--- (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: <sip:192.168.50.66:5060;maddr=192.168.50.66>
list_route: hop: <sip:6421434103@192.168.50.68:5060>
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: <sip:6421434103@192.168.50.68>;tag=5340A87C-52C
To: <sip:6499651027@192.168.50.66>
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: <sip:099651031@192.168.28.2>
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: <sip:192.168.50.66:5060;maddr=192.168.50.66>
list_route: hop: <sip:6421434103@192.168.50.68:5060>
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: <sip:6421434103@192.168.50.68>;tag=536D42C4-C27
To: <sip:6499651034@192.168.50.66>
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: <sip:801@192.168.28.2>
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