSil= enceDetector::Param param;> >>>> param.Mode =3D OpalSilenceDetector= ::NoSilenceDetection;> >>>> sfManager.SetSilenceDetectParams( param= s );> >>>>> >>>>> >>>>> >>>>> >>>> Hope that helps.> >>>>> >>>> de...@wh...= wrote:> >>>> > >>>> > >>>> > >>>>> Joegen,> >>>>>> >>>>> Th= anks for the reply. I've been trying different jitterbuffer > >>>>> setting= s 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 h= elping. 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 won= dering if it may have something to do with the Silence > >>>>> detection> >= >>>> > >>>>> > >>>>> > >>>> portion> >>>> > >>>> = > >>>> > >>>>> of Opal. I've noticed in the opal.log file that t= he Silence Threshold> >>>>> > >>>>> > >>>>> > >>>> cr= eeps> >>>> > >>>> > >>>> > >>>>> upwards the longer the pers= on 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 t= he code.> >>>>> I may be on the wrong track but I can't figure out this str= ange > >>>>> behavior.> >>>>> Any help/ideas/suggestions would be greatly a= ppreciated!> >>>>>> >>>>> Whit> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >= >>>>> >>>>> -----Original Message-----> >>>>> From: opensipstack-devel-boun= ce...@li...> >>>>> [mailto:opensipstack-devel-bounces@lists.s= ourceforge.net] On Behalf Of> >>>>> > >>>>> > >>>>> >= >>>> Joegen> >>>> > >>>> > >>>> > >>>>> E. Baclor> >>>>> Se= nt: Monday, April 09, 2007 5:19 AM> >>>>> To: ope...@li...ur= ceforge.net> >>>>> 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> >>>>>> > >>>>>> > >>>>>> > >>>> sa= mples> >>>> > >>>> > >>>> > >>>>>> 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 s= ystem 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 sayi= ng. 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 probl= ems.> >>>>>>> >>>>>> So my question is:> >>>>>>> >>>>>>> >>>>>> 1. Can I se= nd 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-buff= er or comfort noise settings? > >>>>>> Is this> >>>>>> > >>>>>> = > >>>>>> > >>>> done> >>>> > >>>> > >>>> > = >>>>>> > >>>>>> > >>>>>> > >>>>> SoftPhoneM= anager::SetAudioJitterDelay(). It is not yet exposed an the> >>>>> Active= X properties. Feel free to send in a patch if you get the chance> >>>>> to= expose it.> >>>>>> >>>>> > >>>>> > >>>>> > >>>>>> i= n the code itself?> >>>>>> 3. Maybe I'm on the wrong track and any suggesti= ons are welcome!> >>>>>>> >>>>>>> >>>>>> Look forward to working more with = everyone on this exciting project!> >>>>>>> >>>>>> Whit> >>>>>>> >>>>>>> >>= >>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> ----------------------= --------------------------------------------------- > >>>>>>> >>>>>> Take S= urveys. 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=3Djoin.php&p= =3Dsourceforge&CID=3DDEVDEV > >>>>>>> >>>>>> ______________________________= _________________> >>>>>> opensipstack-devel mailing list> >>>>>> opensipst= ack...@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 & bu= siness topics through brief surveys-and earn cash> >>>>> http://www.techsay= .com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3DDEVDEV > >>>>>> >>>>= > _______________________________________________> >>>>> opensipstack-devel= mailing list> >>>>> ope...@li...> >>>>> https:= //lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>>>>> >>>>>> >>= >>> -----------------------------------------------------------------------= -- > >>>>>> >>>>> Take Surveys. Earn Cash. 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From: <le...@da...> - 2007-06-02 09:30:40
|
Hi all, I'm using the SoftphoneInterface class in my SIP client. I noticed that if I reject a call manually, the client will send 403. The server we're connecting to prefers us to send 480. Is there anyway to force the response code? I've tried calling ClearCall directly with various reasons (EndedByNoAnswer, EndedByLocalBusy and EndedByAnswerDenied), but it always sends 403. Is there a mapping table that shows which response code maps to which reason. Regards and TIA. Leo |
|
From: Joegen E. B. <jb...@so...> - 2007-06-02 01:50:28
|
Hi, Can you copy and paste the error from the MSVC output window? What is your OS version? Joegen web...@dz... wrote: > Hello, > > I have the latest cvs opensipstack and atlsip, i'm using microsoft visual c++ 2005 (not express edition), i folowed the steps on how to compile opensipstack from the wiki, using release build first but it faild, is there any requirements before starting the compilation process, can i know what are the steps of compilation to have atlsip compiled. > > Thanks for everyone. > > > |
|
From: <web...@dz...> - 2007-06-01 17:18:08
|
Hello, I have the latest cvs opensipstack and atlsip, i'm using microsoft visual c++ 2005 (not express edition), i folowed the steps on how to compile opensipstack from the wiki, using release build first but it faild, is there any requirements before starting the compilation process, can i know what are the steps of compilation to have atlsip compiled. Thanks for everyone. |
|
From: Joegen E. B. <jb...@so...> - 2007-05-30 14:28:21
|
Super! Thanks Ilian. Ilian Jeri C. Pinzon wrote: > 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 > > |
|
From: Joegen E. B. <jb...@so...> - 2007-05-30 14:25:00
|
Kolneath SOMETH wrote: > > Hi Joegen, > > Yes, it refers to the admin data and also the configuration data ie. > OBSC mode, routing while add or remove...etc. > > BR, > Kolneath SOMETH Take a look at OSSAppConfig.cxx |
|
From: Kolneath S. <kol...@sa...> - 2007-05-30 14:07:54
|
Hi Joegen,
Yes, it refers to the admin data and also the configuration data ie. OBSC=20
mode, routing while add or remove...etc.
BR,
Kolneath SOMETH
"Joegen E. Baclor" <jb...@so...>
Envoy=E9 par : ope...@li...
30/05/2007 15:57
Veuillez r=E9pondre =E0 jbaclor; Veuillez r=E9pondre =E0 opensipstack-devel
Remis le : 30/05/2007 16:01
=20
Pour : jb...@so...
cc : ope...@li..., (ccc : Kolneath=20
SOMETH/DRD/SAGEM)
Objet : Re: [OpenSIPStack] SQLite Database for OSBC
Joegen E. Baclor wrote:
> Kolneath SOMETH wrote:
>>
>> Hi everyone,
>>
>> I wonder where/how the database files are stored (embedded)...and=20
>> which .cxx files concerns.
>>
>> Thanks and Best regards,
>> Kolneath SOMETH
>>
>
>
> By default, SQLite support is not enabled in OpenSIPStack. The main=20
> purpose of the SQLite classes is for registration recovery. This is=20
> to make sure that if ever OpenSBC is restarted from a crash, previous=20
> registrations are recovered. For more details see the following=20
methods:
>
> BOOL RegistrationDatabase::PrepareContactRecoveryDB()
> BOOL RegistrationDatabase::AddRegistrationRecovery()
>
>
> To enable the SQLite classes you need to set HAS=5FCPPSQLITE in=20
> ossbuildopts.h
>
> Joegen
>
>
Hold on. By any chance, are you referring to the data stored in the=20
HTTP admin? If this is what you are asking for, the data is being store=20
in the system registry in windows and in .ini files for linux. The=20
default location in linux is $(HOME)/.pwlib=5Fconfig (there is a dot (.) )
-------------------------------------------------------------------------
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control of your XML. No limits. Just data. Click to get it now.
http://sourceforge.net/powerbar/db2/
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
opensipstack-devel mailing list
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https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
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From: Joegen E. B. <jb...@so...> - 2007-05-30 14:00:55
|
Joegen E. Baclor wrote: > Kolneath SOMETH wrote: >> >> Hi everyone, >> >> I wonder where/how the database files are stored (embedded)...and >> which .cxx files concerns. >> >> Thanks and Best regards, >> Kolneath SOMETH >> > > > By default, SQLite support is not enabled in OpenSIPStack. The main > purpose of the SQLite classes is for registration recovery. This is > to make sure that if ever OpenSBC is restarted from a crash, previous > registrations are recovered. For more details see the following methods: > > BOOL RegistrationDatabase::PrepareContactRecoveryDB() > BOOL RegistrationDatabase::AddRegistrationRecovery() > > > To enable the SQLite classes you need to set HAS_CPPSQLITE in > ossbuildopts.h > > Joegen > > Hold on. By any chance, are you referring to the data stored in the HTTP admin? If this is what you are asking for, the data is being store in the system registry in windows and in .ini files for linux. The default location in linux is $(HOME)/.pwlib_config (there is a dot (.) ) |
|
From: Joegen E. B. <jb...@so...> - 2007-05-30 13:50:26
|
Kolneath SOMETH wrote:
>
> Hi everyone,
>
> I wonder where/how the database files are stored (embedded)...and
> which .cxx files concerns.
>
> Thanks and Best regards,
> Kolneath SOMETH
>
By default, SQLite support is not enabled in OpenSIPStack. The main
purpose of the SQLite classes is for registration recovery. This is to
make sure that if ever OpenSBC is restarted from a crash, previous
registrations are recovered. For more details see the following methods:
BOOL RegistrationDatabase::PrepareContactRecoveryDB()
BOOL RegistrationDatabase::AddRegistrationRecovery()
To enable the SQLite classes you need to set HAS_CPPSQLITE in ossbuildopts.h
Joegen
|
|
From: Kolneath S. <kol...@sa...> - 2007-05-30 13:39:15
|
Hi everyone, I wonder where/how the database files are stored (embedded)...and which=20 .cxx files concerns. Thanks and Best regards, Kolneath SOMETH " Ce courriel et les documents qui y sont attaches peuvent contenir des inf= ormations confidentielles. Si vous n'etes pas le destinataire escompte, me= rci d'en informer l'expediteur immediatement et de detruire ce courriel ai= nsi que tous les documents attaches de votre systeme informatique. Toute di= vulgation, distribution ou copie du present courriel et des documents attac= hes sans autorisation prealable de son emetteur est interdite."=20 " This e-mail and any attached documents may contain confidential or propri= etary information. If you are not the intended recipient, please advise the= sender immediately and delete this e-mail and all attached documents from = your computer system. Any unauthorised disclosure, distribution or copying = hereof is prohibited." |
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From: Joegen E. B. <jb...@so...> - 2007-05-30 13:32:17
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Yes. One of the major functionality of a Session Border Controller is NAT traversal and network bridging. OpenSBC does both pretty well and many more. Make sure you get the latest from CVS for optimum functionality. If you are a developer, you can ask development questions in the mailing list https://lists.sourceforge.net/lists/listinfo/opensipstack-devel<br /> |
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From: Blackeye1010 <bla...@gm...> - 2007-05-30 13:18:47
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Hello,<br /><br />I just founf this project... It seems to be quite interesting.<br /><br />I would just want to know, before i dig into it, if it is possible to:<br /><br />1.<br />use it for a replacement of CISCO IOS... would it perform the same functions as the "ip nat sip-alg" nat command on IOS ?<br />What it basically does is to replace all internall ip address in any of the SIP signalling and SDP descriptors with the public ip address... and back the other way for the awnsers<br /><br />2.<br />Get it to work at a central location with only public IP address on the ethernet interfaces. SBC would get the public IP in each request from src_ip_addr (as in openser) replacing all private non routable address in the SIP signalling and SDP with that src_ip_addr... <br />The other way around for replies also...<br /><br />Humm... this may not be well written...i hope you understand... |
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From: Joegen E. B. <jb...@so...> - 2007-05-30 12:44:46
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I'm not sure what you mean by Actualization. I would assume you can't
perform a CVS update? I have just tried downloading via anonymous CVS
and it worked for me
In C:\: "C:\Program Files (x86)\TortoiseCVS\cvs.exe" -q checkout
-P opensipstack
CVSROOT=:pserver:ano...@op...:/cvsroot/opensipstack
Empty password used - try 'cvs login' with a real password
U opensipstack/Makefile.in
U opensipstack/OpenSIPStack-7.10.sln
U opensipstack/OpenSIPStack-7.10.vcproj
U opensipstack/OpenSIPStack.sln
U opensipstack/OpenSIPStack.vcproj
U opensipstack/ReadMe.txt
U opensipstack/buildindex.exe
U opensipstack/buildindex.h
U opensipstack/config.guess
U opensipstack/config.sub
Make sure your firewall is not blocking CVS. I experienced a similar
situation once when we installed a new firewall software in the office.
Joegen
Counet Jérémy wrote:
>
> Hi,
>
> When can I actualize the OpenSIPStack directory? It’s not the first
> time that I try to actualize it but no way…
>
> Dans C:\Documents and Settings\JCOUNET\My Documents\Visual Studio
> 2005\Projects\opensipstack : "C:\Program Files\TortoiseCVS\cvs.exe" -q
> update -d -P
>
> CVSROOT=:pserver:ano...@op...:/cvsroot/opensipstack
>
> connect to opensipstack.cvs.sourceforge.net:2401 failed: No connection
> could be made because the target machine actively refused it.
>
> Erreur : échec de l'opération CVS
>
> Regards,
>
> Jeremy.
>
> ------------------------------------------------------------------------
>
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From: Joegen E. B. <jb...@so...> - 2007-05-30 12:38:52
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If your registrar is also the same entity as your outbound proxy, make sure you call SetProxyAddress() aside from SetSIPServer() to 10.16.212.40. Counet Jérémy wrote: > > Hi, > > I’ve a problem with AccountAddress. In OSSPhone, you build the TO > header with the SIPServerAddress. On my SIP server the INVITE is > Forbidden because the From header, is not the same at REGISTER and > INVITE. My SIP server is distant and not on the same computer. > > * In my application * , I’ve set AccountAddress with the local host > address. But when I do an INVITE, the stack build the TO header with > the AccounAddress instead of the SIPServerAddress. So in your case, > I’ve REGISTER with a wrong URI, in my case, the TO INVITE header is > not good. > > Can you change this plz or say me where can I fixe this problem. > > Thanks a lot, > > Jeremy. > > ----------------144:08:09.332---------------- > > *** LISTENER STARTED *** 10.16.213.20:5060 [*** DEFAULT LISTENER ***] > > ----------------144:08:37.442---------------- > > REGISTER sip:10.16.212.40:5060 SIP/2.0 > > From: 9555 <sip:9555@10.16.212.40 -> * 10.16.213.20 > *>;tag=a627f1f9cdf81810918589195ccf8571 > > To: sip:9555@10.16.212.40 -> * 10.16.213.20 * > > … > > ----------------144:08:37.615---------------- > > SIP/2.0 100 Trying > > … > > ----------------144:08:37.632---------------- > > SIP/2.0 200 OK > > … > > ----------------144:09:13.001---------------- > > SEND: XOR=0 644 Bytes to 10.16.212.40:5060:UDP (INVITE > sip:101@10.16.212.40 SIP/2.0) Interface Address=10.16.213.20 > > INVITE sip:101@10.16.212.40 -> * 10.16.213.20 * SIP/2.0 > > From: 9555 <sip:9555@10.16.213.20>;tag=845b27facdf81810918789195ccf8571 > > To: sip:101@10.16.212.40 > > … > > ----------------144:09:13.019---------------- > > SIP/2.0 100 Trying > > … > > ----------------144:09:13.030---------------- > > SIP/2.0 403 Forbidden > > … > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > 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 > |
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