opensipstack-devel Mailing List for OpenSIPStack (Page 64)
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From: Joegen E. B. <jb...@so...> - 2007-05-14 07:37:19
|
Hi Cedric, I have committed code in CVS for DNS/SRV failover. I would appreciate it if you can test this functionality and give me feedback of your results. Joegen |
From: Joegen E. B. <joe...@gm...> - 2007-05-14 02:57:49
|
Message Waiting Indicator. Presence, never the less, is also not yet supported in opensbc. This should not affect calls from working so you may ignore this safely. |
From: Joegen E. B. <joe...@gm...> - 2007-05-14 02:51:47
|
You need to download opensipstack using a CVS client. See http://sourceforge.net/cvs/?group_id=156710 for instructions. modules are opensipstack, atlsip and opensbc bondan wrote: > Dear Joegen, > > I visit http://www.opensipstack.org/, and the current version of > OpenSIPStack is 1.1.5 version. Also, the current version of ATLSIP is 1.0.2 > version. Would you inform to me where I have to download the newest version > of OpenSIPStack and ALTSIP? > > > Many thanks, > Bondan Wisnuwardhana > > > Send an Instant Message > > -----Original Message----- > From: Joegen E. Baclor [mailto:jb...@so...] > Sent: Friday, May 11, 2007 11:38 AM > To: jb...@so...; ope...@li... > Cc: bondan > Subject: Re: [OpenSIPStack] FW: compiled OSSPhone could not use to make call > > Bondan, > > One more thing. Looking at your logs, it seems you are using an old > copy of OpenSIPStack? Please download the latest CVS head. Current > version is > > OpenSIPStack-1.1.6-133 > > > Joegen > > Joegen E. Baclor wrote: > >> Hi Bondan, >> >> I am not able to replicate your problem. I just recompiled OSSPhone >> from CVS and it's able to autheticate calls against OpenSBC terminating >> to an asterisk box. See logs below. >> >> BTW, is this modified code you are working with? >> >> ----------------3:44:55.989---------------- >> *** LISTENER STARTED *** 127.0.0.1:5060 >> ----------------3:44:56.022---------------- >> *** LISTENER STARTED *** 192.168.0.161:5060 [*** DEFAULT LISTENER ***] >> ----------------3:44:56.028---------------- >> *** LISTENER STARTED *** 192.168.6.1:5060 >> ----------------3:44:56.031---------------- >> *** LISTENER STARTED *** 192.168.15.1:5060 >> ----------------3:46:44.147---------------- >> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP >> (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: <sip:186...@i2...>;tag=1359945326143 >> To: <sip:63.116.254.88:12000> >> Via: SIP/2.0/UDP >> >> > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ee5500004e3500000099;rport=6 > 2787;received=125.60.243.66 > >> CSeq: 64 OPTIONS >> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 >> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY >> Content-Length: 0 >> >> ----------------3:51:05.733---------------- >> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP >> (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: <sip:186...@i2...>;tag=1386096812568 >> To: <sip:63.116.254.88:12000> >> Via: SIP/2.0/UDP >> >> > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ef5b000027b10000009a;rport=6 > 2787;received=125.60.243.66 > >> CSeq: 65 OPTIONS >> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 >> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY >> Content-Length: 0 >> >> ----------------3:54:05.787---------------- >> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP >> (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: <sip:186...@i2...>;tag=1404096821517 >> To: <sip:63.116.254.88:12000> >> Via: SIP/2.0/UDP >> >> > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f00f000014b30000009b;rport=6 > 2787;received=125.60.243.66 > >> CSeq: 66 OPTIONS >> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 >> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY >> Content-Length: 0 >> >> ----------------3:56:49.262---------------- >> SEND: enc=0 507 Bytes to 192.168.6.1:5061:UDP (REGISTER >> sip:192.168.6.1:5061 SIP/2.0) Interface Address= >> REGISTER sip:192.168.6.1:5061 SIP/2.0 >> From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc >> To: sip:192.168.6.1 >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3825;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas > -addr=192.168.6.1;rport > >> CSeq: 1 REGISTER >> Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc >> Contact: <sip:192.168.6.1:5060;transport=udp> >> User-Agent: OpenSIPStack-1.1.6-133 >> Expires: 3600 >> Max-Forwards: 10 >> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS >> Content-Length: 0 >> >> ----------------3:56:49.320---------------- >> RCV: enc=0 424 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP >> (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc >> To: sip:192.168.6.1 >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3185;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas > -addr=192.168.6.1;rport=5060;received=192.168.6.1 > >> CSeq: 1 REGISTER >> Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc >> Contact: <sip:192.168.6.1:5060;transport=udp> >> Server: OpenSIPStack-1.1.6-133 >> Expires: 3600 >> Content-Length: 0 >> >> ----------------3:57:00.206---------------- >> SEND: enc=0 762 Bytes to 192.168.6.1:5061:UDP (INVITE >> sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 >> INVITE sip:613@192.168.6.1 SIP/2.0 >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: sip:613@192.168.6.1 >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas > -addr=192.168.6.1;rport > >> CSeq: 4711 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Contact: "1111" <sip:1111@192.168.6.1:5060> >> User-Agent: OpenSIPStack-1.1.6-133 >> Max-Forwards: 10 >> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS >> Content-Type: application/sdp >> Content-Length: 228 >> >> v=0 >> o=- 1178857661 1178857661 IN IP4 192.168.6.1 >> s=OSS RTP Session >> c=IN IP4 192.168.6.1 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 3 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:3 GSM/8000 >> ----------------3:57:00.222---------------- >> RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP >> (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: sip:613@192.168.6.1 >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas > -addr=192.168.6.1 > >> CSeq: 4711 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Content-Length: 0 >> >> ----------------3:57:00.248---------------- >> RCV: enc=0 557 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP >> (SIP/2.0 407 Proxy Authentication Required) >> SIP/2.0 407 Proxy Authentication Required >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas > -addr=192.168.6.1;rport=5060;received=192.168.6.1 > >> CSeq: 4711 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Proxy-Authenticate: Digest realm=192.168.6.1, >> nonce="92a98f3ef419bf98501149af46ee7604", >> opaque="e430240b53dfd47acdf7cbda42cdb64c", algorithm=MD5 >> Content-Length: 0 >> >> ----------------3:57:00.252---------------- >> SEND: enc=0 535 Bytes to 192.168.6.1:5061:UDP (ACK sip:613@192.168.6.1 >> SIP/2.0) Interface Address=192.168.6.1 >> ACK sip:613@192.168.6.1 SIP/2.0 >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas > -addr=192.168.6.1;rport > >> CSeq: 4711 ACK >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Contact: "1111" <sip:1111@192.168.6.1:5060> >> User-Agent: OpenSIPStack-1.1.6-133 >> Max-Forwards: 10 >> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS >> Content-Length: 0 >> >> ----------------3:57:00.258---------------- >> SEND: enc=0 957 Bytes to 192.168.6.1:5061:UDP (INVITE >> sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 >> INVITE sip:613@192.168.6.1 SIP/2.0 >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: sip:613@192.168.6.1 >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas > -addr=192.168.6.1;rport > >> CSeq: 4712 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Contact: "1111" <sip:1111@192.168.6.1:5060> >> User-Agent: OpenSIPStack-1.1.6-133 >> Max-Forwards: 10 >> Proxy-Authorization: Digest username="1111", realm="192.168.6.1", >> nonce="92a98f3ef419bf98501149af46ee7604", uri="sip:613@192.168.6.1", >> response="bf9a133f7e481849ae92fac139ad5525", algorithm=MD5 >> Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS >> Content-Type: application/sdp >> Content-Length: 228 >> >> v=0 >> o=- 1178857661 1178857661 IN IP4 192.168.6.1 >> s=OSS RTP Session >> c=IN IP4 192.168.6.1 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 3 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:3 GSM/8000 >> ----------------3:57:00.276---------------- >> RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP >> (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: sip:613@192.168.6.1 >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas > -addr=192.168.6.1 > >> CSeq: 4712 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Content-Length: 0 >> >> ----------------3:57:01.535---------------- >> RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP >> (SIP/2.0 180 Ringing) >> SIP/2.0 180 Ringing >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas > -addr=192.168.6.1;rport=5060;received=192.168.6.1 > >> CSeq: 4712 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Server: OpenSIPStack-1.1.6-133 >> Content-Length: 0 >> >> ----------------3:57:01.652---------------- >> RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP >> (SIP/2.0 180 Ringing) >> SIP/2.0 180 Ringing >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas > -addr=192.168.6.1;rport=5060;received=192.168.6.1 > >> CSeq: 4712 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Server: OpenSIPStack-1.1.6-133 >> Content-Length: 0 >> >> ----------------3:57:05.840---------------- >> RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP >> (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: <sip:186...@i2...>;tag=1422096824326 >> To: <sip:63.116.254.88:12000> >> Via: SIP/2.0/UDP >> >> > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f0c3000002e90000009c;rport=6 > 2787;received=125.60.243.66 > >> CSeq: 67 OPTIONS >> Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 >> Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY >> Content-Length: 0 >> >> ----------------3:57:08.535---------------- >> RCV: enc=0 764 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP >> (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas > -addr=192.168.6.1;rport=5060;received=192.168.6.1 > >> CSeq: 4712 INVITE >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Contact: <sip:192.168.6.1:5061> >> Record-Route: <sip:192.168.6.1:5061;lr> >> Server: OpenSIPStack-1.1.6-133 >> Content-Type: application/sdp >> Content-Length: 241 >> >> v=0 >> o=root 29469 29469 IN IP4 69.90.168.13 >> s=session >> c=IN IP4 69.90.168.13 >> t=0 0 >> m=audio 13674 RTP/AVP 3 97 101 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> ----------------3:57:08.782---------------- >> SEND: enc=0 439 Bytes to 192.168.6.1:5061:UDP (ACK sip:192.168.6.1:5061 >> SIP/2.0) Interface Address=192.168.6.1 >> ACK sip:192.168.6.1:5061 SIP/2.0 >> From: <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc >> To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df >> Via: SIP/2.0/UDP >> >> > 192.168.6.1:5060;branch=z9hG4bKdbb6f45ba7f818109c8e875f9c108cdc;uas-addr=192 > .168.6.1;rport > >> CSeq: 4712 ACK >> Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc >> Contact: "1111" <sip:1111@192.168.6.1:5060> >> Route: <sip:192.168.6.1:5061;lr> >> Content-Length: 0 >> >> >> >> >> bondan wrote: >> >> >>> Resend... >>> >>> >>> >>> Send an Instant Message >>> -----Original Message----- >>> From: bondan [mailto:bo...@ci...] >>> Sent: Friday, May 04, 2007 10:30 AM >>> To: 'jb...@so...'; >>> > 'ope...@li...' > >>> Cc: 'ki...@ci...'; 'A.Rafmawan'; 'indragilang'; 'bondan' >>> Subject: RE: [OpenSIPStack] compiled OSSPhone could not use to make call >>> >>> >>> Dear Joegen, >>> >>> This attachment is the log file and capture screen of OSSPhone. >>> Thank you for helping me... >>> >>> Best regards, >>> Bondan Wisnuwardhana >>> >>> >>> >>> Send an Instant Message >>> -----Original Message----- >>> From: ope...@li... >>> [mailto:ope...@li...] On Behalf Of >>> Joegen E. Baclor >>> Sent: Wednesday, May 02, 2007 8:45 PM >>> To: ope...@li... >>> Cc: ki...@ci...; 'A.Rafmawan'; 'indragilang' >>> Subject: Re: [OpenSIPStack] compiled OSSPhone could not use to make call >>> >>> bondan wrote: >>> >>> >>> >>>> Dear Joegen, >>>> >>>> I download OSSPhone source, and compile it into application. With the >>>> application, I could login into asterisk server and receive call too. >>>> But when I use the application to make a call, the error message >>>> "Status: 407 Proxy Authentication Required" is appear. >>>> >>>> I use codec setting : >>>> >>>> Low Bit Rate : iLBC-13k3 >>>> >>>> High Bit Rate : G.711-ALaw-64k >>>> >>>> When I use OSSPhone (based on Windows installer) with the same >>>> setting, the OSSPhone can use to make a call. >>>> >>>> Would you give advice for me how to make call from OSSPhone source? >>>> Many thank for your helping,,, >>>> >>>> Best regards, >>>> >>>> Bondan Wisnuwardhana >>>> >>>> >>>> >>>> >>> Please send me the ATLSIP logs off list so I can see what went wrong. >>> >>> >>> >>> >>>> ------------------------------------------------------------------------ >>>> >>>> >>>> >>> ------------------------------------------------------------------------- >>> 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 >>> >>> __________ NOD32 2235 (20070502) Information __________ >>> >>> This message was checked by NOD32 antivirus system. >>> http://www.eset.com >>> >>> >>> ------------------------------------------------------------------------ >>> >>> ----------------54:03.175---------------- >>> *** LISTENER STARTED *** 127.0.0.1:5060 >>> ----------------54:03.250---------------- >>> *** LISTENER STARTED *** 192.168.1.70:5060 [*** DEFAULT LISTENER ***] >>> ----------------54:04.340---------------- >>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 >>> > SIP/2.0) Interface Address= > >>> REGISTER sip:192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >>> Expires: 3600 >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ----------------54:04.909---------------- >>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 >>> > SIP/2.0) Interface Address= > >>> REGISTER sip:192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >>> Expires: 3600 >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ----------------54:05.899---------------- >>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 >>> > SIP/2.0) Interface Address= > >>> REGISTER sip:192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >>> Expires: 3600 >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ----------------54:07.908---------------- >>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 >>> > SIP/2.0) Interface Address= > >>> REGISTER sip:192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >>> Expires: 3600 >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ----------------54:09.453---------------- >>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE >>> > sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > >>> INVITE sip:102@192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >>> To: sip:102@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 4711 INVITE >>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060> >>> Max-Forwards: 70 >>> Content-Type: application/sdp >>> Content-Length: 231 >>> >>> v=0 >>> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >>> s=OSS RTP Session >>> c=IN IP4 192.168.1.70 >>> t=0 0 >>> m=audio 5000 RTP/AVP 101 106 8 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=rtpmap:106 iLBC/8000 >>> a=rtpmap:8 PCMA/8000 >>> >>> >>> ----------------54:10.007---------------- >>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE >>> > sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > >>> INVITE sip:102@192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >>> To: sip:102@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 4711 INVITE >>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060> >>> Max-Forwards: 70 >>> Content-Type: application/sdp >>> Content-Length: 231 >>> >>> v=0 >>> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >>> s=OSS RTP Session >>> c=IN IP4 192.168.1.70 >>> t=0 0 >>> m=audio 5000 RTP/AVP 101 106 8 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=rtpmap:106 iLBC/8000 >>> a=rtpmap:8 PCMA/8000 >>> >>> >>> ----------------54:11.004---------------- >>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE >>> > sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > >>> INVITE sip:102@192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >>> To: sip:102@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 4711 INVITE >>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060> >>> Max-Forwards: 70 >>> Content-Type: application/sdp >>> Content-Length: 231 >>> >>> v=0 >>> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >>> s=OSS RTP Session >>> c=IN IP4 192.168.1.70 >>> t=0 0 >>> m=audio 5000 RTP/AVP 101 106 8 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=rtpmap:106 iLBC/8000 >>> a=rtpmap:8 PCMA/8000 >>> >>> >>> ----------------54:11.924---------------- >>> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 >>> > SIP/2.0) Interface Address= > >>> REGISTER sip:192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >>> Expires: 3600 >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ----------------54:13.007---------------- >>> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE >>> > sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > >>> INVITE sip:102@192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >>> To: sip:102@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 4711 INVITE >>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060> >>> Max-Forwards: 70 >>> Content-Type: application/sdp >>> Content-Length: 231 >>> >>> v=0 >>> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >>> s=OSS RTP Session >>> c=IN IP4 192.168.1.70 >>> t=0 0 >>> m=audio 5000 RTP/AVP 101 106 8 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=rtpmap:106 iLBC/8000 >>> a=rtpmap:8 PCMA/8000 >>> >>> >>> ----------------54:15.573---------------- >>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 100 Trying) > >>> SIP/2.0 100 Trying >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.597---------------- >>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 401 Unauthorized) > >>> SIP/2.0 401 Unauthorized >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73;tag=as087cafd3 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >>> > nonce="5c7fc4df" > >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.620---------------- >>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 100 Trying) > >>> SIP/2.0 100 Trying >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.661---------------- >>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 401 Unauthorized) > >>> SIP/2.0 401 Unauthorized >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73;tag=as087cafd3 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >>> > nonce="5c7fc4df" > >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.734---------------- >>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 100 Trying) > >>> SIP/2.0 100 Trying >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.802---------------- >>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 401 Unauthorized) > >>> SIP/2.0 401 Unauthorized >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73;tag=as087cafd3 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >>> > nonce="5c7fc4df" > >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.827---------------- >>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 100 Trying) > >>> SIP/2.0 100 Trying >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.896---------------- >>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 401 Unauthorized) > >>> SIP/2.0 401 Unauthorized >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73;tag=as087cafd3 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >>> > nonce="5c7fc4df" > >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.961---------------- >>> RCV: XOR=0 586 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 407 Proxy Authentication Required) > >>> SIP/2.0 407 Proxy Authentication Required >>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >>> To: sip:102@192.168.1.73;tag=as63b5a09f >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 4711 INVITE >>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >>> Contact: <sip:102@192.168.1.73> >>> User-Agent: Asterisk PBX >>> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", >>> > nonce="49982f66" > >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:15.991---------------- >>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 100 Trying) > >>> SIP/2.0 100 Trying >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:16.053---------------- >>> SEND: XOR=0 409 Bytes to 192.168.1.73:5060:UDP (ACK sip:102@192.168.1.73 >>> > SIP/2.0) Interface Address=192.168.1.70 > >>> ACK sip:102@192.168.1.73 SIP/2.0 >>> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >>> To: sip:102@192.168.1.73;tag=as63b5a09f >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a > ddr=192.168.1.73;rport > >>> CSeq: 4711 ACK >>> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >>> Contact: "101" <sip:101@192.168.1.70:5060> >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ----------------54:16.164---------------- >>> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 401 Unauthorized) > >>> SIP/2.0 401 Unauthorized >>> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >>> To: sip:101@192.168.1.73;tag=as087cafd3 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 1 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >>> > nonce="5c7fc4df" > >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:16.321---------------- >>> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 100 Trying) > >>> SIP/2.0 100 Trying >>> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 >>> To: sip:101@192.168.1.73 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 2 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.73> >>> User-Agent: Asterisk PBX >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> ----------------54:16.353---------------- >>> RCV: XOR=0 564 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP >>> > (SIP/2.0 200 OK) > >>> SIP/2.0 200 OK >>> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 >>> To: sip:101@192.168.1.73;tag=as087cafd3 >>> Via: SIP/2.0/UDP >>> > 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a > ddr=192.168.1.73;received=192.168.1.70 > >>> CSeq: 2 REGISTER >>> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >>> Contact: <sip:101@192.168.1.70:5060;transport=udp>;expires=3600 >>> Date: Fri, 04 May 2007 02:42:07 GMT >>> User-Agent: Asterisk PBX >>> Expires: 3600 >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> >>> >>> ------------------------------------------------------------------------ >>> >>> >>> >> ------------------------------------------------------------------------- >> 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 >> >> >> >> > > > __________ NOD32 2256 (20070510) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > > > |
From: FischeFr <fo...@op...> - 2007-05-13 06:06:57
|
I have no idea what MWI is, but I'm subscribing for the event called presence.<br />Can you help me? It's really important.<br /> |
From: Eric H. <er...@so...> - 2007-05-11 23:32:56
|
It wont let me do that. It says I have an old version of GTK installed. Is there a way to remove the old GTK? I can't seem to find it as a previously installed program. Joegen E. Baclor wrote: > OpenSIPStack requires that PProcess is created first before any other > objects. If you are using it in an MFC project you may do this by > declaring PProcess as a member variable of CWinApp > > class COSSPhoneMFCApp : public CWinApp > { > public: > /**** OpenSIPStack Start ****/ > class PWLibProcess : public PProcess { > PCLASSINFO(PWLibProcess, PProcess); > void Main() { } > } pwlibProcess; > /**** OpenSIPStack End ****/ > > > > If you are using OpenSIPStack in a console application, make sure you > use PProcess::Main() instead of creating your own int main() method > > class YourProcess : public PProcess > { > public: > PCLASSINFO( YourProcess , PProcess ); > void Main() > { > /// your code here > } > }; > > PCREATE_PROCESS( YourProcess ); > > > The PCREATE_PROCESS creates the int main() for you. > > Joegen > > Counet Jérémy wrote: >> Hi, >> >> When I use the lib file who as generate by the opensipstack porject, I >> have this error… >> >> Catastrophic failure, PProcess::Current() = NULL!! >> >> Must I initiate something before creating a SoftPhoneInterface? >> >> Thx, >> >> Jeremy Counet. >> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> 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: Cedric L. <fo...@op...> - 2007-05-11 16:26:31
|
Hello (again :-) )<br /><br />In B2BUpperReg mode, does OpenSBC provide a f= ull support for DNS SRV queries as stated in RFC 3263 ? Let me explain my c= ase :<br /><br />I would like to have to registrar for the domaine domain.m= e.localhost. For that purpose, My DNS has got these records for the domain = me.localhost :<br /><br />domain &= nbsp; &nbs= p; NAPTR&= nbsp; 100 50 &q= uot;s" "SIP+D2U" &n= bsp; "" _sip._udp.domai= n.me.localhost.<br />_sip._udp.domain &n= bsp;  = ; SRV 100 = 1 5060 domain1<br />= &nb= sp; = &nb= sp; SRV 200 1 &n= bsp; 5060 domain2<br />domain1&nb= sp; = &nb= sp; A &nbs= p; <ip 1><br />domain2 = ; &n= bsp;  = ; A = <ip 2><br /><br />That say= s that the domain domain.me.localhost can be reached at domain1.me.loc= alhost with the higher priority (100) and in case of non response at domain= 2.me.localhost<br /><br />Using that record and an 'Upper Registration = Route' configured as [sip:*@domain.me.localhost*] sip:domain.me.localho= st:5060, everything works fine when domain1 is working. But when domain1 is= not working, openSBC should reroute its calls to domain2. It looks like it= does not. It still try to route the call to domain1.<br /><br />Am I right= ? Is is a bug ? IS it a feature not yet supported ? Please advise.<br /><b= r />Regards,<br />Cedric |
From: Cedric L. <fo...@op...> - 2007-05-11 15:04:37
|
It works now perfectly. Thank you for correcting this bug so quickly <img class="emoticon" src="http://www.opensourcesip.org:8080/jiveforums/images/emoticons/happy.gif" border="0" alt="" width="16" height="16" /><br /><br />Best regards,<br />Cedric |
From: Joegen E. B. <jb...@so...> - 2007-05-11 14:20:15
|
OpenSIPStack requires that PProcess is created first before any other objects. If you are using it in an MFC project you may do this by declaring PProcess as a member variable of CWinApp class COSSPhoneMFCApp : public CWinApp { public: /**** OpenSIPStack Start ****/ class PWLibProcess : public PProcess { PCLASSINFO(PWLibProcess, PProcess); void Main() { } } pwlibProcess; /**** OpenSIPStack End ****/ If you are using OpenSIPStack in a console application, make sure you use PProcess::Main() instead of creating your own int main() method class YourProcess : public PProcess { public: PCLASSINFO( YourProcess , PProcess ); void Main() { /// your code here } }; PCREATE_PROCESS( YourProcess ); The PCREATE_PROCESS creates the int main() for you. Joegen Counet Jérémy wrote: > > Hi, > > When I use the lib file who as generate by the opensipstack porject, I > have this error… > > Catastrophic failure, PProcess::Current() = NULL!! > > Must I initiate something before creating a SoftPhoneInterface? > > Thx, > > Jeremy Counet. > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > 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: <jer...@ze...> - 2007-05-11 13:32:07
|
Hi, =20 When I use the lib file who as generate by the opensipstack porject, I = have this error... =20 Catastrophic failure, PProcess::Current() =3D NULL!! =20 Must I initiate something before creating a SoftPhoneInterface? =20 Thx, Jeremy Counet. |
From: joegen <jb...@so...> - 2007-05-11 12:58:10
|
Congratualations! You just found a genuine bug. Fix is now in CVS. Thanks for the bug report.<br /> |
From: Cedric L. <fo...@op...> - 2007-05-11 08:55:25
|
Thank you for that answer. That is what I have done and it 'quite' = works. 'Quite' because I have encounter a strange behaviour :<br />= - OpenSBC behave as a B2BUA with Upper registration for domain 'd',= it consider 'd' as a trusted domain and add n 'opensbc route&#= 39; declared as [sip:*@d*] sip:<ip of portasip>:5060<br />- my sippho= ne register to domain 'd' and it is declared to use opens= bc as proxy : it works. Every REGISTER requests goes though openSBC and sip= phone get authenticated.<br />- Opensbc routes each call coming from Portas= ip to the sipphone perfectly. Media proxy works fine. Great :-)<br />- for = calls coming from my sipphone :<br /> SIPPhone issue an INVITE<br />&= nbsp; this INVITE is forwarded to PortaSIP following the opensbc route<br /= > PortaSIP answers an 'SIP/2.0 401 Unauthorized' because no a= uthentication information are available in the request<br /> Here is = the problem : OPENSBC TRANSLATE IT INTO A 'SIP/2.0 407 Proxy Authentica= tion Required'. As a result, my sipphone authenticate with a 'Proxy= -Authorization:' header. <br /> This answers is forwarded to my P= ortaSIP which is waiting for a 'Authorization:' header because it i= ssued a 401 error code. As a result, PortaSIP does not authenticate my phon= e and call can not be established.<br /><br />Do you think that it is norma= l to concert 401 responses to 407 responses when openSBC is not used as a p= roxy but as a B2BUA ? I do not think so but perhaps am I missing some other= point ?<br /><br />In fact, that conversion make it impossible for me to u= se opensbc as I wish. do you think that it would be possible to take my pro= blem into account and at least offer the choice to convert or not 401 t0 40= 7, depending of the targetted use for opensbc ?<br /><br />thanks in a= dvance,<br />best regards,<br />Cedric<br /><br /> |
From: bondan <bo...@ci...> - 2007-05-11 07:57:12
|
Dear Joegen, Thank you for the fast reply. I'll download the OpenSIPStack-1.1.6-133 version and try it. Best regards, Bondan Wisnuwardhana Send an Instant Message -----Original Message----- From: Joegen E. Baclor [mailto:jb...@so...] Sent: Friday, May 11, 2007 11:38 AM To: jb...@so...; ope...@li... Cc: bondan Subject: Re: [OpenSIPStack] FW: compiled OSSPhone could not use to make call Bondan, One more thing. Looking at your logs, it seems you are using an old copy of OpenSIPStack? Please download the latest CVS head. Current version is OpenSIPStack-1.1.6-133 Joegen Joegen E. Baclor wrote: > Hi Bondan, > > I am not able to replicate your problem. I just recompiled OSSPhone > from CVS and it's able to autheticate calls against OpenSBC terminating > to an asterisk box. See logs below. > > BTW, is this modified code you are working with? > > ----------------3:44:55.989---------------- > *** LISTENER STARTED *** 127.0.0.1:5060 > ----------------3:44:56.022---------------- > *** LISTENER STARTED *** 192.168.0.161:5060 [*** DEFAULT LISTENER ***] > ----------------3:44:56.028---------------- > *** LISTENER STARTED *** 192.168.6.1:5060 > ----------------3:44:56.031---------------- > *** LISTENER STARTED *** 192.168.15.1:5060 > ----------------3:46:44.147---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1359945326143 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ee5500004e3500000099;rport=6 2787;received=125.60.243.66 > CSeq: 64 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:51:05.733---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1386096812568 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ef5b000027b10000009a;rport=6 2787;received=125.60.243.66 > CSeq: 65 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:54:05.787---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1404096821517 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f00f000014b30000009b;rport=6 2787;received=125.60.243.66 > CSeq: 66 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:56:49.262---------------- > SEND: enc=0 507 Bytes to 192.168.6.1:5061:UDP (REGISTER > sip:192.168.6.1:5061 SIP/2.0) Interface Address= > REGISTER sip:192.168.6.1:5061 SIP/2.0 > From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc > To: sip:192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3825;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas -addr=192.168.6.1;rport > CSeq: 1 REGISTER > Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc > Contact: <sip:192.168.6.1:5060;transport=udp> > User-Agent: OpenSIPStack-1.1.6-133 > Expires: 3600 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------3:56:49.320---------------- > RCV: enc=0 424 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc > To: sip:192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3185;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas -addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 1 REGISTER > Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc > Contact: <sip:192.168.6.1:5060;transport=udp> > Server: OpenSIPStack-1.1.6-133 > Expires: 3600 > Content-Length: 0 > > ----------------3:57:00.206---------------- > SEND: enc=0 762 Bytes to 192.168.6.1:5061:UDP (INVITE > sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 > INVITE sip:613@192.168.6.1 SIP/2.0 > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas -addr=192.168.6.1;rport > CSeq: 4711 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > User-Agent: OpenSIPStack-1.1.6-133 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Type: application/sdp > Content-Length: 228 > > v=0 > o=- 1178857661 1178857661 IN IP4 192.168.6.1 > s=OSS RTP Session > c=IN IP4 192.168.6.1 > t=0 0 > m=audio 5000 RTP/AVP 101 106 3 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:3 GSM/8000 > ----------------3:57:00.222---------------- > RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP > (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas -addr=192.168.6.1 > CSeq: 4711 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Content-Length: 0 > > ----------------3:57:00.248---------------- > RCV: enc=0 557 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 407 Proxy Authentication Required) > SIP/2.0 407 Proxy Authentication Required > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas -addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4711 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Proxy-Authenticate: Digest realm=192.168.6.1, > nonce="92a98f3ef419bf98501149af46ee7604", > opaque="e430240b53dfd47acdf7cbda42cdb64c", algorithm=MD5 > Content-Length: 0 > > ----------------3:57:00.252---------------- > SEND: enc=0 535 Bytes to 192.168.6.1:5061:UDP (ACK sip:613@192.168.6.1 > SIP/2.0) Interface Address=192.168.6.1 > ACK sip:613@192.168.6.1 SIP/2.0 > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas -addr=192.168.6.1;rport > CSeq: 4711 ACK > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > User-Agent: OpenSIPStack-1.1.6-133 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------3:57:00.258---------------- > SEND: enc=0 957 Bytes to 192.168.6.1:5061:UDP (INVITE > sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 > INVITE sip:613@192.168.6.1 SIP/2.0 > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas -addr=192.168.6.1;rport > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > User-Agent: OpenSIPStack-1.1.6-133 > Max-Forwards: 10 > Proxy-Authorization: Digest username="1111", realm="192.168.6.1", > nonce="92a98f3ef419bf98501149af46ee7604", uri="sip:613@192.168.6.1", > response="bf9a133f7e481849ae92fac139ad5525", algorithm=MD5 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Type: application/sdp > Content-Length: 228 > > v=0 > o=- 1178857661 1178857661 IN IP4 192.168.6.1 > s=OSS RTP Session > c=IN IP4 192.168.6.1 > t=0 0 > m=audio 5000 RTP/AVP 101 106 3 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:3 GSM/8000 > ----------------3:57:00.276---------------- > RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP > (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas -addr=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Content-Length: 0 > > ----------------3:57:01.535---------------- > RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 180 Ringing) > SIP/2.0 180 Ringing > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas -addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Server: OpenSIPStack-1.1.6-133 > Content-Length: 0 > > ----------------3:57:01.652---------------- > RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 180 Ringing) > SIP/2.0 180 Ringing > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas -addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Server: OpenSIPStack-1.1.6-133 > Content-Length: 0 > > ----------------3:57:05.840---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1422096824326 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f0c3000002e90000009c;rport=6 2787;received=125.60.243.66 > CSeq: 67 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:57:08.535---------------- > RCV: enc=0 764 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas -addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: <sip:192.168.6.1:5061> > Record-Route: <sip:192.168.6.1:5061;lr> > Server: OpenSIPStack-1.1.6-133 > Content-Type: application/sdp > Content-Length: 241 > > v=0 > o=root 29469 29469 IN IP4 69.90.168.13 > s=session > c=IN IP4 69.90.168.13 > t=0 0 > m=audio 13674 RTP/AVP 3 97 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ----------------3:57:08.782---------------- > SEND: enc=0 439 Bytes to 192.168.6.1:5061:UDP (ACK sip:192.168.6.1:5061 > SIP/2.0) Interface Address=192.168.6.1 > ACK sip:192.168.6.1:5061 SIP/2.0 > From: <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;branch=z9hG4bKdbb6f45ba7f818109c8e875f9c108cdc;uas-addr=192 .168.6.1;rport > CSeq: 4712 ACK > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > Route: <sip:192.168.6.1:5061;lr> > Content-Length: 0 > > > > > bondan wrote: > >> Resend... >> >> >> >> Send an Instant Message >> -----Original Message----- >> From: bondan [mailto:bo...@ci...] >> Sent: Friday, May 04, 2007 10:30 AM >> To: 'jb...@so...'; 'ope...@li...' >> Cc: 'ki...@ci...'; 'A.Rafmawan'; 'indragilang'; 'bondan' >> Subject: RE: [OpenSIPStack] compiled OSSPhone could not use to make call >> >> >> Dear Joegen, >> >> This attachment is the log file and capture screen of OSSPhone. >> Thank you for helping me... >> >> Best regards, >> Bondan Wisnuwardhana >> >> >> >> Send an Instant Message >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of >> Joegen E. Baclor >> Sent: Wednesday, May 02, 2007 8:45 PM >> To: ope...@li... >> Cc: ki...@ci...; 'A.Rafmawan'; 'indragilang' >> Subject: Re: [OpenSIPStack] compiled OSSPhone could not use to make call >> >> bondan wrote: >> >> >>> Dear Joegen, >>> >>> I download OSSPhone source, and compile it into application. With the >>> application, I could login into asterisk server and receive call too. >>> But when I use the application to make a call, the error message >>> "Status: 407 Proxy Authentication Required" is appear. >>> >>> I use codec setting : >>> >>> Low Bit Rate : iLBC-13k3 >>> >>> High Bit Rate : G.711-ALaw-64k >>> >>> When I use OSSPhone (based on Windows installer) with the same >>> setting, the OSSPhone can use to make a call. >>> >>> Would you give advice for me how to make call from OSSPhone source? >>> Many thank for your helping,,, >>> >>> Best regards, >>> >>> Bondan Wisnuwardhana >>> >>> >>> >> Please send me the ATLSIP logs off list so I can see what went wrong. >> >> >> >>> ------------------------------------------------------------------------ >>> >>> >> ------------------------------------------------------------------------- >> 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 >> >> __________ NOD32 2235 (20070502) Information __________ >> >> This message was checked by NOD32 antivirus system. >> http://www.eset.com >> >> >> ------------------------------------------------------------------------ >> >> ----------------54:03.175---------------- >> *** LISTENER STARTED *** 127.0.0.1:5060 >> ----------------54:03.250---------------- >> *** LISTENER STARTED *** 192.168.1.70:5060 [*** DEFAULT LISTENER ***] >> ----------------54:04.340---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:04.909---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:05.899---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:07.908---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:09.453---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:10.007---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:11.004---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:11.924---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:13.007---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:15.573---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.597---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.620---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.661---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.734---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.802---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.827---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.896---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.961---------------- >> RCV: XOR=0 586 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) >> SIP/2.0 407 Proxy Authentication Required >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73;tag=as63b5a09f >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: <sip:102@192.168.1.73> >> User-Agent: Asterisk PBX >> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49982f66" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.991---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:16.053---------------- >> SEND: XOR=0 409 Bytes to 192.168.1.73:5060:UDP (ACK sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> ACK sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73;tag=as63b5a09f >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-a ddr=192.168.1.73;rport >> CSeq: 4711 ACK >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:16.164---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:16.321---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 2 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:16.353---------------- >> RCV: XOR=0 564 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-a ddr=192.168.1.73;received=192.168.1.70 >> CSeq: 2 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.70:5060;transport=udp>;expires=3600 >> Date: Fri, 04 May 2007 02:42:07 GMT >> User-Agent: Asterisk PBX >> Expires: 3600 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> >> >> ------------------------------------------------------------------------ >> >> > > > ------------------------------------------------------------------------- > 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 > > > __________ NOD32 2256 (20070510) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com |
From: Joegen E. B. <jb...@so...> - 2007-05-11 04:40:46
|
Bondan, One more thing. Looking at your logs, it seems you are using an old copy of OpenSIPStack? Please download the latest CVS head. Current version is OpenSIPStack-1.1.6-133 Joegen Joegen E. Baclor wrote: > Hi Bondan, > > I am not able to replicate your problem. I just recompiled OSSPhone > from CVS and it's able to autheticate calls against OpenSBC terminating > to an asterisk box. See logs below. > > BTW, is this modified code you are working with? > > ----------------3:44:55.989---------------- > *** LISTENER STARTED *** 127.0.0.1:5060 > ----------------3:44:56.022---------------- > *** LISTENER STARTED *** 192.168.0.161:5060 [*** DEFAULT LISTENER ***] > ----------------3:44:56.028---------------- > *** LISTENER STARTED *** 192.168.6.1:5060 > ----------------3:44:56.031---------------- > *** LISTENER STARTED *** 192.168.15.1:5060 > ----------------3:46:44.147---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1359945326143 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ee5500004e3500000099;rport=62787;received=125.60.243.66 > CSeq: 64 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:51:05.733---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1386096812568 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ef5b000027b10000009a;rport=62787;received=125.60.243.66 > CSeq: 65 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:54:05.787---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1404096821517 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f00f000014b30000009b;rport=62787;received=125.60.243.66 > CSeq: 66 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:56:49.262---------------- > SEND: enc=0 507 Bytes to 192.168.6.1:5061:UDP (REGISTER > sip:192.168.6.1:5061 SIP/2.0) Interface Address= > REGISTER sip:192.168.6.1:5061 SIP/2.0 > From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc > To: sip:192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3825;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport > CSeq: 1 REGISTER > Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc > Contact: <sip:192.168.6.1:5060;transport=udp> > User-Agent: OpenSIPStack-1.1.6-133 > Expires: 3600 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------3:56:49.320---------------- > RCV: enc=0 424 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc > To: sip:192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3185;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 1 REGISTER > Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc > Contact: <sip:192.168.6.1:5060;transport=udp> > Server: OpenSIPStack-1.1.6-133 > Expires: 3600 > Content-Length: 0 > > ----------------3:57:00.206---------------- > SEND: enc=0 762 Bytes to 192.168.6.1:5061:UDP (INVITE > sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 > INVITE sip:613@192.168.6.1 SIP/2.0 > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport > CSeq: 4711 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > User-Agent: OpenSIPStack-1.1.6-133 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Type: application/sdp > Content-Length: 228 > > v=0 > o=- 1178857661 1178857661 IN IP4 192.168.6.1 > s=OSS RTP Session > c=IN IP4 192.168.6.1 > t=0 0 > m=audio 5000 RTP/AVP 101 106 3 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:3 GSM/8000 > ----------------3:57:00.222---------------- > RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP > (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1 > CSeq: 4711 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Content-Length: 0 > > ----------------3:57:00.248---------------- > RCV: enc=0 557 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 407 Proxy Authentication Required) > SIP/2.0 407 Proxy Authentication Required > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4711 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Proxy-Authenticate: Digest realm=192.168.6.1, > nonce="92a98f3ef419bf98501149af46ee7604", > opaque="e430240b53dfd47acdf7cbda42cdb64c", algorithm=MD5 > Content-Length: 0 > > ----------------3:57:00.252---------------- > SEND: enc=0 535 Bytes to 192.168.6.1:5061:UDP (ACK sip:613@192.168.6.1 > SIP/2.0) Interface Address=192.168.6.1 > ACK sip:613@192.168.6.1 SIP/2.0 > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport > CSeq: 4711 ACK > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > User-Agent: OpenSIPStack-1.1.6-133 > Max-Forwards: 10 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Length: 0 > > ----------------3:57:00.258---------------- > SEND: enc=0 957 Bytes to 192.168.6.1:5061:UDP (INVITE > sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 > INVITE sip:613@192.168.6.1 SIP/2.0 > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > User-Agent: OpenSIPStack-1.1.6-133 > Max-Forwards: 10 > Proxy-Authorization: Digest username="1111", realm="192.168.6.1", > nonce="92a98f3ef419bf98501149af46ee7604", uri="sip:613@192.168.6.1", > response="bf9a133f7e481849ae92fac139ad5525", algorithm=MD5 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS > Content-Type: application/sdp > Content-Length: 228 > > v=0 > o=- 1178857661 1178857661 IN IP4 192.168.6.1 > s=OSS RTP Session > c=IN IP4 192.168.6.1 > t=0 0 > m=audio 5000 RTP/AVP 101 106 3 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:3 GSM/8000 > ----------------3:57:00.276---------------- > RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP > (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: sip:613@192.168.6.1 > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Content-Length: 0 > > ----------------3:57:01.535---------------- > RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 180 Ringing) > SIP/2.0 180 Ringing > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Server: OpenSIPStack-1.1.6-133 > Content-Length: 0 > > ----------------3:57:01.652---------------- > RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 180 Ringing) > SIP/2.0 180 Ringing > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Server: OpenSIPStack-1.1.6-133 > Content-Length: 0 > > ----------------3:57:05.840---------------- > RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: <sip:186...@i2...>;tag=1422096824326 > To: <sip:63.116.254.88:12000> > Via: SIP/2.0/UDP > 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f0c3000002e90000009c;rport=62787;received=125.60.243.66 > CSeq: 67 OPTIONS > Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 > Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY > Content-Length: 0 > > ----------------3:57:08.535---------------- > RCV: enc=0 764 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 > CSeq: 4712 INVITE > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: <sip:192.168.6.1:5061> > Record-Route: <sip:192.168.6.1:5061;lr> > Server: OpenSIPStack-1.1.6-133 > Content-Type: application/sdp > Content-Length: 241 > > v=0 > o=root 29469 29469 IN IP4 69.90.168.13 > s=session > c=IN IP4 69.90.168.13 > t=0 0 > m=audio 13674 RTP/AVP 3 97 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ----------------3:57:08.782---------------- > SEND: enc=0 439 Bytes to 192.168.6.1:5061:UDP (ACK sip:192.168.6.1:5061 > SIP/2.0) Interface Address=192.168.6.1 > ACK sip:192.168.6.1:5061 SIP/2.0 > From: <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc > To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df > Via: SIP/2.0/UDP > 192.168.6.1:5060;branch=z9hG4bKdbb6f45ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport > CSeq: 4712 ACK > Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc > Contact: "1111" <sip:1111@192.168.6.1:5060> > Route: <sip:192.168.6.1:5061;lr> > Content-Length: 0 > > > > > bondan wrote: > >> Resend... >> >> >> >> Send an Instant Message >> -----Original Message----- >> From: bondan [mailto:bo...@ci...] >> Sent: Friday, May 04, 2007 10:30 AM >> To: 'jb...@so...'; 'ope...@li...' >> Cc: 'ki...@ci...'; 'A.Rafmawan'; 'indragilang'; 'bondan' >> Subject: RE: [OpenSIPStack] compiled OSSPhone could not use to make call >> >> >> Dear Joegen, >> >> This attachment is the log file and capture screen of OSSPhone. >> Thank you for helping me... >> >> Best regards, >> Bondan Wisnuwardhana >> >> >> >> Send an Instant Message >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of >> Joegen E. Baclor >> Sent: Wednesday, May 02, 2007 8:45 PM >> To: ope...@li... >> Cc: ki...@ci...; 'A.Rafmawan'; 'indragilang' >> Subject: Re: [OpenSIPStack] compiled OSSPhone could not use to make call >> >> bondan wrote: >> >> >>> Dear Joegen, >>> >>> I download OSSPhone source, and compile it into application. With the >>> application, I could login into asterisk server and receive call too. >>> But when I use the application to make a call, the error message >>> "Status: 407 Proxy Authentication Required" is appear. >>> >>> I use codec setting : >>> >>> Low Bit Rate : iLBC-13k3 >>> >>> High Bit Rate : G.711-ALaw-64k >>> >>> When I use OSSPhone (based on Windows installer) with the same >>> setting, the OSSPhone can use to make a call. >>> >>> Would you give advice for me how to make call from OSSPhone source? >>> Many thank for your helping,,, >>> >>> Best regards, >>> >>> Bondan Wisnuwardhana >>> >>> >>> >> Please send me the ATLSIP logs off list so I can see what went wrong. >> >> >> >>> ------------------------------------------------------------------------ >>> >>> >> ------------------------------------------------------------------------- >> 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 >> >> __________ NOD32 2235 (20070502) Information __________ >> >> This message was checked by NOD32 antivirus system. >> http://www.eset.com >> >> >> ------------------------------------------------------------------------ >> >> ----------------54:03.175---------------- >> *** LISTENER STARTED *** 127.0.0.1:5060 >> ----------------54:03.250---------------- >> *** LISTENER STARTED *** 192.168.1.70:5060 [*** DEFAULT LISTENER ***] >> ----------------54:04.340---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:04.909---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:05.899---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:07.908---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:09.453---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:10.007---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:11.004---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:11.924---------------- >> SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= >> REGISTER sip:192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> >> Expires: 3600 >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:13.007---------------- >> SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> INVITE sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Type: application/sdp >> Content-Length: 231 >> >> v=0 >> o=- 1178246522 1178246522 IN IP4 192.168.1.70 >> s=OSS RTP Session >> c=IN IP4 192.168.1.70 >> t=0 0 >> m=audio 5000 RTP/AVP 101 106 8 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=rtpmap:106 iLBC/8000 >> a=rtpmap:8 PCMA/8000 >> >> >> ----------------54:15.573---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.597---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.620---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.661---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.734---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.802---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.827---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.896---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.961---------------- >> RCV: XOR=0 586 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) >> SIP/2.0 407 Proxy Authentication Required >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73;tag=as63b5a09f >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 4711 INVITE >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: <sip:102@192.168.1.73> >> User-Agent: Asterisk PBX >> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49982f66" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:15.991---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:16.053---------------- >> SEND: XOR=0 409 Bytes to 192.168.1.73:5060:UDP (ACK sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 >> ACK sip:102@192.168.1.73 SIP/2.0 >> From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 >> To: sip:102@192.168.1.73;tag=as63b5a09f >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport >> CSeq: 4711 ACK >> Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 >> Contact: "101" <sip:101@192.168.1.70:5060> >> Max-Forwards: 70 >> Content-Length: 0 >> >> ----------------54:16.164---------------- >> RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) >> SIP/2.0 401 Unauthorized >> From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 1 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:16.321---------------- >> RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) >> SIP/2.0 100 Trying >> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 >> To: sip:101@192.168.1.73 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 2 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.73> >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> ----------------54:16.353---------------- >> RCV: XOR=0 564 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 200 OK) >> SIP/2.0 200 OK >> From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 >> To: sip:101@192.168.1.73;tag=as087cafd3 >> Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 >> CSeq: 2 REGISTER >> Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 >> Contact: <sip:101@192.168.1.70:5060;transport=udp>;expires=3600 >> Date: Fri, 04 May 2007 02:42:07 GMT >> User-Agent: Asterisk PBX >> Expires: 3600 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Content-Length: 0 >> >> >> >> ------------------------------------------------------------------------ >> >> > > > ------------------------------------------------------------------------- > 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. 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From: Joegen E. B. <jb...@so...> - 2007-05-11 04:34:14
|
Hi Bondan, I am not able to replicate your problem. I just recompiled OSSPhone from CVS and it's able to autheticate calls against OpenSBC terminating to an asterisk box. See logs below. BTW, is this modified code you are working with? ----------------3:44:55.989---------------- *** LISTENER STARTED *** 127.0.0.1:5060 ----------------3:44:56.022---------------- *** LISTENER STARTED *** 192.168.0.161:5060 [*** DEFAULT LISTENER ***] ----------------3:44:56.028---------------- *** LISTENER STARTED *** 192.168.6.1:5060 ----------------3:44:56.031---------------- *** LISTENER STARTED *** 192.168.15.1:5060 ----------------3:46:44.147---------------- RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: <sip:186...@i2...>;tag=1359945326143 To: <sip:63.116.254.88:12000> Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ee5500004e3500000099;rport=62787;received=125.60.243.66 CSeq: 64 OPTIONS Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY Content-Length: 0 ----------------3:51:05.733---------------- RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: <sip:186...@i2...>;tag=1386096812568 To: <sip:63.116.254.88:12000> Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKc0a800a1000000104643ef5b000027b10000009a;rport=62787;received=125.60.243.66 CSeq: 65 OPTIONS Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY Content-Length: 0 ----------------3:54:05.787---------------- RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: <sip:186...@i2...>;tag=1404096821517 To: <sip:63.116.254.88:12000> Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f00f000014b30000009b;rport=62787;received=125.60.243.66 CSeq: 66 OPTIONS Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY Content-Length: 0 ----------------3:56:49.262---------------- SEND: enc=0 507 Bytes to 192.168.6.1:5061:UDP (REGISTER sip:192.168.6.1:5061 SIP/2.0) Interface Address= REGISTER sip:192.168.6.1:5061 SIP/2.0 From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc To: sip:192.168.6.1 Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3825;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport CSeq: 1 REGISTER Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc Contact: <sip:192.168.6.1:5060;transport=udp> User-Agent: OpenSIPStack-1.1.6-133 Expires: 3600 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------3:56:49.320---------------- RCV: enc=0 424 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: <sip:192.168.6.1>;tag=76e9d65ba7f818109c8c875f9c108cdc To: sip:192.168.6.1 Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3185;branch=z9hG4bK76e9d65ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 CSeq: 1 REGISTER Call-ID: 76e9d65b-a7f8-1810-9f09-875f9c108cdc Contact: <sip:192.168.6.1:5060;transport=udp> Server: OpenSIPStack-1.1.6-133 Expires: 3600 Content-Length: 0 ----------------3:57:00.206---------------- SEND: enc=0 762 Bytes to 192.168.6.1:5061:UDP (INVITE sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 INVITE sip:613@192.168.6.1 SIP/2.0 From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: sip:613@192.168.6.1 Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport CSeq: 4711 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Contact: "1111" <sip:1111@192.168.6.1:5060> User-Agent: OpenSIPStack-1.1.6-133 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 228 v=0 o=- 1178857661 1178857661 IN IP4 192.168.6.1 s=OSS RTP Session c=IN IP4 192.168.6.1 t=0 0 m=audio 5000 RTP/AVP 101 106 3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:106 iLBC/8000 a=rtpmap:3 GSM/8000 ----------------3:57:00.222---------------- RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP (SIP/2.0 100 Trying) SIP/2.0 100 Trying From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: sip:613@192.168.6.1 Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1 CSeq: 4711 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Content-Length: 0 ----------------3:57:00.248---------------- RCV: enc=0 557 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 CSeq: 4711 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Proxy-Authenticate: Digest realm=192.168.6.1, nonce="92a98f3ef419bf98501149af46ee7604", opaque="e430240b53dfd47acdf7cbda42cdb64c", algorithm=MD5 Content-Length: 0 ----------------3:57:00.252---------------- SEND: enc=0 535 Bytes to 192.168.6.1:5061:UDP (ACK sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 ACK sip:613@192.168.6.1 SIP/2.0 From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: <sip:613@192.168.6.1>;tag=3bace75ba7f818108772ce63857af2df Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3826;branch=z9hG4bKd193e75ba7f818109c8d875f9c108cdc;uas-addr=192.168.6.1;rport CSeq: 4711 ACK Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Contact: "1111" <sip:1111@192.168.6.1:5060> User-Agent: OpenSIPStack-1.1.6-133 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------3:57:00.258---------------- SEND: enc=0 957 Bytes to 192.168.6.1:5061:UDP (INVITE sip:613@192.168.6.1 SIP/2.0) Interface Address=192.168.6.1 INVITE sip:613@192.168.6.1 SIP/2.0 From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: sip:613@192.168.6.1 Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport CSeq: 4712 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Contact: "1111" <sip:1111@192.168.6.1:5060> User-Agent: OpenSIPStack-1.1.6-133 Max-Forwards: 10 Proxy-Authorization: Digest username="1111", realm="192.168.6.1", nonce="92a98f3ef419bf98501149af46ee7604", uri="sip:613@192.168.6.1", response="bf9a133f7e481849ae92fac139ad5525", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 228 v=0 o=- 1178857661 1178857661 IN IP4 192.168.6.1 s=OSS RTP Session c=IN IP4 192.168.6.1 t=0 0 m=audio 5000 RTP/AVP 101 106 3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:106 iLBC/8000 a=rtpmap:3 GSM/8000 ----------------3:57:00.276---------------- RCV: enc=0 316 Bytes from RCVADDR: 192.168.0.161:RCVPORT: 5061:UDP (SIP/2.0 100 Trying) SIP/2.0 100 Trying From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: sip:613@192.168.6.1 Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3826;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1 CSeq: 4712 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Content-Length: 0 ----------------3:57:01.535---------------- RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP (SIP/2.0 180 Ringing) SIP/2.0 180 Ringing From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 CSeq: 4712 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Server: OpenSIPStack-1.1.6-133 Content-Length: 0 ----------------3:57:01.652---------------- RCV: enc=0 421 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP (SIP/2.0 180 Ringing) SIP/2.0 180 Ringing From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 CSeq: 4712 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Server: OpenSIPStack-1.1.6-133 Content-Length: 0 ----------------3:57:05.840---------------- RCV: enc=0 400 Bytes from RCVADDR: 63.116.254.88:RCVPORT: 12000:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: <sip:186...@i2...>;tag=1422096824326 To: <sip:63.116.254.88:12000> Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKc0a800a1000000104643f0c3000002e90000009c;rport=62787;received=125.60.243.66 CSeq: 67 OPTIONS Call-ID: 2D8CE8A4-1F1D-4F83-A069-09FF191DBEA6@192.168.0.161 Allow: INVITE, ACK, CANCEL, OPTIONS, INFO, MESSAGE, BYE, REFER, NOTIFY Content-Length: 0 ----------------3:57:08.535---------------- RCV: enc=0 764 Bytes from RCVADDR: 192.168.6.1:RCVPORT: 5061:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: 1111 <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df Via: SIP/2.0/UDP 192.168.6.1:5060;iid=3186;branch=z9hG4bK56b2e75ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport=5060;received=192.168.6.1 CSeq: 4712 INVITE Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Contact: <sip:192.168.6.1:5061> Record-Route: <sip:192.168.6.1:5061;lr> Server: OpenSIPStack-1.1.6-133 Content-Type: application/sdp Content-Length: 241 v=0 o=root 29469 29469 IN IP4 69.90.168.13 s=session c=IN IP4 69.90.168.13 t=0 0 m=audio 13674 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ----------------3:57:08.782---------------- SEND: enc=0 439 Bytes to 192.168.6.1:5061:UDP (ACK sip:192.168.6.1:5061 SIP/2.0) Interface Address=192.168.6.1 ACK sip:192.168.6.1:5061 SIP/2.0 From: <sip:1111@192.168.6.1>;tag=d193e75ba7f818109c8e875f9c108cdc To: <sip:613@192.168.6.1>;tag=b8a0e95ba7f818108772ce63857af2df Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bKdbb6f45ba7f818109c8e875f9c108cdc;uas-addr=192.168.6.1;rport CSeq: 4712 ACK Call-ID: d193e75b-a7f8-1810-9f0a-875f9c108cdc Contact: "1111" <sip:1111@192.168.6.1:5060> Route: <sip:192.168.6.1:5061;lr> Content-Length: 0 bondan wrote: > Resend... > > > > Send an Instant Message > -----Original Message----- > From: bondan [mailto:bo...@ci...] > Sent: Friday, May 04, 2007 10:30 AM > To: 'jb...@so...'; 'ope...@li...' > Cc: 'ki...@ci...'; 'A.Rafmawan'; 'indragilang'; 'bondan' > Subject: RE: [OpenSIPStack] compiled OSSPhone could not use to make call > > > Dear Joegen, > > This attachment is the log file and capture screen of OSSPhone. > Thank you for helping me... > > Best regards, > Bondan Wisnuwardhana > > > > Send an Instant Message > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Joegen E. Baclor > Sent: Wednesday, May 02, 2007 8:45 PM > To: ope...@li... > Cc: ki...@ci...; 'A.Rafmawan'; 'indragilang' > Subject: Re: [OpenSIPStack] compiled OSSPhone could not use to make call > > bondan wrote: > >> Dear Joegen, >> >> I download OSSPhone source, and compile it into application. With the >> application, I could login into asterisk server and receive call too. >> But when I use the application to make a call, the error message >> "Status: 407 Proxy Authentication Required" is appear. >> >> I use codec setting : >> >> Low Bit Rate : iLBC-13k3 >> >> High Bit Rate : G.711-ALaw-64k >> >> When I use OSSPhone (based on Windows installer) with the same >> setting, the OSSPhone can use to make a call. >> >> Would you give advice for me how to make call from OSSPhone source? >> Many thank for your helping,,, >> >> Best regards, >> >> Bondan Wisnuwardhana >> >> > > Please send me the ATLSIP logs off list so I can see what went wrong. > > >> ------------------------------------------------------------------------ >> > > ------------------------------------------------------------------------- > 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 > > __________ NOD32 2235 (20070502) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > > ------------------------------------------------------------------------ > > ----------------54:03.175---------------- > *** LISTENER STARTED *** 127.0.0.1:5060 > ----------------54:03.250---------------- > *** LISTENER STARTED *** 192.168.1.70:5060 [*** DEFAULT LISTENER ***] > ----------------54:04.340---------------- > SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= > REGISTER sip:192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > ----------------54:04.909---------------- > SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= > REGISTER sip:192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > ----------------54:05.899---------------- > SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= > REGISTER sip:192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > ----------------54:07.908---------------- > SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= > REGISTER sip:192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > ----------------54:09.453---------------- > SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > INVITE sip:102@192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 > To: sip:102@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 4711 INVITE > Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060> > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 231 > > v=0 > o=- 1178246522 1178246522 IN IP4 192.168.1.70 > s=OSS RTP Session > c=IN IP4 192.168.1.70 > t=0 0 > m=audio 5000 RTP/AVP 101 106 8 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:8 PCMA/8000 > > > ----------------54:10.007---------------- > SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > INVITE sip:102@192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 > To: sip:102@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 4711 INVITE > Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060> > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 231 > > v=0 > o=- 1178246522 1178246522 IN IP4 192.168.1.70 > s=OSS RTP Session > c=IN IP4 192.168.1.70 > t=0 0 > m=audio 5000 RTP/AVP 101 106 8 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:8 PCMA/8000 > > > ----------------54:11.004---------------- > SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > INVITE sip:102@192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 > To: sip:102@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 4711 INVITE > Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060> > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 231 > > v=0 > o=- 1178246522 1178246522 IN IP4 192.168.1.70 > s=OSS RTP Session > c=IN IP4 192.168.1.70 > t=0 0 > m=audio 5000 RTP/AVP 101 106 8 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:8 PCMA/8000 > > > ----------------54:11.924---------------- > SEND: XOR=0 426 Bytes to 192.168.1.73:5060:UDP (REGISTER sip:192.168.1.73 SIP/2.0) Interface Address= > REGISTER sip:192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > ----------------54:13.007---------------- > SEND: XOR=0 668 Bytes to 192.168.1.73:5060:UDP (INVITE sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > INVITE sip:102@192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 > To: sip:102@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 4711 INVITE > Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060> > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 231 > > v=0 > o=- 1178246522 1178246522 IN IP4 192.168.1.70 > s=OSS RTP Session > c=IN IP4 192.168.1.70 > t=0 0 > m=audio 5000 RTP/AVP 101 106 8 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:106 iLBC/8000 > a=rtpmap:8 PCMA/8000 > > > ----------------54:15.573---------------- > RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.597---------------- > RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) > SIP/2.0 401 Unauthorized > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73;tag=as087cafd3 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.620---------------- > RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.661---------------- > RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) > SIP/2.0 401 Unauthorized > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73;tag=as087cafd3 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.734---------------- > RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.802---------------- > RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) > SIP/2.0 401 Unauthorized > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73;tag=as087cafd3 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.827---------------- > RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.896---------------- > RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) > SIP/2.0 401 Unauthorized > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73;tag=as087cafd3 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.961---------------- > RCV: XOR=0 586 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) > SIP/2.0 407 Proxy Authentication Required > From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 > To: sip:102@192.168.1.73;tag=as63b5a09f > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 4711 INVITE > Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 > Contact: <sip:102@192.168.1.73> > User-Agent: Asterisk PBX > Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49982f66" > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:15.991---------------- > RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:16.053---------------- > SEND: XOR=0 409 Bytes to 192.168.1.73:5060:UDP (ACK sip:102@192.168.1.73 SIP/2.0) Interface Address=192.168.1.70 > ACK sip:102@192.168.1.73 SIP/2.0 > From: 101 <sip:101@192.168.1.73>;tag=49653a2199f81810841384af902b1204 > To: sip:102@192.168.1.73;tag=as63b5a09f > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=2;branch=z9hG4bK49653a2199f81810841284af902b1204;uas-addr=192.168.1.73;rport > CSeq: 4711 ACK > Call-ID: 14593a21-99f8-1810-8f86-84af902b1204 > Contact: "101" <sip:101@192.168.1.70:5060> > Max-Forwards: 70 > Content-Length: 0 > > ----------------54:16.164---------------- > RCV: XOR=0 566 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) > SIP/2.0 401 Unauthorized > From: 101 <sip:101@192.168.1.73>;tag=f4b7322199f81810841284af902b1204 > To: sip:101@192.168.1.73;tag=as087cafd3 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5ed0322199f81810841284af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 1 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c7fc4df" > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:16.321---------------- > RCV: XOR=0 469 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 > To: sip:101@192.168.1.73 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 2 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.73> > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > ----------------54:16.353---------------- > RCV: XOR=0 564 Bytes from RCVADDR: 192.168.1.73:RCVPORT: 5060:UDP (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: 101 <sip:101@192.168.1.73>;tag=350d442199f81810841384af902b1204 > To: sip:101@192.168.1.73;tag=as087cafd3 > Via: SIP/2.0/UDP 192.168.1.70:5060;iid=1;branch=z9hG4bK5013442199f81810841384af902b1204;uas-addr=192.168.1.73;received=192.168.1.70 > CSeq: 2 REGISTER > Call-ID: 0ebe3221-99f8-1810-8f85-84af902b1204 > Contact: <sip:101@192.168.1.70:5060;transport=udp>;expires=3600 > Date: Fri, 04 May 2007 02:42:07 GMT > User-Agent: Asterisk PBX > Expires: 3600 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > > > ------------------------------------------------------------------------ > |
From: bondan <bo...@ci...> - 2007-05-11 03:04:55
|
Resend... Send an Instant Message -----Original Message----- From: bondan [mailto:bo...@ci...] Sent: Friday, May 04, 2007 10:30 AM To: 'jb...@so...'; 'ope...@li...' Cc: 'ki...@ci...'; 'A.Rafmawan'; 'indragilang'; 'bondan' Subject: RE: [OpenSIPStack] compiled OSSPhone could not use to make call Dear Joegen, This attachment is the log file and capture screen of OSSPhone. Thank you for helping me... Best regards, Bondan Wisnuwardhana Send an Instant Message -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Wednesday, May 02, 2007 8:45 PM To: ope...@li... Cc: ki...@ci...; 'A.Rafmawan'; 'indragilang' Subject: Re: [OpenSIPStack] compiled OSSPhone could not use to make call bondan wrote: > > Dear Joegen, > > I download OSSPhone source, and compile it into application. With the > application, I could login into asterisk server and receive call too. > But when I use the application to make a call, the error message > "Status: 407 Proxy Authentication Required" is appear. > > I use codec setting : > > Low Bit Rate : iLBC-13k3 > > High Bit Rate : G.711-ALaw-64k > > When I use OSSPhone (based on Windows installer) with the same > setting, the OSSPhone can use to make a call. > > Would you give advice for me how to make call from OSSPhone source? > Many thank for your helping,,, > > Best regards, > > Bondan Wisnuwardhana > Please send me the ATLSIP logs off list so I can see what went wrong. > ------------------------------------------------------------------------ ------------------------------------------------------------------------- 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 __________ NOD32 2235 (20070502) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com |
From: Joegen E. B. <jb...@so...> - 2007-05-11 02:09:19
|
Hi Everyone, The recent posts from the OSS forum have HTML tags in them which makes them almost unreadable. This has now been corrected. For those who wants to read the posts, please visit this link http://www.opensourcesip.org:8080/jiveforums/forum.jspa?forumID=4&start=0 Joegen |
From: joegen <jb...@so...> - 2007-05-11 01:52:57
|
Hi there,<br /><br />Can you give mo info about your topology. Is SER= hairpinning calls back to OpenSBC?<br /><br />UA1 ---> OSBC ---> SER= ---> OSBC ---> UA2 ?<br /><br /><br />----------------<br /><br /><s= trong><span class=3D"jive-subject">OpenSBC and SER, invites are not passed = from the inside interface to UAC </span><br /></strong><span class=3D"jive-= description"><span style=3D"font-family: Verdana">Posted: May 8, 2007 9:23 = PM <br /></span></span><br />Hi,<br /><br />sorry the background is a bit l= ong winded, but I think essential to understand what I am seeing.<br /><br = />I have a setup with an external network and an internal network, with Ope= nSBC acting as the "bridge" between the two. On the outside= are the phones (UAC) and on the inside is a SER (sip express router) actoi= ng as the soft switch/registrar.<br /><br />OpenSBC is setup in "full&= quot; mode with proxy RTP checked, and with no routing added into the confi= guration pages. OpenSBC does name resolution against a DNS server.&nb= sp;<br /><br />The clients register with the ser server (called ser1.locald= omain) with the outbound proxy set to the external interface address of the= OpenSBC. The OpenSBC resolves ser1.localdomain to the internal addre= ss of the SER server.<br /><br />Using SIPFlow, and reviewing the logs of t= he client, i can get the clients to register with the SER server quite succ= esfully.<br /><br />When I call from one client to another the problems sta= rt. I see the invite arrive at the OenSBC and traverse to the SER (th= e sip handling 100 and ACK between OpenSBC and client work fine). SER handl= es the invite correctly and makes an INVITE to the requested client, the In= vite traversing to the internal interface of OPENSBC. This invite nev= er traverses OPenSBC and is never seen on the external interface.<br = /><br />I guess I have set something up incorreclty, but I need some direct= ion on what I should have done.<br /><br />Thanks<br />Tim |
From: joegen <jb...@so...> - 2007-05-11 01:50:31
|
<br />Kolneath,<br /><br />There is a mailing list available for dedicated = to OpenSIPStack development at <a href=3D"https://lists.sourceforge.net/lis= ts/listinfo/opensipstack-devel">https://lists.sourceforge.net/lists/listinf= o/opensipstack-devel</a><br /><br />Although there is no formal documentati= on yet for the API, questions concerning development of OpenSBC and OpenSIP= Stack are always answered in this list.<br /><br /><br /><strong><span clas= s=3D"jive-subject">OSBC development guidelines </span><br /></strong><span = class=3D"jive-description"><span style=3D"font-family: Verdana">Posted: May= 9, 2007 2:19 PM <br /></span></span><br />Hello,<br /><br />Thank you for = your help. Finally, I am able to arrive to the configuration http page. Now= , I am interested and would like to explore more, but unfortunately I can&#= 39;t find any development guidelines document explaining how to use the exi= sting OSBC,OSS source to develop the customized OSBC.<br /><br />Any help a= nd guideline appreciated.<br /><br />Thanks n Regards,<br />Kolneath SOMETH= <br /> |
From: joegen <jb...@so...> - 2007-05-11 01:44:49
|
<p>Correct. B2BUpperReg is the correct setting for your topology. In this mode, if you point your domains to resolve to OpenSBC, OpenSBC would think it is authoritative of that domain and would therefore try to authenticate the calls. For upper registration to work, you need to point the domains to PortaSIP. And set upper registration routes to that domain.<br /><br />-- Joegen --</p> |
From: joegen <jb...@so...> - 2007-05-11 01:39:26
|
What event package are you Subscribing to? Assume this is MWI? Currently work in MWI support in OpenSBC is incomplete peding completion of the VXML IVR. If you use OpenSBC just as an outbound Proxy, then Subscribes would just pass through.<br /><br />-- Joegen -- |
From: <jer...@ze...> - 2007-05-11 01:29:32
|
Hi, Since one month that I try to have someone with your stack but nothing... The lib file that has generated by the OpenSIPStack project is really functional? When I have this library, just by generating project, can I make my own sip phone, with no problems? This library contains all in? All that she must have to work? Regards, Jeremy Counet. [att1.html] ------------------------------------------------------------------------- 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...> - 2007-05-11 01:29:29
|
Hi Everyone, I ave just checked in code that fixes memory leaks in SIPMessage.cxx introduced with recent changes to the stack. Together with this fix is the addition of code in B2BConnection to send a 391 response for reInvites while a previous IST transaction is still in progress. This would avoid race conditions caused by clicking hold and unhold at a very fast rate. Garbage collection throttling is also introduces in GC which proves to have better performance over the old none throttled implementation. Now is a good time to bring out your favorite CVS client. We are weeks away from a new development release. Joegen ------------------------------------------------------------------------- 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: Julian T. <jf...@id...> - 2007-05-11 01:29:27
|
Hi guys, The last update for compatability issues with xten also solved other problem I've encountered specially with rtc and other softphones from taiwan and korea. Regards, Julian F. Tasis, III [att1.html] ------------------------------------------------------------------------- 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...> - 2007-05-11 01:29:26
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This is done by setting "Custom Build Steps" in the property for the=20 input files involved. sqlite3.dll.in Flex.exe.in expat.dll.in Bison.simple.in Bison.hairy.in Bison.exe.in For the Configuration build steps, it is done using "Custom Build Steps"=20 for ossbuildopts.in by calling configure.exe Counet J=E9r=E9my wrote: > > Hi, > > I have very difficult to understand why and how the OpenSIPStack=20 > project makes this before compile c++ sources files. > > Copying sqlite3.dll > > 1 file(s) copied. > > Copying Flex.exe > > 1 file(s) copied. > > Copying expat.dll > > 1 file(s) copied. > > Copying Bison.simple > > 1 file(s) copied. > > Copying Bison.hairy > > 1 file(s) copied. > > Copying Bison > > 1 file(s) copied. > > Configuration Build Options > > Opened configure.ac > > Located DNS Resolver at c:\Program Files\Microsoft SDK\ > > Located IPv6 Support at c:\Program Files\Microsoft SDK\include\ > > Located QoS Support at c:\Program Files\Microsoft SDK\include\ > > Searching C:\ > > Located OPAL at C:\Documents and Settings\JCOUNET\My Documents\Visual=20 > Studio 2005\Projects\opensipstack-1.1.5-MPL-cxx\ > > Located SQLITE at C:\Documents and Settings\JCOUNET\My=20 > Documents\Visual Studio=20 > 2005\Projects\opensipstack-1.1.5-MPL-cxx\external\CppSQLite\3.1\Common\ > > Located CPPSQLITE at C:\Documents and Settings\JCOUNET\My=20 > Documents\Visual Studio=20 > 2005\Projects\opensipstack-1.1.5-MPL-cxx\external\CppSQLite\3.1\Common\ > > Located Expat XML at C:\Documents and Settings\JCOUNET\My=20 > Documents\Visual Studio=20 > 2005\Projects\opensipstack-1.1.5-MPL-cxx\external\Expat-2.0.0\ > > Byte Order set to PLITTLE_ENDIAN > > Semaphore Support enabled > > Plugin Support enabled > > DNS Resolver enabled > > Cyrus SASL DISABLED > > Open SSL DISABLED > > Open SSL AES DISABLED > > Expat XML enabled > > VXML enabled > > Jabber enabled > > XMLRPC enabled > > SOAP enabled > > Speech API DISABLED > > IPv6 Support enabled > > QoS Support enabled > > Simple DirectMedia Layer DISABLED > > Video enabled > > ASN Support enabled > > STUN Support enabled > > pipe channel Support enabled > > DTMF encoding/decoding support enabled > > WAV file support enabled > > SOCKS protocol support enabled > > FTP protocol support enabled > > SNMP protocol support enabled > > Telnet protocol support enabled > > remote connection support enabled > > Serial Support enabled > > POP3/SMTP Support enabled > > HTTP Support enabled > > HTTP service Support enabled > > Config file protocol support enabled > > Socket aggregation support enabled > > Video file support enabled > > Audio enabled > > ffmpeg RFC2190 Library DISABLED > > iLBC support enabled > > Quicknet Internet xJACK cards enabled > > Voice Blaster enabled > > Voicetrnix vpb support DISABLED > > SQLITE enabled > > CPPSQLITE enabled > > MYACMLIB DISABLED > > VOICEAGE DISABLED > > OPAL enabled > > LibJingle DISABLED > > Configuration completed. > > =C9tape de g=E9n=E9ration personnalis=E9e en cours > > getdate.y contains 8 shift/reduce conflicts and 18 reduce/reduce=20 > conflicts. > > ? > > Can you explain to me this first step in the project?s generation? > > I must to know that because when I change project properties: Type of=20 > configuration: Dynamic Library (.dll), is it impossible to generate=20 > the project if I don?t have this with project properties: Type of=20 > configuration: Static Library (.lib) before. > > Why with the lib generation, all are generate in the good step? Which=20 > configuration makes that correctly and how can I do the same with a=20 > dll generation? > > Someone can help me please? > > Thanks a lot. > > Jeremy Counet. > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > 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 > =20 ------------------------------------------------------------------------- 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: <jer...@ze...> - 2007-05-11 01:29:23
|
Hi, =20 I have very difficult to understand why and how the OpenSIPStack project ma= kes this before compile c++ sources files. =20 Copying sqlite3.dll 1 file(s) copied. Copying Flex.exe 1 file(s) copied. Copying expat.dll 1 file(s) copied. Copying Bison.simple 1 file(s) copied. Copying Bison.hairy 1 file(s) copied. Copying Bison 1 file(s) copied. Configuration Build Options Opened configure.ac Located DNS Resolver at c:\Program Files\Microsoft SDK\ Located IPv6 Support at c:\Program Files\Microsoft SDK\include\ Located QoS Support at c:\Program Files\Microsoft SDK\include\ Searching C:\ Located OPAL at C:\Documents and Settings\JCOUNET\My Documents\Visual Studi= o 2005\Projects\opensipstack-1.1.5-MPL-cxx\ Located SQLITE at C:\Documents and Settings\JCOUNET\My Documents\Visual Stu= dio 2005\Projects\opensipstack-1.1.5-MPL-cxx\external\CppSQLite\3.1\Common\ Located CPPSQLITE at C:\Documents and Settings\JCOUNET\My Documents\Visual = Studio 2005\Projects\opensipstack-1.1.5-MPL-cxx\external\CppSQLite\3.1\Comm= on\ Located Expat XML at C:\Documents and Settings\JCOUNET\My Documents\Visual = Studio 2005\Projects\opensipstack-1.1.5-MPL-cxx\external\Expat-2.0.0\ Byte Order set to PLITTLE_ENDIAN Semaphore Support enabled Plugin Support enabled DNS Resolver enabled Cyrus SASL DISABLED Open SSL DISABLED Open SSL AES DISABLED Expat XML enabled VXML enabled Jabber enabled XMLRPC enabled SOAP enabled Speech API DISABLED IPv6 Support enabled QoS Support enabled Simple DirectMedia Layer DISABLED Video enabled ASN Support enabled STUN Support enabled pipe channel Support enabled DTMF encoding/decoding support enabled WAV file support enabled SOCKS protocol support enabled FTP protocol support enabled SNMP protocol support enabled Telnet protocol support enabled remote connection support enabled Serial Support enabled POP3/SMTP Support enabled HTTP Support enabled HTTP service Support enabled Config file protocol support enabled Socket aggregation support enabled Video file support enabled Audio enabled ffmpeg RFC2190 Library DISABLED iLBC support enabled Quicknet Internet xJACK cards enabled Voice Blaster enabled Voicetrnix vpb support DISABLED SQLITE enabled CPPSQLITE enabled MYACMLIB DISABLED VOICEAGE DISABLED OPAL enabled LibJingle DISABLED Configuration completed. =C9tape de g=E9n=E9ration personnalis=E9e en cours getdate.y contains 8 shift/reduce conflicts and 18 reduce/reduce conflicts. ... =20 Can you explain to me this first step in the project's generation? I must to know that because when I change project properties: Type of confi= guration: Dynamic Library (.dll), is it impossible to generate the project = if I don't have this with project properties: Type of configuration: Static= Library (.lib) before. Why with the lib generation, all are generate in the good step? Which confi= guration makes that correctly and how can I do the same with a dll generati= on? =20 Someone can help me please? =20 Thanks a lot. =20 Jeremy Counet. =20 [att1.html] ------------------------------------------------------------------------- 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 |