Znuff

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--- (10 headers 13 lines) ---
Sending to 10.0.0.1 : 5060 (no NAT)
Using INVITE request as basis request - 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
Found peer 'trunk_1'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 10.0.0.1:10000
Found description format PCMA for ID 8
Found description format telephone-event for ID 97
Found description format PCMU for ID 0
Found description format G729 for ID 18
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.0.1:10000
Looking for 23003 in DID_trunk_1 (domain 10.0.0.2)
list_route: hop: <sip:23001@10.0.0.1:5060;transport=UDP>
voip*CLI>
<--- Transmitting (no NAT) to 10.0.0.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-970-24fb54-3488;received=10.0.0.1
From: "TEST"<sip:23001@10.0.0.1:5060>;tag=a000001-13c4-970-24fb54-382b
To: <sip:23003@10.0.0.2>
Call-ID: 9794353c-a000001-13c4-970-24fb54-3ca5@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0