== Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Audio is at 10.0.0.1 port 16388
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.2.3:5060:
INVITE sip:XXXXXXXXX@10.0.2.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1ff9735f
Max-Forwards: 70
From: "XXXXXXXXX" <sip:XXXXXXXXX@10.0.0.1>;tag=as395c135a
To: <sip:XXXXXXXXX@10.0.2.3>
Contact: <sip:XXXXXXXXX@10.0.0.1>
Call-ID: 7ba395546f434bc45d68e5aa29d2a97e@10.0.0.1
CSeq: 102 INVITE
User-Agent: TRIVENET VoIP Service
Date: Wed, 14 Apr 2010 07:20:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 1950427116 1950427116 IN IP4 10.0.0.1
s=Asterisk PBX 1.6.0.22
c=IN IP4 10.0.0.1
t=0 0
m=audio 16388 RTP/AVP 18 8 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called XXXXXXXXX/XXXXXXXXX
SVA01*CLI>
<--- SIP read from UDP://10.0.2.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1ff9735f;received=10.0.0.1
From: "XXXXXXXXX" <sip:XXXXXXXXX@10.0.0.1>;tag=as395c135a
To: <sip:XXXXXXXXX@10.0.2.3>
Call-ID: 7ba395546f434bc45d68e5aa29d2a97e@10.0.0.1
CSeq: 102 INVITE
User-Agent: wildixgw
X-WildixUniqueid: wildixbox-1271229646.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:XXXXXXXXX@10.0.2.3>
Content-Length: 0


------------->
--- (12 headers 0 lines) ---
SVA01*CLI>
<--- SIP read from UDP://10.0.2.3:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1ff9735f;received=10.0.0.1
From: "XXXXXXXXX" <sip:XXXXXXXXX@10.0.0.1>;tag=as395c135a
To: <sip:XXXXXXXXX@10.0.2.3>;tag=as4b4e4d22
Call-ID: 7ba395546f434bc45d68e5aa29d2a97e@10.0.0.1
CSeq: 102 INVITE
User-Agent: wildixgw
X-WildixUniqueid: wildixbox-1271229646.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:XXXXXXXXX@10.0.2.3>
Content-Type: application/sdp
Content-Length: 202

v=0
o=root 1805 1805 IN IP4 10.0.2.3
s=session
c=IN IP4 10.0.2.3
t=0 0
m=audio 11560 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

<------------->
--- (13 headers 10 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.2.3:11560