SIP Debugging Enabled for IP: 62.41.i.j:5060
Audio is at 80.92.a.b port 11894
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.41.i.j:5060:
INVITE sip:00352691600efg@62.41.i.j SIP/2.0
Via: SIP/2.0/UDP 80.92.a.b:5060;branch=z9hG4bK7ca48906;rport
From: "Marc" <sip:3522030stuv@voip.domain.com>;tag=as7fa31488
To: <sip:00352691600efg@62.41.i.j>
Contact: <sip:3522030stuv@80.92.a.b>
Call-ID: 1441e45003559a871af4e7f165e4d1d4@voip.domain.com
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Wed, 08 Nov 2006 18:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 30698 30698 IN IP4 80.92.a.b
s=session
c=IN IP4 80.92.a.b
t=0 0
m=audio 11894 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Found RTP audio format 8
Found RTP audio format 100
Peer audio RTP is at port 62.41.66.203:17500
Found description format PCMA for ID 8
Found description format X-NSE for ID 100
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 62.41.66.203:17500
Reliably Transmitting (no NAT) to 62.41.i.j:5060:
OPTIONS sip:62.41.i.j SIP/2.0
Via: SIP/2.0/UDP 80.92.a.b:5060;branch=z9hG4bK3516b558;rport
From: "Anonymous" <sip:Anonymous@voip.domain.com>;tag=as79354526
To: <sip:62.41.i.j>
Contact: <sip:Anonymous@80.92.a.b>
Call-ID: 00ded4880dd48e425fe875503916615c@voip.domain.com
CSeq: 102 OPTIONS
User-Agent: Asterisk
Max-Forwards: 70
Date: Wed, 08 Nov 2006 18:37:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '00ded4880dd48e425fe875503916615c@voip.domain.com' Method: OPTIONS
Scheduling destruction of SIP dialog '1441e45003559a871af4e7f165e4d1d4@voip.domain.com' in 6400 ms (Method: INVITE)
Really destroying SIP dialog '1441e45003559a871af4e7f165e4d1d4@voip.domain.com' Method: INVITE