[asterisk-users] cant get incoming calls in asterisk

[Антон Сацкий]

*friends help me *
*cant get incoming calls in asterisk*
*(when i connect **80081 in softphone ---every thing is ok**)*


*<--- SIP read from UDP:200.152.125.221:5060 <http://200.152.125.221:5060>
--->*
*INVITE sip:80081 at 10.47.10.10:5060 <http://sip:80081@10.47.10.10:5060>
SIP/2.0*
*Record-Route: <sip:200.152.125.221;lr;ftag=as6872d065>*
*Via: SIP/2.0/UDP 200.152.125.221;branch=z9hG4bKd4fd.b3489837.0*
*Via: SIP/2.0/UDP 200.152.125.213:5060;branch=z9hG4bK2214551d;rport=5060*
*From: "8008 <<< 21982008200" <sip:111111 at ser.sipcode.com.br
<sip%3A111111 at ser.sipcode.com.br>>;tag=as6872d065*
*To: <sip:80081 at ser.sipcode.com.br <sip%3A80081 at ser.sipcode.com.br>>*
*Contact: <sip:111111 at 200.152.125.213 <sip%3A111111 at 200.152.125.213>>*
*Call-ID: 5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br
<5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br>*
*CSeq: 105 INVITE*
*User-Agent: FPBX-2.9.0(1.4.41)*
*Max-Forwards: 69*
*Date: Fri, 06 Mar 2015 18:17:21 GMT*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO*
*Supported: replaces*
*Content-Type: application/sdp*
*Content-Length: 338*

*v=0*
*o=root 3211 3214 IN IP4 200.152.125.213*
*s=session*
*c=IN IP4 200.152.125.213*
*t=0 0*
*m=audio 14686 RTP/AVP 0 8 3 18 101*
*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:101 telephone-event/8000*
*a=fmtp:101 0-16*
*a=silenceSupp:off - - - -*
*a=ptime:20*
*a=sendrecv*
*<------------->*
*--- (16 headers 16 lines) ---*
*Sending to 200.152.125.221:5060 <http://200.152.125.221:5060> (no NAT)*
*Using INVITE request as basis request -
5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br
<5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br>*
*Found peer '111111' for '111111' from 200.152.125.221:5060
<http://200.152.125.221:5060>*

*<--- Reliably Transmitting (no NAT) to 200.152.125.221:5060
<http://200.152.125.221:5060> --->*
*SIP/2.0 401 Unauthorized*
*Via: SIP/2.0/UDP
200.152.125.221;branch=z9hG4bKd4fd.b3489837.0;received=200.152.125.221*
*Via: SIP/2.0/UDP 200.152.125.213:5060;branch=z9hG4bK2214551d;rport=5060*
*From: "8008 <<< 21982008200" <sip:111111 at ser.sipcode.com.br
<sip%3A111111 at ser.sipcode.com.br>>;tag=as6872d065*
*To: <sip:80081 at ser.sipcode.com.br
<sip%3A80081 at ser.sipcode.com.br>>;tag=as09849411*
*Call-ID: 5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br
<5c385d117f894c0f2dd79a3f2129b8f5 at ser.sipcode.com.br>*
*CSeq: 105 INVITE*
*Server: FPBX-12.0.42(11.14.1)*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE*
*Supported: replaces, timer*
*WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63fdf36b"*
*Content-Length: 0*


*<------------>*






-- 
Best regards
Antony
моб (066) 919-75-33
моб (063) 656-43-40
satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com>
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