Hi all, I am testing with pjsip + CsipSImple with an asterisk pbx. During the tests I found a problem that appears when I call from a terminal A yo terminal B, but I end the call from terminal B (call receiver) Under such conditions terminal B hangout but terminal A are still assuming that call continues. The log I extracted from terminal A is the following. Any Idea how to fix it? Thanks Moises ---------------------------------------------- E/libpjsip( 1003): 00:09:45.941 sip_transport. !Error processing 609 bytes packet from UDP x.x.x.x:5060 : PJSIP syntax error exception when parsing '' header on line 9 col 40: E/libpjsip( 1003): BYE sip:Client2@ x.x.x.y ;ob SIP/2.0 E/libpjsip( 1003): Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK73390fc7;rport E/libpjsip( 1003): Max-Forwards: 70 E/libpjsip( 1003): From: <sip:Client1@x.x.x.x>;tag=as3ff28e14 E/libpjsip( 1003): To: <sip:Client2 at x.x.x.x>;tag=9aYx75IetR66FvFZK4mCHwfKiRa5ey8V E/libpjsip( 1003): Call-ID: rxNVU8Rf52vLyX58qFIuDL1U44D7yir9 E/libpjsip( 1003): CSeq: 104 BYE E/libpjsip( 1003): User-Agent: Asterisk PBX 10.12.2 E/libpjsip( 1003): Proxy-Authorization: Digest username=""Cliente2"", realm="asterisk", algorithm=MD5, uri="sip:x.x.x.x", nonce="", response="5cc04e4b63dead2317bd4d3b7d025f17" E/libpjsip( 1003): X-Asterisk-HangupCause: Normal Clearing E/libpjsip( 1003): X-Asterisk-HangupCauseCode: 16 E/libpjsip( 1003): Content-Length: 0 E/libpjsip( 1003): E/libpjsip( 1003): E/libpjsip( 1003): -- end of packet. E/libpjsip( 1003): 00:09:45.943 sip_transport. Error processing 609 bytes packet from UDP x.x.x.x:5060 : PJSIP syntax error exception when parsing '' header on line 9 col 40: E/libpjsip( 1003): BYE sip:Client2 at x.x.x.y:42385;ob SIP/2.0 E/libpjsip( 1003): Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK73390fc7;rport E/libpjsip( 1003): Max-Forwards: 70 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131021/ec5bdd8b/attachment-0001.html>
