--- (22 headers 14 lines) --- Using INVITE request as basis request - 587F7052-921C11DC-8073FB35-7AF324D3@192.168.50.68 Sending to 192.168.50.66 : 5060 (non-NAT) Found peer 'Compass-David' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 125 Found RTP audio format 101 Peer audio RTP is at port 192.168.50.68:17426 Peer video RTP is at port 192.168.50.68:65535 Found description format G729 Found description format PCMA Found description format PCMU Found description format X-CCD Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 099651031 in default (domain 192.168.28.2) list_route: hop: list_route: hop: Transmitting (no NAT) to 192.168.50.66:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.66:5060;branch=z9hG4bK09ad02804cfd6d676651142ffca14b7e.8b7929fb;received=192.168.50.66 Via: SIP/2.0/UDP 192.168.50.68:5060;x-route-tag="cid:AVAYA-AK@192.168.50.68";branch=z9hG4bKBA921785 From: ;tag=5340A87C-52C To: Call-ID: 587F7052-921C11DC-8073FB35-7AF324D3@192.168.50.68 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 -- -- Executing Goto("SIP/6499651027-0820c1d0", "main|s|1") in new stack -- Goto (main,s,1) -- Executing Ringing("SIP/6499651027-0820c1d0", "") in new stack Transmitting (no NAT) to 192.168.50.66:5060: SIP/2.0 180 Ringing ---------- working call ---------- --- (22 headers 14 lines) --- Using INVITE request as basis request - 26D023E8-922311DC-82E7FB35-7AF324D3@192.168.50.68 Sending to 192.168.50.66 : 5060 (non-NAT) Found peer 'Compass-David' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 125 Found RTP audio format 101 Peer audio RTP is at port 192.168.50.68:18626 Peer video RTP is at port 192.168.50.68:65535 Found description format G729 Found description format PCMA Found description format PCMU Found description format X-CCD Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 801 in default (domain 192.168.28.2) list_route: hop: list_route: hop: Transmitting (NAT) to 192.168.50.66:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.66:5060;branch=z9hG4bK966eaa5db187ab36dbc28aad010d6797.3b095ae9;received=192.168.50.66 Via: SIP/2.0/UDP 192.168.50.68:5060;x-route-tag="cid:AVAYA-AK@192.168.50.68";branch=z9hG4bKBAF9972 From: ;tag=536D42C4-C27 To: Call-ID: 26D023E8-922311DC-82E7FB35-7AF324D3@192.168.50.68 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/6499651027-081f6388", "check-state|901") in new stack -- Executing RealTime("SIP/6499651027-081f6388", "example|exten|901|var_") in new stack -- Executing NoOp("SIP/6499651027-081f6388", "1") in new stack -- Executing GotoIf("SIP/6499651027-081f6388", "1?20") in new stack -- Goto (macro-check-state,s,20) -- Executing Set("SIP/6499651027-081f6388", "XTEN=801") in new stack -- Executing NoOp("SIP/6499651027-081f6388", "801 is available.") in new stack -- Executing Macro("SIP/6499651027-081f6388", "dialuser|SIP/davidkelly") in new stack -- Executing Set("SIP/6499651027-081f6388", "USER=davidkelly") in new stack -- Executing NoOp("SIP/6499651027-081f6388", "Called dial user") in new stack -- Executing Dial("SIP/6499651027-081f6388", "SIP/davidkelly|15|rtM(csr_log|6421434103)") in new stack We're at 192.168.1.149 port 11932