OfficeSIP Softphone

I am developing the free SIP softphone (audio+video) for Windows. And I have some issues with asterisk 1.6 compatibility. I am new in asterisk, so I guess, I have no enough skills to config asterisk properly. I have enable tcp transport mode and register client, but can not make a call. The server report 491 Request Pending on invite message. Any ideas, help are welcome.

Here is link to the softphone:
officesip.com/download/offic … ne-1.0.msi

INVITE sip:58@trixbox1.local SIP/2.0
Via: SIP/2.0/TCP 192.168.1.15:54265
Max-Forwards: 70
From: ;tag=c049124e8c;epid=7548044289
To:
Call-ID: 942016c18b5b4870b022d5a54906c77c
CSeq: 2 INVITE
Contact: ;proxy=replace;+sip.instance=""
User-Agent: UCCAPI/2.0.6362.67
Supported: timer
Supported: ms-sender
Supported: ms-early-media
Supported: Replaces
ms-keep-alive: UAC;hop-hop=yes
Authorization: Digest username=“56”, realm=“asterisk”, algorithm=MD5, uri=“sip:58@trixbox1.local”, nonce=“7c688121”, response="1e2c268e041d74dd2fb5818ad2c9b73f"
Content-Type: application/sdp
Content-Length: 2143

v=0
o=- 0 0 IN IP4 192.168.1.15
s=session
c=IN IP4 192.168.1.15
b=CT:99980
t=0 0
m=audio 9856 RTP/AVP 114 111 112 115 116 4 8 0 97 101
k=base64:3ggW6lhkS6IykF8RHi4SpP2L8vztgn4UwltgeKXYcv2mQGa3yo1JwJWYNTPe
a=candidate:rjUVSi7/RY8xQWhaMXwoLtvbWy/Wxv1U319JRIkOVwQ 1 UiMTVK5dYYSkNWRkR2PZjQ UDP 0.830 192.168.1.15 9856
a=candidate:rjUVSi7/RY8xQWhaMXwoLtvbWy/Wxv1U319JRIkOVwQ 2 UiMTVK5dYYSkNWRkR2PZjQ UDP 0.830 192.168.1.15 44928
a=candidate:0snbNc9MqMr4jO3vvljjb9IuEcMsDP9uV5AfS7mkxEU 1 yaJgGb/PhWbLyAZTYszL5g UDP 0.840 192.168.56.1 37376
a=candidate:0snbNc9MqMr4jO3vvljjb9IuEcMsDP9uV5AfS7mkxEU 2 yaJgGb/PhWbLyAZTYszL5g UDP 0.840 192.168.56.1 26752
a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:5eShADbM6DRrkNPzONa97ItxI7kt4ShQQJdgaHjB|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:owBhgVeVBnl4SuxwdW4S24QQ9/oEzvTwXwoAE2ud|2^31|1:1
a=maxptime:200
a=rtcp:44928
a=rtpmap:114 x-msrta/16000
a=fmtp:114 bitrate=29000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:116 AAL2-G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=encryption:optional
m=video 9600 RTP/AVP 121 34
k=base64:Sxwwj9nT7Wh3aE5aXsOA0aBYkIY5cNMZkR95T8ZqrDa/7CUtfoQXcW0LOhvg
a=candidate:RCOpKCc80XURX/FArSg3zcavmTVfpcn3ZtHFI4EUSow 1 0nluuAP1uriXuEP5njBpJg UDP 0.850 192.168.1.15 9600
a=candidate:RCOpKCc80XURX/FArSg3zcavmTVfpcn3ZtHFI4EUSow 2 0nluuAP1uriXuEP5njBpJg UDP 0.850 192.168.1.15 38784
a=candidate:YdZD712LJ49q+WrTpIYegZgxqzelhQkiqS+qfF6Tyj0 1 b9//3XJoT2M5AaD7oy4FCQ UDP 0.860 192.168.56.1 6272
a=candidate:YdZD712LJ49q+WrTpIYegZgxqzelhQkiqS+qfF6Tyj0 2 b9//3XJoT2M5AaD7oy4FCQ UDP 0.860 192.168.56.1 49024
a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:PjRCApjdS9szuRiFRVydoJOG6X4ecCO86kWMjvBn|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:SqiaKceHJ5M3QOc0/CetDbsOqzo0i9G3b50NHt+f|2^31|1:1
a=maxptime:200
a=rtcp:38784
a=rtpmap:121 x-rtvc1/90000
a=rtpmap:34 H263/90000
a=encryption:optional

SIP/2.0 491 Request Pending
Via: SIP/2.0/TCP 192.168.1.15:54265;received=192.168.1.15
From: ;tag=c049124e8c;epid=7548044289
To: ;tag=as72ef9e42
Call-ID: 942016c18b5b4870b022d5a54906c77c
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

Not exactly sure why you get the error, may be bad error recovery, but the productions for addr-spec and name-addr are not permitted to be empty, so your INVITE is invalid, both for the From: and To: headers.

The fields are not empty actually, a forum cut them.

Does asterisk support TCP? Or is it undocumented/unused feature?