Hello,
I want use SIP with Openfire Plugin, but i have a problem and i not understand where is the problem.
I copy the /var/log/info.log
If you need other information don’t hesitate :
2015.09.15 10:18:25 org.jitsi.jigasi.openfire.JigasiPlugin - CallControlComponent - handleIQSet <iq xmlns="jabber:client" type="set" to="ofmeet-call-control.subdomain.domain.net" id="947:sendIQ" from="admin@subdomain.domain.net/ofmeet5244">
<dial xmlns="urn:xmpp:rayo:1" to="0272XXXXXX" from="reunion@conference.subdomain.domain.net">
<header name="JvbRoomName" value="reunion@conference.subdomain.domain.net"/>
</dial>
</iq>
2015.09.15 10:18:25 org.jitsi.jigasi.openfire.JigasiPlugin - CallControlComponent - Dial
2015.09.15 10:18:25 org.jitsi.jigasi.openfire.JigasiPlugin - Got dial request reunion@conference.subdomain.domain.net -> 0272XXXXXX confId 9ab7a30d8c0ac7f4 callId 14fd0155dd1
2015.09.15 10:18:25 org.jitsi.jigasi.openfire.JigasiPlugin - CallControlComponent - makeCall reunion@conference.subdomain.domain.net 0272XXXXXX 14fd0155dd1
2015.09.15 10:18:25 org.ifsoft.sip.CallSession - CallSession creation subdomain.domain.net
2015.09.15 10:18:25 org.jitsi.jigasi.openfire.JigasiPlugin - CallControlComponent - makeCall looking up +0272XXXXXX
2015.09.15 10:18:25 org.jitsi.jigasi.openfire.JigasiPlugin - CallControlComponent - makeCall with outbound proxy sip:+0272XXXXXX@domainvoip.voip sip:poste1@domainvoip.voip
2015.09.15 10:18:25 org.ifsoft.sip.CallSession - buildSDP v=0
o=JabberGW 3651293905 3651293905 IN IP4 11.22.33.44
s=Jabber Call
c=IN IP4 11.22.33.44
t=0 0
m=audio 5002 RTP/AVP 0 101
a=a:sendrecv
a=a:rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
And in openfire in the meeting page i just enter value in SIP CLIENT REGISTRATION and SIP SERVER REGISTRATION is enable :
Thanks in advance.
Have a nice day.
Regards.