[ATLSIP] How to generate "180 Ringing" with ATLSIP

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I'm using OSSPhone to test ATLSIP.  It detects incoming calls, but it doesn't send out "180 Ringing" packets so caller doesn't get rings, and proxy eventually routes call to voicemail.

Is there a way to automatically or manually send the "180 Ringing" messages with ATLSIP?

Here is a log from OSSPhone:
{quote}----------------26:34:32.350----------------
Querying STUN server at stun01.sipphone.com.
This may take a while ... 

----------------26:34:42.772----------------
STUN server "stun01.sipphone.com" replies Cone NAT, external IP 72.240.235.122
----------------26:34:47.820----------------
*** LISTENER STARTED *** [OPAL] 127.0.0.1:5060
----------------26:34:47.880----------------
*** LISTENER STARTED *** [OPAL] 192.168.0.20:5060 [*** DEFAULT LISTENER ***]
----------------26:34:47.936----------------
>>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP  SRC: 72.240.235.122:22013 enc=0 bytes=629
REGISTER sip:proxy01.sipphone.com SIP/2.0
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport
CSeq: 1 REGISTER
Call-ID: c3c...@pr...
Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp>
User-Agent: OpenSIPStack v1.1.7-24
Expires: 3600
Max-Forwards: 70
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK
Content-Length: 0

----------------26:34:48.069----------------
<<< SIP/2.0 401 Unauthorized  SRC: 198.65.166.131:5060:UDP enc=0 bytes=548
SIP/2.0 401 Unauthorized
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.76c8
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013
CSeq: 1 REGISTER
Call-ID: c3c...@pr...
WWW-Authenticate: Digest realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1"
Content-Length: 0

----------------26:34:48.125----------------
>>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP  SRC: 72.240.235.122:22013 enc=0 bytes=858
REGISTER sip:proxy01.sipphone.com SIP/2.0
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport
CSeq: 2 REGISTER
Call-ID: c3c...@pr...
Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp>
User-Agent: OpenSIPStack v1.1.7-24
Expires: 3600
Max-Forwards: 70
Authorization: Digest username="recipient", realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1", uri="sip:proxy01.sipphone.com", response="4a78a8e0712b92b26d17ba786a89a4eb", opaque="", algorithm=MD5
Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK
Content-Length: 0

----------------26:34:48.226----------------
<<< SIP/2.0 200 OK  SRC: 198.65.166.131:5060:UDP enc=0 bytes=509
SIP/2.0 200 OK
From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13
To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.90ab
Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013
CSeq: 2 REGISTER
Call-ID: c3c...@pr...
Contact: <sip:recipient@72.240.235.122:22013;transport=udp>;expires=3600
Content-Length: 0

----------------26:34:58.666----------------
<<< INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0  SRC: 198.65.166.131:5060:UDP enc=0 bytes=1103
INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To:  <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Contact: <sip:caller@66.54.140.46>
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Date: Wed, 18 Jun 2008 13:47:42 GMT
User-Agent: Asterisk PBX
Max-Forwards: 16
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
RemoteIP: 66.54.140.46
Content-Type: application/sdp
Content-Length: 379

v=0
o=root 16325 16325 IN IP4 66.54.140.46
s=session
c=IN IP4 66.54.140.46
t=0 0
m=audio 14868 RTP/AVP 0 8 3 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
----------------26:34:58.684----------------
>>> SIP/2.0 100 Trying DST: 198.65.166.131:5060:UDP  SRC: 72.240.235.122:22013 enc=0 bytes=411
SIP/2.0 100 Trying
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To:  <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Content-Length: 0

----------------26:35:24.111----------------
<<< CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0  SRC: 198.65.166.131:5060:UDP enc=0 bytes=332
CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To:  <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
CSeq: 102 CANCEL
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Content-Length: 0

----------------26:35:24.151----------------
>>> SIP/2.0 200 OK DST: 198.65.166.131:5060:UDP  SRC: 72.240.235.122:22013 enc=0 bytes=358
SIP/2.0 200 OK
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To:  <sip:rec...@pr...>
Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131
CSeq: 102 CANCEL
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Server: OpenSIPStack v1.1.7-24
Content-Length: 0

----------------26:35:24.163----------------
>>> SIP/2.0 487 Request Cancelled DST: 198.65.166.131:5060:UDP  SRC: 72.240.235.122:22013 enc=0 bytes=535
SIP/2.0 487 Request Cancelled
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To:  <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13
Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060
CSeq: 102 INVITE
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr>
Server: OpenSIPStack v1.1.7-24
Content-Length: 0

----------------26:35:24.263----------------
<<< ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0  SRC: 198.65.166.131:5060:UDP enc=0 bytes=364
ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0
From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c
To:  <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0
CSeq: 102 ACK
Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46
Content-Length: 0{quote}

Recipient OSSPhone was built using latest code in CVS on 2008-06-13.  I tested by calling in through an ipKall phone number.  No rings were heard from the caller phone system.  Eventually the call was routed to voicemail.

Finest regards,
Bill Root