why the call disconnected?

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You need check asterisk. BYE was from *.

regards,
Gang

2008/8/12 Liusheng <liu_sx2006 at hotmail.com>

> I developed a PJSUA, when I callsetup with a phone, but when I hear ring
> and pick up the phone, the call was disconnected, please help me check the
> log why it is disconnected?
>
>
>  17:50:39.577     PjCallback Call 0 state=CONFIRMED
>  17:50:40.577   pjsua_core.c RX 340 bytes Request msg BYE/cseq=102
> (rdata0x81af75c) from UDP 192.168.1.129:5060:
> BYE sip:3002 at 192.168.1.179:5061 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.129:5060;branch=z9hG4bK09df26ad;rport
> From: sip:3000@192.168.1.129 <sip%3A3000 at 192.168.1.129>;tag=as0c568aff
> To: sip:3002 at 192.168.1.129 <sip%3A3002 at 192.168.1.129>
> ;tag=MPKqkcMa77.oe47iOWybv8V.iCvdf2ub
> Call-ID: YNWRCJq9nRvDWEJN.5oISSL-LtLz5p0i
> CSeq: 102 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
> --end msg--
>  17:50:40.577   pjsua_core.c TX 304 bytes Response msg 200/BYE/cseq=102
> (tdta0x81cb108) to UDP 192.168.1.129:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.129:5060;rport=5060;received=192.168.1.129
> ;branch=z9hG4bK09df26ad
> Call-ID: YNWRCJq9nRvDWEJN.5oISSL-LtLz5p0i
> From: <sip:3000@192.168.1.129 <sip%3A3000 at 192.168.1.129>>;tag=as0c568aff
> To: <sip:3002 at 192.168.1.129 <sip%3A3002 at 192.168.1.129>
> >;tag=MPKqkcMa77.oe47iOWybv8V.iCvdf2ub
> CSeq: 102 BYE
> Content-Length:  0
>
> --end msg--
>  17:50:40.577     PjCallback Call 0 state=DISCONNCTD
> SSSigGwAgent::Send data: <DISCONNECT>
>  17:50:40.577  pjsua_media.c Media session for call 0 is destroyed
>
>
>
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