Hi,

I hope this would be my last question about this SIP subject, I have managed to overcome the last issue I had by asking a friend to help me from a remote computer, I'm able to connect between the computers, but here is the thing, according to all the examples I saw, the Callee should invoke the Ringing response, but in my application case I didn't implement it yet, but I still receive on the Caller UAC a Ringing response, this is the SIP messages that are on the caller end:

Outgoing Request 5:

INVITE sip:[email protected] SIP/2.0
Contact: "Client 310" <sip:[email protected]>
From: "Client 310" <sip:[email protected]>
Max-Forwards: 32
CSeq: 2 INVITE
Call-ID: [email protected]
Allow: INVITE,CANCEL,ACK,BYE,OPTIONS
Content-Type: application/sdp
Proxy-Authorization: Digest username="310",nonce="012afffb",realm="asterisk",uri="sip:[email protected]",algorithm=MD5,response="d19ca5b98450b4be7bd4045edb8a3a2f"
Via: SIP/2.0/UDP hostName.hn:5060
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Content-Length: 257

v=0
o=310 7108915969559970847 7108915969559970847 IN IP4 xxx.xxx.x.xxx
s=-
i=Nu-Art Software - TacB0sS VoIP information
c=IN IP4 xxx.xxx.x.xxx
m=audio 3312 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

Incoming Response 6:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

Incoming Response 7:

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0

Call to: [email protected] is Ringing

Incoming Response 8:

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 27669 27669 IN IP4 yy.yy.yy.yy
s=session
c=IN IP4 yy.yy.yy.yy
t=0 0
m=audio 10914 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Incoming Response 9:

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Content-Length: 0

I do not respond to the invite, that is why all this is happening, but why am I getting a ringing if I'm not the one sending it.

Thanks,

Adam.