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I had compiled the code from this link, also I am able to successfully register a account on sip server. But when I make a sip call then it gives a trap error in pjsua_call_media_init. I need help to understand the reason for error mentioned below.

--end msg--
14:51:34.422    pjsua_acc.c  ....SIP outbound status for acc 0 is not active
14:51:34.422    pjsua_acc.c  ....sip:ssssingh@sip.antisip.com: registration success,     status=200 (OK), will re-register in 300 seconds
14:51:34.422    pjsua_acc.c  ....Keep-alive timer started for acc 0, destination:91.121.81.212:5060, interval:15s
2013-07-04 14:51:34.423 GossipExample[1049:4603] Gossip: dispatchRegistrationState(0)
2013-07-04 14:51:46.651 GossipExample[1049:907] Gossip: ringbackWithSoundNamed: /var/mobile/Applications/1B440F28-7F74-46D5-A120-9C0B3C35AD65/GossipExample.app/ringtone.wav
14:51:46.653    pjsua_aud.c !Creating playlist with 1 file(s)..
14:51:46.655 wav_playlist.c  .WAV playlist 'WAV playlist' created: samp.rate=44100, ch=2, bufsize=4KB
14:51:46.657    pjsua_aud.c  .Playlist created, id=0, slot=1
14:51:47.708   pjsua_call.c  Making call with acc #0 to sip:chakrit2@getonsip.com
14:51:47.710    pjsua_aud.c  .Set sound device: capture=-1, playback=-2
14:51:47.711    pjsua_aud.c  ..Opening sound device PCM@16000/1/20ms
14:51:47.712 coreaudio_dev.  ...Using RemoteIO audio unit
14:51:48.013 coreaudio_dev.  ...core audio stream started
14:51:48.021  pjsua_media.c  .Call 0: initializing media..

After the above sequence of events there is a trap error, image is below

enter image description here

Amar
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user2380738
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  • Please guide me guys... – user2380738 Jul 05 '13 at 05:36
  • What I do/did in these situations is to compile pjsip with debugging enabled, that way you can see source files / line numbers when it breaks into the debugger (or from a stack trace). Adding the debugging flags to the compiler did mean I had to change there build scripts tho. You may like to try pjsip by itself and see if you get the same problems. With the information you have given it's going to very hard for anyone to help you. – Shane Powell Mar 01 '14 at 22:12
  • Does the machine you are trying you run this on have a microphone and speaker connected to it? If there is no source or sink PjSIP will fail to make an outbound call. Inbound calls should work though. – Nithish Nov 11 '14 at 07:57

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