<------------->
--- (0 headers 0 lines) Nat keepalive ---
internal*CLI>
<--- SIP read from 203.166.6.160:5060 --->
INVITE sip:61280642141@121.44.8.82:5060 SIP/2.0
Record-Route: <sip:203.166.6.160;ftag=7062e6f9882f71ba07ba6015b644d39d;lr>
Via: SIP/2.0/UDP 203.166.6.160;branch=z9hG4bK728d.2cc77ca8c8b89410e7c306462dfa63a9.0
Via: SIP/2.0/UDP 203.166.6.160:5061;branch=z9hG4bK6b610698c958de075e4490d1c87890bc;rport=5061
Max-Forwards: 16
From: <sip:0410526452@203.166.6.160>;tag=7062e6f9882f71ba07ba6015b644d39d
To: <sip:61280642141@203.166.6.160>
Call-ID: 63F046B5-11711DD-8221F052-82EB7ED6@202.85.241.123
CSeq: 200 INVITE
Contact: Anonymous <sip:203.166.6.160:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 1676451116-18289117-2183065682-2196471510
h323-conf-id: 1676451116-18289117-2183065682-2196471510
Content-Length: 385
Content-Type: application/sdp
v=0
o=Sippy 145298092 0 IN IP4 203.166.6.160
s=SIP Call
t=0 0
m=audio 16984 RTP/AVP 18 3 4 8 0 2 101 19
c=IN IP4 202.85.241.123
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
<------------->
--- (16 headers 17 lines) ---
Sending to 203.166.6.160 : 5060 (no NAT)
Using INVITE request as basis request - 63F046B5-11711DD-8221F052-82EB7ED6@202.85.241.123
Found peer 'Pennytel-out'
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port 202.85.241.123:16984
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 19
Capabilities: us - 0x0 (nothing), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
<--- Reliably Transmitting (NAT) to 203.166.6.160:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 203.166.6.160;branch=z9hG4bK728d.2cc77ca8c8b89410e7c306462dfa63a9.0;received=203.166.6.160
Via: SIP/2.0/UDP 203.166.6.160:5061;branch=z9hG4bK6b610698c958de075e4490d1c87890bc;rport=5061
From: <sip:0410526452@203.166.6.160>;tag=7062e6f9882f71ba07ba6015b644d39d
To: <sip:61280642141@203.166.6.160>;tag=as68927a69
Call-ID: 63F046B5-11711DD-8221F052-82EB7ED6@202.85.241.123
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0