Why am I getting an immediate BYE when initiating SIP? Here’s the conversation:
[code]INVITE sip:0@192.168.202.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.202:5060;branch=z9hG4bK5da478e0;rport
From: “19783384612” sip:1978338461219783384612@192.168.202.202;tag=as1deab001
To: sip:0@192.168.202.10
Contact: sip:1978338461219783384612@192.168.202.202
Call-ID: 02757bd44dbd59486f3749040220cf52@192.168.202.202
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 03 Aug 2010 23:06:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 31086 31086 IN IP4 192.168.202.202
s=session
c=IN IP4 192.168.202.202
t=0 0
m=audio 14594 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv[/code]
I respond with
SIP/2.0 100 TRYING
Via: SIP/2.0/UDP 192.168.202.202:5060;branch=z9hG4bK5da478e0;rport
From: "19783384612" <sip:1978338461219783384612@192.168.202.202>;tag=as1deab001
To: <sip:0@192.168.202.10>
Date: Tue, 03 Aug 2010 23:06:07 GMT
Call-ID: 02757bd44dbd59486f3749040220cf52@192.168.202.202
CSeq: 102 INVITE
Content-Length: 0
and then
[code]SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.202.202:5060;branch=z9hG4bK5da478e0;rport
From: “19783384612” sip:1978338461219783384612@192.168.202.202;tag=as1deab001
To: sip:0@192.168.202.10
Date: Tue, 03 Aug 2010 23:06:07 GMT
Call-ID: 02757bd44dbd59486f3749040220cf52@192.168.202.202
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 31086 31086 IN IP4 192.168.202.10
s=session
c=IN IP4 192.168.202.10
t=0 0
m=audio 40000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv[/code]
I get back
ACK sip:0@192.168.202.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.202:5060;branch=z9hG4bK4b1079a8;rport
From: "19783384612" <sip:1978338461219783384612@192.168.202.202>;tag=as1deab001
To: <sip:0@192.168.202.10>
Contact: <sip:1978338461219783384612@192.168.202.202>
Call-ID: 02757bd44dbd59486f3749040220cf52@192.168.202.202
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
and then immediately (15 ms later) I get
BYE sip:0@192.168.202.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.202:5060;branch=z9hG4bK6691ff09;rport
From: "19783384612" <sip:1978338461219783384612@192.168.202.202>;tag=as1deab001
To: <sip:0@192.168.202.10>
Call-ID: 02757bd44dbd59486f3749040220cf52@192.168.202.202
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
I have to ignore this BYE to continue my call and then respond to the second BYE when the call actually terminates.
Why am I getting this first BYE? Any help would be greatly appreciated.