<-------------> --- (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: 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";tag=a000001-13c4-970-24fb54-382b To: 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: Content-Length: 0