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From: Jonas G. <jon...@gm...> - 2007-11-01 07:30:27
|
That helped. Thank you. On 11/1/07, Ilian Jeri C. Pinzon <ip...@so...> wrote: > Hi, > > Most probably, the UA you called does not support the codecs you set. > From what I see, > you have iLBC and GSM (which are the default if no codecs are set, by > the way.). Try using > G.711. > > - Ilian > > Jonas Gauffin wrote: > > Ignore the event part of my previous message. The events work fine. > > > > I still haven't got makeCall to work. Here is a log: > > > > > > ----------------7:49:55.642---------------- > > SEND: XOR=0 434 Bytes to 192.168.0.58:5060:UDP (REGISTER > > sip:192.168.0.58 SIP/2.0) Interface Address= > > REGISTER sip:192.168.0.58 SIP/2.0 > > From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1 > > To: sip:jo...@th... > > Via: SIP/2.0/UDP > > 192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport > > CSeq: 1 REGISTER > > Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1 > > Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp> > > Expires: 3600 > > Max-Forwards: 70 > > Content-Length: 0 > > > > > > ----------------7:49:55.799---------------- > > RCV: XOR=0 627 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP > > (SIP/2.0 200 OK) > > SIP/2.0 200 OK > > From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1 > > To: <sip:jo...@th...>;tag=KUD9S70vH8X7a > > Via: SIP/2.0/UDP > > 192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 > > CSeq: 1 REGISTER > > Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1 > > Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp>;expires=3600 > > User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: 100rel, precondition, timer > > Content-Length: 0 > > > > > > ----------------7:49:57.641---------------- > > SEND: XOR=0 674 Bytes to 192.168.0.58:5060:UDP (INVITE > > sip:83...@th... SIP/2.0) Interface Address=192.168.0.58 > > INVITE sip:83...@th... SIP/2.0 > > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > > To: sip:83...@th... > > Via: SIP/2.0/UDP > > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport > > CSeq: 4711 INVITE > > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > > Contact: "jonas" <sip:jonas@192.168.0.58:5061> > > Max-Forwards: 70 > > Content-Type: application/sdp > > Content-Length: 230 > > > > v=0 > > o=- 1193852436 1193852436 IN IP4 192.168.0.58 > > s=OSS RTP Session > > c=IN IP4 192.168.0.58 > > 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 > > > > > > > > ----------------7:49:57.648---------------- > > RCV: XOR=0 381 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP > > (SIP/2.0 100 Trying) > > SIP/2.0 100 Trying > > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > > To: <sip:83...@th...> > > Via: SIP/2.0/UDP > > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 > > CSeq: 4711 INVITE > > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > > User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia > > Content-Length: 0 > > > > > > ----------------7:49:57.659---------------- > > RCV: XOR=0 594 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP > > (SIP/2.0 488 Not Acceptable Here) > > SIP/2.0 488 Not Acceptable Here > > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > > To: <sip:83...@th...>;tag=m461U2H0eHmtp > > Via: SIP/2.0/UDP > > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 > > CSeq: 4711 INVITE > > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > > User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: 100rel, precondition, timer > > Content-Length: 0 > > > > > > ----------------7:49:57.663---------------- > > SEND: XOR=0 422 Bytes to 192.168.0.58:5060:UDP (ACK > > sip:83...@th... SIP/2.0) Interface Address=192.168.0.58 > > ACK sip:83...@th... SIP/2.0 > > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > > To: <sip:83...@th...>;tag=m461U2H0eHmtp > > Via: SIP/2.0/UDP > > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport > > CSeq: 4711 ACK > > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > > Contact: "jonas" <sip:jonas@192.168.0.58:5061> > > Max-Forwards: 70 > > Content-Length: 0 > > > > > > > > > > On 10/31/07, Jonas Gauffin <jon...@gm...> wrote: > > > >> Hello > >> > >> I've downloaded ATLsip and is testing it with C#. > >> I'm using FreeSWITCH as sip pbx. > >> > >> I can register OK, but I do not receive any events to any of the > >> handler methods. Any suggestions on why? > >> > >> What should the SIP Uri look like when making calls? I've tried some > >> different variants, but FreeSWITCH returns INCOMPATIBLE_DESTINATION > >> on the INVITE. > >> > >> Thanks, > >> Jonas > >> > >> > > > > ------------------------------------------------------------------------- > > This SF.net email is sponsored by: Splunk Inc. > > Still grepping through log files to find problems? Stop. > > Now Search log events and configuration files using AJAX and a browser. > > Download your FREE copy of Splunk now >> http://get.splunk.com/ > > _______________________________________________ > > Opensipstack-atlsipdevel mailing list > > Ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > |
|
From: Ilian J. C. P. <ip...@so...> - 2007-11-01 02:45:51
|
Hi, Most probably, the UA you called does not support the codecs you set. From what I see, you have iLBC and GSM (which are the default if no codecs are set, by the way.). Try using G.711. - Ilian Jonas Gauffin wrote: > Ignore the event part of my previous message. The events work fine. > > I still haven't got makeCall to work. Here is a log: > > > ----------------7:49:55.642---------------- > SEND: XOR=0 434 Bytes to 192.168.0.58:5060:UDP (REGISTER > sip:192.168.0.58 SIP/2.0) Interface Address= > REGISTER sip:192.168.0.58 SIP/2.0 > From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1 > To: sip:jo...@th... > Via: SIP/2.0/UDP > 192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport > CSeq: 1 REGISTER > Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1 > Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > > ----------------7:49:55.799---------------- > RCV: XOR=0 627 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP > (SIP/2.0 200 OK) > SIP/2.0 200 OK > From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1 > To: <sip:jo...@th...>;tag=KUD9S70vH8X7a > Via: SIP/2.0/UDP > 192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 > CSeq: 1 REGISTER > Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1 > Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp>;expires=3600 > User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, precondition, timer > Content-Length: 0 > > > ----------------7:49:57.641---------------- > SEND: XOR=0 674 Bytes to 192.168.0.58:5060:UDP (INVITE > sip:83...@th... SIP/2.0) Interface Address=192.168.0.58 > INVITE sip:83...@th... SIP/2.0 > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > To: sip:83...@th... > Via: SIP/2.0/UDP > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport > CSeq: 4711 INVITE > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > Contact: "jonas" <sip:jonas@192.168.0.58:5061> > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 230 > > v=0 > o=- 1193852436 1193852436 IN IP4 192.168.0.58 > s=OSS RTP Session > c=IN IP4 192.168.0.58 > 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 > > > > ----------------7:49:57.648---------------- > RCV: XOR=0 381 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP > (SIP/2.0 100 Trying) > SIP/2.0 100 Trying > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > To: <sip:83...@th...> > Via: SIP/2.0/UDP > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 > CSeq: 4711 INVITE > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia > Content-Length: 0 > > > ----------------7:49:57.659---------------- > RCV: XOR=0 594 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP > (SIP/2.0 488 Not Acceptable Here) > SIP/2.0 488 Not Acceptable Here > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > To: <sip:83...@th...>;tag=m461U2H0eHmtp > Via: SIP/2.0/UDP > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 > CSeq: 4711 INVITE > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, precondition, timer > Content-Length: 0 > > > ----------------7:49:57.663---------------- > SEND: XOR=0 422 Bytes to 192.168.0.58:5060:UDP (ACK > sip:83...@th... SIP/2.0) Interface Address=192.168.0.58 > ACK sip:83...@th... SIP/2.0 > From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 > To: <sip:83...@th...>;tag=m461U2H0eHmtp > Via: SIP/2.0/UDP > 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport > CSeq: 4711 ACK > Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 > Contact: "jonas" <sip:jonas@192.168.0.58:5061> > Max-Forwards: 70 > Content-Length: 0 > > > > > On 10/31/07, Jonas Gauffin <jon...@gm...> wrote: > >> Hello >> >> I've downloaded ATLsip and is testing it with C#. >> I'm using FreeSWITCH as sip pbx. >> >> I can register OK, but I do not receive any events to any of the >> handler methods. Any suggestions on why? >> >> What should the SIP Uri look like when making calls? I've tried some >> different variants, but FreeSWITCH returns INCOMPATIBLE_DESTINATION >> on the INVITE. >> >> Thanks, >> Jonas >> >> > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > |
|
From: Jonas G. <jon...@gm...> - 2007-10-31 17:44:23
|
Ignore the event part of my previous message. The events work fine. I still haven't got makeCall to work. Here is a log: ----------------7:49:55.642---------------- SEND: XOR=0 434 Bytes to 192.168.0.58:5060:UDP (REGISTER sip:192.168.0.58 SIP/2.0) Interface Address= REGISTER sip:192.168.0.58 SIP/2.0 From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1 To: sip:jo...@th... Via: SIP/2.0/UDP 192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport CSeq: 1 REGISTER Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1 Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp> Expires: 3600 Max-Forwards: 70 Content-Length: 0 ----------------7:49:55.799---------------- RCV: XOR=0 627 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: jonas <sip:jo...@th...>;tag=a254b77b04fa18108176e2bfe2b376e1 To: <sip:jo...@th...>;tag=KUD9S70vH8X7a Via: SIP/2.0/UDP 192.168.0.58:5061;iid=1;branch=z9hG4bKa254b77b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 CSeq: 1 REGISTER Call-ID: a254b77b-04fa-1810-9fe4-e2bfe2b376e1 Contact: "jonas" <sip:jonas@192.168.0.58:5061;transport=udp>;expires=3600 User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Length: 0 ----------------7:49:57.641---------------- SEND: XOR=0 674 Bytes to 192.168.0.58:5060:UDP (INVITE sip:83...@th... SIP/2.0) Interface Address=192.168.0.58 INVITE sip:83...@th... SIP/2.0 From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 To: sip:83...@th... Via: SIP/2.0/UDP 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport CSeq: 4711 INVITE Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 Contact: "jonas" <sip:jonas@192.168.0.58:5061> Max-Forwards: 70 Content-Type: application/sdp Content-Length: 230 v=0 o=- 1193852436 1193852436 IN IP4 192.168.0.58 s=OSS RTP Session c=IN IP4 192.168.0.58 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 ----------------7:49:57.648---------------- RCV: XOR=0 381 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP (SIP/2.0 100 Trying) SIP/2.0 100 Trying From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 To: <sip:83...@th...> Via: SIP/2.0/UDP 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 CSeq: 4711 INVITE Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia Content-Length: 0 ----------------7:49:57.659---------------- RCV: XOR=0 594 Bytes from RCVADDR: 192.168.0.58:RCVPORT: 5060:UDP (SIP/2.0 488 Not Acceptable Here) SIP/2.0 488 Not Acceptable Here From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 To: <sip:83...@th...>;tag=m461U2H0eHmtp Via: SIP/2.0/UDP 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport=5061 CSeq: 4711 INVITE Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 User-Agent: FreeSWITCH-1.0.pre1 (6094)-mod_sofia Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Length: 0 ----------------7:49:57.663---------------- SEND: XOR=0 422 Bytes to 192.168.0.58:5060:UDP (ACK sip:83...@th... SIP/2.0) Interface Address=192.168.0.58 ACK sip:83...@th... SIP/2.0 From: jonas <sip:jo...@th...>;tag=433dba7b04fa18108178e2bfe2b376e1 To: <sip:83...@th...>;tag=m461U2H0eHmtp Via: SIP/2.0/UDP 192.168.0.58:5061;iid=2;branch=z9hG4bK433dba7b04fa18108177e2bfe2b376e1;uas-addr=192.168.0.58;rport CSeq: 4711 ACK Call-ID: a418ba7b-04fa-1810-9fe5-e2bfe2b376e1 Contact: "jonas" <sip:jonas@192.168.0.58:5061> Max-Forwards: 70 Content-Length: 0 On 10/31/07, Jonas Gauffin <jon...@gm...> wrote: > Hello > > I've downloaded ATLsip and is testing it with C#. > I'm using FreeSWITCH as sip pbx. > > I can register OK, but I do not receive any events to any of the > handler methods. Any suggestions on why? > > What should the SIP Uri look like when making calls? I've tried some > different variants, but FreeSWITCH returns INCOMPATIBLE_DESTINATION > on the INVITE. > > Thanks, > Jonas > |
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From: Jonas G. <jon...@gm...> - 2007-10-31 16:41:22
|
Hello I've downloaded ATLsip and is testing it with C#. I'm using FreeSWITCH as sip pbx. I can register OK, but I do not receive any events to any of the handler methods. Any suggestions on why? What should the SIP Uri look like when making calls? I've tried some different variants, but FreeSWITCH returns INCOMPATIBLE_DESTINATION on the INVITE. Thanks, Jonas |
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From: tom s. <tom...@gm...> - 2007-10-10 06:37:07
|
thank you Joegen ! it all makes sense now and after setting the values explicit for both REGISTER and INVITE so the proxy is the same as the asterisk server the library automatically re-sends the INVITE request with the correct details after a "407 proxy authentication required" from the server.. i am a very happy owner of a very small voip software phone now :) thank you again. the initial steps to a working program are always the hardest and i had some wrong assumptions in there. best regards, tom |
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From: Joegen E. B. <joe...@gm...> - 2007-10-10 03:17:38
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tom schuring wrote: > hello Joegen, > > > thank you for taking the time to reply. i am using ATLSIP in my own program but i am using it as an activeX component. > > > i do set the AuthenticationUser property but i can't fin REGISTER method (i assume that is handled by the DoLogin method ?). > Yes REGISTER will be sent out upon calling DoLogin() method > > my question was about how i find out what the ProxyAuthenticationUser should be. where can i find this ? > ProxyAuthenticationUser is set by the application. If you want to use the same credentials for both REGISTER and INVITE, you need to explicitly set their values as equal by calling AuthenticationUser and ProxyAuthenticationUser and feed them a common value. Do the same for the password. Do not forget to call InitializeSIP() after changing any property in ATLSIP atlsip.AuthenticationUser = atlsip.ProxyAuthenticationUser = "myusername"; atlsip.AuthenticationPassword = atlsip.ProxyAuthenticationPassword = "password"; atlsip.InitializeSIP(); > > when i use a sip client like x-lite i never specify what the ProxyAuthenticationUser is and it finds out by looking at the response from the server when it returns a "407 Proxy Authentication Required" > The server never sends the user in the challenge. You have the wrong assumption here. The only difference with x-lite is that x-lite assumes that the authentication credentials for the registrar and the proxy is always the same. ATLSIP allows you to have separate credentials for registrar and proxy so you must explicitly set them both. > > i can see in a network trace that in this "407" sip message it has a +Proxy-Authenticate: Digest realm="asterisk", nonce="453baf90"+ section that x-lite must copy to send the next INVITE.. > > > this is all what i assume happens, please let me know if this is not the case. > > > so in short i would like to know where i can find what the Proxy-Authenticate should be... > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > > |
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From: tom s. <tom...@gm...> - 2007-10-09 23:41:10
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hello Joegen, thank you for taking the time to reply. i am using ATLSIP in my own program but i am using it as an activeX component. i do set the AuthenticationUser property but i can't fin REGISTER method (i assume that is handled by the DoLogin method ?). my question was about how i find out what the ProxyAuthenticationUser should be. where can i find this ? when i use a sip client like x-lite i never specify what the ProxyAuthenticationUser is and it finds out by looking at the response from the server when it returns a "407 Proxy Authentication Required" i can see in a network trace that in this "407" sip message it has a +Proxy-Authenticate: Digest realm="asterisk", nonce="453baf90"+ section that x-lite must copy to send the next INVITE.. this is all what i assume happens, please let me know if this is not the case. so in short i would like to know where i can find what the Proxy-Authenticate should be... |
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From: Joegen E. B. <joe...@gm...> - 2007-10-09 08:25:13
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Tom, Are you using ATLSIP SIP in a custom application? Make sure you set the following SetAuthenticationUser() - REGISTER (WWW-Authneticate) SetProxyAuthenticationUser() - INVITE (Proxy-Authenticate) Then you must call Initialize() after you have set the values and every time they change. Joegen tom schuring wrote: > hello, > > > i'm trying to connect to an asterisk server using the ATLSIPobject. > > > i can connect (login allright) but when i call MakeCall() > > > i see the following in the sip logfile: > > > 2007/10/09 16:37:30.636 Transaction DBG: [CID=0x0a6d] ICT(108422460) Timer D( 32000 ms ) STARTED > 2007/10/09 16:37:30.636 UserAgent DTL: [CID=0x0a6d] *** MESSAGE ARRIVAL *** for SIP Session 12a8ee4c-d7f9-1810-9e7f-fc3ea91abefc > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] SIP/2.0 407 Proxy Authentication Required > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] From: 1234 <sip:1234@172.31.26.68>;tag=12a8ee4cd7f918109cc1fc3ea91abefc > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] To: sip:4321@172.31.26.68;tag=as37056ca2 > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Via: SIP/2.0/UDP 172.31.26.109:5060;iid=1110;branch=z9hG4bK12a8ee4cd7f918109cc0fc3ea91abefc;uas-addr=172.31.26.68;received=172.31.26.109 > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] CSeq: 4711 INVITE > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Call-ID: 12a8ee4c-d7f9-1810-9e7f-fc3ea91abefc > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Contact: <sip:4321@172.31.26.68> > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] User-Agent: Asterisk PBX > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Proxy-Authenticate: Digest realm="asterisk", nonce="2a6aec31" > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Content-Length: 0 > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] > 2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] > 2007/10/09 16:37:30.636 Call WRN: [CID=0x0a6d] *** Proxy Authentication User is EMPTY!!! *** > 2007/10/09 16:37:30.636 Call WRN: [CID=0x0a6d] *** Unable to authenticate Call!!! *** > > > > > > -at what stage can i find out what the proxyAuthenticationUser should be ? > > > is it right after i login ? > does it come back in the event arguments of OnLoginSuccessful() ? > or is there a member i can call so i can set the proxyAuthenticationUser before i call MakeCall ? > > > i looked in the eventarg of OnOutgoingCallRejected but it only shows the error message, not the expected proxyAuthenticationUser.. > > > or is there any documentation that explains this ? i have had a look at the OSSPhone but i can't find if it. > > > any help is much appreciated. > > > best regards, > > > tom > > > > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > > |
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From: tom s. <tom...@gm...> - 2007-10-09 07:12:15
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admin: can we please get rid of these test posts ? |
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From: tom s. <tom...@gm...> - 2007-10-09 07:09:03
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hello,
i'm trying to connect to an asterisk server using the ATLSIPobject.
i can connect (login allright) but when i call MakeCall()
i see the following in the sip logfile:
2007/10/09 16:37:30.636 Transaction DBG: [CID=0x0a6d] ICT(108422460) Timer D( 32000 ms ) STARTED
2007/10/09 16:37:30.636 UserAgent DTL: [CID=0x0a6d] *** MESSAGE ARRIVAL *** for SIP Session 12a8ee4c-d7f9-1810-9e7f-fc3ea91abefc
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d]
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] SIP/2.0 407 Proxy Authentication Required
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] From: 1234 <sip:1234@172.31.26.68>;tag=12a8ee4cd7f918109cc1fc3ea91abefc
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] To: sip:4321@172.31.26.68;tag=as37056ca2
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Via: SIP/2.0/UDP 172.31.26.109:5060;iid=1110;branch=z9hG4bK12a8ee4cd7f918109cc0fc3ea91abefc;uas-addr=172.31.26.68;received=172.31.26.109
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] CSeq: 4711 INVITE
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Call-ID: 12a8ee4c-d7f9-1810-9e7f-fc3ea91abefc
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Contact: <sip:4321@172.31.26.68>
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] User-Agent: Asterisk PBX
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Proxy-Authenticate: Digest realm="asterisk", nonce="2a6aec31"
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d] Content-Length: 0
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d]
2007/10/09 16:37:30.636 UserAgent DBG: [CID=0x0a6d]
2007/10/09 16:37:30.636 Call WRN: [CID=0x0a6d] *** Proxy Authentication User is EMPTY!!! ***
2007/10/09 16:37:30.636 Call WRN: [CID=0x0a6d] *** Unable to authenticate Call!!! ***
-at what stage can i find out what the proxyAuthenticationUser should be ?
is it right after i login ?
does it come back in the event arguments of OnLoginSuccessful() ?
or is there a member i can call so i can set the proxyAuthenticationUser before i call MakeCall ?
i looked in the eventarg of OnOutgoingCallRejected but it only shows the error message, not the expected proxyAuthenticationUser..
or is there any documentation that explains this ? i have had a look at the OSSPhone but i can't find if it.
any help is much appreciated.
best regards,
tom
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From: Ilian J. P. <ip...@so...> - 2007-09-12 05:01:06
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Hello. I have checked in code for USB phone support. The relevant classes are: HidManager - Manages all HIDs (i.e. USB phones) and sends events to the handlers. HidHandler - Handles/listens for events from HidManager. Override this to provide custom actions in response to events. HidDevice - Override this to implement a USB phone interface. This is where the USB phone SDK code should go. The relevant files are: Hid.cxx/h - base code for USB phone support CmHidDevice.cxx/h - the ATCOM USB phone interface implementation. This is currently implemented in the OSSPhoneSvc project under ATLSIP but it should also be trivial to do the same to OSSPhone .NET and OSSPhone MFC. Will do this later. For now, only ATCOM USB phones are supported. To enable this, you will need to request a copy of the SDK from ATCOM (http://www.atcom.cn/). Then put all relevant files (i.e. cm_hid.*) in the opensipstack/external/hid folder and then do a rebuild. Regards, Ilian |
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From: <ope...@op...> - 2007-08-22 10:29:41
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test please ignore |
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From: Joegen E. B. <joe...@gm...> - 2007-08-03 07:33:20
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Test Reply - Please Ignore Administrator wrote: > test > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > Opensipstack-atlsipdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-atlsipdevel > > |
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From: Administrator <rn...@so...> - 2007-08-03 07:29:59
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test |
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From: Joegen E. B. <joe...@gm...> - 2007-08-03 04:32:51
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1 message has been excluded from this view by a project administrator.
