Re: [ATLSIP] ATLSIP and .Net
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Re: [ATLSIP] ATLSIP and .Net
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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 > > |
