*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 at 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 ??? (066) 919-75-33 ??? (063) 656-43-40 satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150306/2181cab3/attachment.html>