rendered paste body--- (10 headers 12 lines)---
Using INVITE request as basis request - 155033a-55@147.135.12.128
Sending to 147.135.12.128 : 5060 (NAT)
Found peer 'BroadVoice'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Peer audio RTP is at port 147.135.12.248:14820
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format G729AB
Found description format iLBC
Capabilities: us - 0x4 (ulaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 3302471150 in ext-did-direct (domain 71.66.110.248)
list_route: hop: <sip:4125284008@147.135.12.128:5060>
Reliably Transmitting (NAT) to 147.135.12.128:5060:
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 147.135.12.128:5060;received=147.135.12.128
From: "Agostine Richar"<sip:4125284008@147.135.12.128;user=phone>;tag=xz02
To: "Rick Bonath"<sip:51224@71.66.110.248;user=phone>;tag=as58a0c41b
Call-ID: 155033a-55@147.135.12.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3302471150@71.66.110.248>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 147.135.12.128:5060:
ACK sip:51224@71.66.110.248:5060 SIP/2.0
Call-ID: 155033a-55@147.135.12.128
CSeq: 1 ACK
From: "Agostine Richar"<sip:4125284008@147.135.12.128;user=phone>;tag=xz02
To: "Rick Bonath"<sip:51224@71.66.110.248;user=phone>;tag=as58a0c41b
Via: SIP/2.0/UDP 147.135.12.128:5060;received=71.66.110.248
Content-Length: 0
--- (7 headers 0 lines)---
Destroying call '155033a-55@147.135.12.128'
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.12.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 71.66.110.248:5060;branch=z9hG4bK64d8d914;rport
From: <sip:3302471150@sip.broadvoice.com>;tag=as72deb3fb
To: <sip:3302471150@sip.broadvoice.com>
Call-ID: 59259b943126fe0134a57eb36b416e5b@127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:51224@71.66.110.248>
Event: registration
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 147.135.12.128:5060:
SIP/2.0 200 OK
Call-ID: 59259b943126fe0134a57eb36b416e5b@127.0.0.1
CSeq: 103 REGISTER
From: <sip:3302471150@sip.broadvoice.com>;tag=as72deb3fb
To: <sip:3302471150@sip.broadvoice.com>
Via: SIP/2.0/UDP 71.66.110.248:5060;branch=z9hG4bK64d8d914
Contact: <sip:51224@71.66.110.248>
Expires: 30
Event: registration
User-Agent: Asterisk PBX
Content-Length: 0
--- (11 headers 0 lines)---
Scheduling destruction of call '59259b943126fe0134a57eb36b416e5b@127.0.0.1' in 32000 ms
asterisk1*CLI>
<-- SIP read from 192.168.1.25:6234:
--- (0 headers 0 lines) Nat keepalive ---
asterisk1*CLI>
[root@asterisk1 etc]#