I made a sip user and called it and shift the call to my java program using asterisk-java FastAGI protocol.
The image shows the java class and asterisk command line status while call is made
The error is:
NOTICE: channel.c:4149 __ast_read: Dropping incompatible voice frame on SIP/1000abc-00000000 of format ulaw since our native format has changed to 0x2 (gsm)
[Jun 2 16:04:38] ERROR: utils.c:1164 ast_carefulwrite: write() returned error: Broken pipe