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Sip "100 try" instead of "180 ringing"

I am developing software using Microsoft Unified Communications and C #. I am using IMVoipSample as the code base. As a voip server, I use an asterisk. Everything is in order, I can register, make calls, accept / reject incoming calls. There is one thing that I cannot decide.

while I call a third-party softphone, there is an answer:

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.250.29:5060;branch=z9hG4bK786d156c;rport=5060
Contact: <sip:6011@192.168.246.203:45134;rinstance=7af05ded7e7e49e6>
To: <sip:6011@192.168.246.203:45134;rinstance=7af05ded7e7e49e6>;tag=9a00d038
From: "6012"<sip:6012@192.168.250.29>;tag=as66995cd4
Call-ID: 7cebe5d1060b11452571a22e0e2cb919@192.168.250.29
CSeq: 102 INVITE
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 0

But when I call my IMVoipSample mobile phone, there is aswer:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.246.203:45134;branch=z9hG4bK-d87543-71570d1c6127bc7a-1--d87543-;received=192.168.246.203;rport=45134
From: "6011"<sip:6011@192.168.250.29>;tag=18345648
To: "6012"<sip:6012@192.168.250.29>
Call-ID: fd7f305d6520cd53YjQ2ZDJmMDAwZDE0YmUwMjRlMGFmM2NmODg5OGM1ODQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.9-2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:6012@192.168.250.29>
Content-Length: 0

samsung officeerv pbx, , -. IMVoipSample - , . , - "sip 100 try" "180 rinning". , : ?

+3
2

1xx , 100 Trying . SIP (UAS) 100 Trying , INVITE, User Agent (UAC), . 100 Trying response 180 Ringing 183 Session Progress. - - , 2xx, 200 Ok.

100 , 180 , , . , , , 4xx.

+8

, sip.conf

exten => _6.,1,Ringing
exten => _6.,n,Dial(SIP/${EXTEN:0}).

@sipwiz

0

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