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[asterisk-users] cant get incoming calls in asterisk


 
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satskiy.a at gmail.com
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PostPosted: Fri Mar 06, 2015 2:32 pm    Post subject: [asterisk-users] cant get incoming calls in asterisk Reply with quote

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 --->
INVITE 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@ser.sipcode.com.br ([email]sip%3A111111@ser.sipcode.com.br[/email])>;tag=as6872d065
To: <sip:80081@ser.sipcode.com.br ([email]sip%3A80081@ser.sipcode.com.br[/email])>
Contact: <sip:111111@200.152.125.213 ([email]sip%3A111111@200.152.125.213[/email])>
Call-ID: 5c385d117f894c0f2dd79a3f2129b8f5@ser.sipcode.com.br (5c385d117f894c0f2dd79a3f2129b8f5@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 (no NAT)
Using INVITE request as basis request - 5c385d117f894c0f2dd79a3f2129b8f5@ser.sipcode.com.br (5c385d117f894c0f2dd79a3f2129b8f5@ser.sipcode.com.br)
Found peer '111111' for '111111' from 200.152.125.221:5060
 
<--- Reliably Transmitting (no NAT) to 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@ser.sipcode.com.br ([email]sip%3A111111@ser.sipcode.com.br[/email])>;tag=as6872d065
To: <sip:80081@ser.sipcode.com.br ([email]sip%3A80081@ser.sipcode.com.br[/email])>;tag=as09849411
Call-ID: 5c385d117f894c0f2dd79a3f2129b8f5@ser.sipcode.com.br (5c385d117f894c0f2dd79a3f2129b8f5@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@gmail.com ([email]mail%3Asatskiy.a@gmail.com[/email])
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