hi all ,
in my pjsip usa , i do a secnario like this , UAC call one special number to proxy , and proxy response with 302 code , and response the right number to uac ,
uac on_call_redirected fuction callback will invoved , and in this callback , i return PJSIP_REDIRECT_ACCEPT_REPLACE , then is pjsip log , i got this sentence like below:
-end msg--
11:11:37.478 pjsua_media.c Call 0: cleaning up provisional media, prov_med_cnt=2, med_cnt=0
11:11:37.491 pjsua_core.c RX 606 bytes Response msg 100/INVITE/cseq=4901 (rdata03F7B040) from TCP 192.168.17.151:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.23.138:4249;rport;branch=z9hG4bKPjdee861cf80f042c2890ed013a08bf8b7;alias
Record-Route: <sip:192.168.17.151:5060;transport=tcp;lr>
From: "88138" <sip:88138@192.168.17.151>;tag=f40119ffcd5d4854b71bbd2c4fab49de
To: <sip:3100 at 172.16.200.24>
Call-ID: 8aae43d842e141c4abf0acc523807953
CSeq: 4901 INVITE
Contact: <sip:3100 at 172.16.200.24:5060>
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: OPTIONS
Allow: BYE
Allow: REFER
Allow: SUBSCRIBE
Allow: NOTIFY
Allow: INFO
Allow: PUBLISH
Server: Asterisk PBX 1.8.7.0
Supported: replaces
Content-Length: 0
and the red parts will cause a assert failed in the next sip call , so could anyone know about this , and how to fix it , thanks a lot !!
Best regards.
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