pjsua2 crash on incoming call

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Hello,

With libpjproject 2.5, my application crashes on every incoming
call with `../src/pjsip-ua/sip_inv.c:2321: pjsip_inv_answer:
Assertion `inv->last_answer' failed.`

For example the example pjsip-apps/src/samples/pjsua2_demo.cpp;
running mainProg1() from that file (adjusted to connect to my
server) and making a call gives me:

-------------------------------------------------------------------------------
17:28:38.112 pjsua_core.c .RX 934 bytes Request msg
INVITE/cseq=19 (rdata0x7f5778009268) from UDP 192.168.1.1:5060:
INVITE sip:...:5060;ob SIP/2.0 Via: SIP/2.0/UDP
192.168.1.1:5060;branch=z9hG4bKF6C9406CE2B74353 From: "Telefon"
<sip:...>;tag=02C2A2303BAE5A8F To: <sip:...:5060;ob> Call-ID:
67633F78A1E5FE72@192.168.1.1 CSeq: 19 INVITE Contact:
<sip:38CA76357F60C7729329E698EB24B@192.168.1.1> Max-Forwards: 70
Expires: 120 Supported: 100rel,replaces,timer Allow-Events:
telephone-event,refer Allow:
INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp Accept: application/sdp,
multipart/mixed Accept-Encoding: identity Content-Length: 218

v=0
o=user 13656679 13656679 IN IP4 192.168.1.1
s=call
c=IN IP4 192.168.1.1
t=0 0
m=audio 40012 RTP/AVP 8 18 101
a=sendrecv
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:40013

--end msg--
17:28:38.112 pjsua_call.c .Incoming Request msg INVITE/cseq=19
(rdata0x7f5778009268) 17:28:38.112 pjsua_media.c ..Call 0:
initializing media.. 17:28:38.113 pjsua_media.c ...RTP socket
reachable at 192.168.1.8:4000 17:28:38.113 pjsua_media.c ...RTCP
socket reachable at 192.168.1.8:4001 17:28:38.113 pjsua_media.c
...Media index 0 selected for audio call 0
*** Incoming Call: "Telefon" <sip:...> [NULL]
17:28:38.113 pjsua_call.c ...Answering call 0: code=200 a.out:
../src/pjsip-ua/sip_inv.c:2321: pjsip_inv_answer: Assertion
`inv->last_answer' failed. [1] 19792 abort (core dumped) ./a.out
------------------------------------------------------------------------

The call handler ist: 
  
-------------------------------------------------------------------------
virtual void onIncomingCall(OnIncomingCallParam &iprm)
    {
        Call *call = new MyCall(*this, iprm.callId);
        CallInfo ci = call->getInfo();
        CallOpParam prm;
        
        std::cout << "*** Incoming Call: " <<  ci.remoteUri << " ["
                  << ci.stateText << "]" << std::endl;
        
        calls.push_back(call);
        prm.statusCode = (pjsip_status_code)200;
        call->answer(prm);
    }
---------------------------------------------------------------------------

Any idea what to do here?

thank you
Gabriel

Attachment: Encryption key for Gabriel Margiani.asc
Description: application/pgp-keys

Attachment: signature.asc
Description: OpenPGP Digital Signature

_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@xxxxxxxxxxxxxxx
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux