SIP DEBUG on old version of asterisk (1.0) which accepts the call fine:

    -- Executing Answer("SIP/66.243.113.18-0995b678", "") in new stack
We're at 10.10.10.10 port 15608
Answering with preferred capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.113.96.116;branch=z9hG4bK18a7.528ad074.0
Via: SIP/2.0/UDP 66.243.109.99:5060;branch=999972d7d19c9cb5c0d988ebb98b652e
Record-Route: <sip:38.113.96.116:5060;lr>
From: <sip:7777777777@66.243.113.18>;tag=840D3E78-26B1
To: 5555555555 <sip:5555555555@66.243.109.99:5060>;tag=as670b772a
Call-ID: 574602-3341254773-500340@66.243.109.99
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5555555555@10.10.10.10>
Content-Type: application/sdp
Content-Length: 218

=========================================================

SIP DEBUG on new version of asterisk (1.2.0) which does not seem to send an answer:

    -- Executing Answer("SIP/66.243.113.18-bef09df0", "") in new stack
We're at 10.10.10.10 port 12158
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 38.113.96.116:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.113.96.116;branch=z9hG4bK6efa.17cdc4b7.0;received=38.113.96.116
Via: SIP/2.0/UDP 66.243.109.99:5060;branch=fcd9cef123b0d98c04b8d9488712c616
Record-Route: <sip:38.113.96.116:5060;lr>
From: <sip:7777777777@66.243.113.18>;tag=84068F98-2424
To: 5555555555 <sip:5555555555@66.243.109.99:5060>;tag=as4508fa80
Call-ID: 563310-3341254335-516518@66.243.109.99
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:5555555555@10.10.10.10>
Content-Type: application/sdp
Content-Length: 218