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From: Jonas G. <jon...@gm...> - 2007-10-31 16:41:22
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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 17:44:23
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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: Ilian J. C. P. <ip...@so...> - 2007-11-01 02:45:51
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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 > > |
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From: Jonas G. <jon...@gm...> - 2007-11-01 07:30:27
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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 > |
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