I paste response but anyway its something wrong here.
hold:
pjsua_call.c !Putting call 0 on hold
pjsua_core.c ....TX 918 bytes Request msg INVITE/cseq=3 (tdta0x7a0a5318)
to UDP xx.x.x.xx:5062:
INVITE sip:xx.x.x.xx:5062 SIP/2.0
Via: SIP/2.0/UDP 192.0.0.4:6000
;rport;branch=z9hG4bKPjaVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
Max-Forwards: 70
From: sip:4xxxxxxxxxxx@xxxxxxxxxxxx
;tag=aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
To: sip:+4yyyyyyyyyyy at neofon.tp.pl
;tag=52560051-14265979364676-gm-po-lucentPCSF-064891
Contact: <sip:4xxxxxxxxxxx at 192.0.0.4:6000>
Call-ID: aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
CSeq: 3 INVITE
Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
User-Agent: PJSIP/2.3-svn
Content-Type: application/sdp
Content-Length: 313
v=0
o=- 3635586736 3635586737 IN IP4 192.0.0.4
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 9 8 0 101
c=IN IP4 192.0.0.4
b=TIAS:64000
a=rtcp:4001 IN IP4 192.0.0.4
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendonly
--end msg--
reinvite:
pjsua_call.c !Sending re-INVITE on call 0
pjsua_core.c ....TX 918 bytes Request msg INVITE/cseq=4 (tdta0x7a0b8638)
to UDP xx.x.x.xx:5062:
INVITE sip:xx.x.x.xx:5062 SIP/2.0
Via: SIP/2.0/UDP 192.0.0.4:6000
;rport;branch=z9hG4bKPjaVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
Max-Forwards: 70
From: sip:4xxxxxxxxxxx@xxxxxxxxxxxx
;tag=aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
To: sip:+4yyyyyyyyyyy at neofon.tp.pl
;tag=52560051-14265979364676-gm-po-lucentPCSF-064891
Contact: <sip:4xxxxxxxxxxx at 192.0.0.4:6000>
Call-ID: aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
CSeq: 4 INVITE
Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
User-Agent: PJSIP/2.3-svn
Content-Type: application/sdp
Content-Length: 313
v=0
o=- 3635586736 3635586738 IN IP4 192.0.0.4
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 9 8 0 101
c=IN IP4 192.0.0.4
b=TIAS:64000
a=rtcp:4001 IN IP4 192.0.0.4
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=inactive
--end msg--
it should be sendrecv instead inactive?
2015-03-17 12:43 GMT+01:00 frogersik <frogersik at gmail.com>:
> hi, if I use call.reinvite with any param, messege will be send but audio
> will be always in inactive state. here is log from app:
> Response msg 200/INVITE/cseq=4 (rdata0x7a71a424) from UDP xx.xx.xx.xx:5062:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> kk.kk.kk.kk:6000;received=212.160.230.3;branch=z9hG4bKPj401k1aYWjzp6ATDoPyrkt2P3fds6BMN6O0C-PWjoO;rport=6000
> From: sip:48xxxxxxxxx@xxxxxxxxx
> ;tag=401k1aYWjzp6ATDoPyrkt2P3fds6BMN6O0C
> To: sip:+48xxxxxxxxx at server.pl
> ;tag=536caa4d-1426592075550893-gm-po-lucentPCSF-034732
> Call-ID: 401k1aYWjzp6ATDoPyrkt2P3fds6BMN6O0C
> CSeq: 4 INVITE
> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
> Supported:
> Accept:
> application/dtmf-relay,application/media_control+xml,application/sdp
> Contact: <sip:XX.XX.xx.xx:5062>
> Content-Type: application/sdp
> Content-Length: 253
> Server: Alcatel-Lucent-HPSS/3.0.3
> v=0
> o=xxxxxxx.pl
> s=-
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 16778 RTP/AVP 9 101
> c=IN IP4 0.0.0.0
> b=TIAS:64000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=inactive
>
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