<------------->
Audio is at 10.1.1.7 port 18988
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 41.221.1.13:5061:
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5070;branch=z9hG4bK4d3fc319;rport
From: "Unknown" <sip:[email protected]:5070>;tag=as39bb1ba5
To: <sip:[email protected]:5061>
Contact: <sip:[email protected]:5070>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 02 Feb 2011 10:48:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 253

v=0
o=root 3715 3715 IN IP4 10.1.1.7
s=session
c=IN IP4 10.1.1.7
t=0 0
m=audio 18988 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (NAT) to 41.221.1.13:5061:
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5070;branch=z9hG4bK4d3fc319;rport
From: "Unknown" <sip:[email protected]:5070>;tag=as39bb1ba5
To: <sip:[email protected]:5061>
Contact: <sip:[email protected]:5070>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 02 Feb 2011 10:48:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 253

v=0
o=root 3715 3715 IN IP4 10.1.1.7
s=session
c=IN IP4 10.1.1.7
t=0 0
m=audio 18988 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
OpenGate*CLI>
<--- SIP read from 41.221.1.13:5060 --->
OPTIONS sip:10.211.12.236:5070 SIP/2.0
Via: SIP/2.0/UDP 41.221.1.13:5060;branch=0
From: sip:[email protected];tag=3f87e526
To: sip:10.211.12.236:5070
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Looking for s in presence-inbound (domain 10.211.12.236)

<--- Transmitting (no NAT) to 41.221.1.13:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 41.221.1.13:5060;branch=0;received=41.221.1.13
From: sip:[email protected];tag=3f87e526
To: sip:10.211.12.236:5070;tag=as160a0116
Call-ID: [email protected]
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:10.1.1.7:5070>
Accept: application/sdp
Content-Length: 0