Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined.
Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten => 12345,1,Dial(SIP,giusy*) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk *dhcppc0*CLI> <--- SIP read from 192.168.1.116:14862 ---> INVITE sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gia...@192.168.1.116:14862> To: "12345"<sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100>> From: "gianca"<sip:gia...@192.168.1.100 <sip%3agia...@192.168.1.100> >;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv* *<-------------> --- (12 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.* *<--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862 From: "gianca"<sip:gia...@192.168.1.100 <sip%3agia...@192.168.1.100> >;tag=db428348 To: "12345"<sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> >;tag=as29d2b71c Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42ebb35e" Content-Length: 0* *<------------> Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE) Found user 'gianca' dhcppc0*CLI> <--- SIP read from 192.168.1.116:14862 ---> ACK sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport To: "12345"<sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> >;tag=as29d2b71c From: "gianca"<sip:gia...@192.168.1.100 <sip%3agia...@192.168.1.100> >;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 ACK Content-Length: 0* *<-------------> --- (7 headers 0 lines) --- dhcppc0*CLI> <--- SIP read from 192.168.1.116:14862 ---> INVITE sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gia...@192.168.1.116:14862> To: "12345"<sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100>> From: "gianca"<sip:gia...@192.168.1.100 <sip%3agia...@192.168.1.100> >;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="gianca",realm="asterisk",nonce="42ebb35e",uri=" sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> ",response="8d00b3e1b28ed2e40681a3a9ee410046",algorithm=MD5 User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265* *v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv* *<-------------> --- (13 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. Found user 'gianca' Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.116:5960 Found unknown media description format BV32 for ID 107 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.116:5960 Looking for 12345 in tutorial (domain 192.168.1.100) list_route: hop: <sip:gia...@192.168.1.116:14862>* *<--- Transmitting (no NAT) to 192.168.1.116:14862 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862 From: "gianca"<sip:gia...@192.168.1.100 <sip%3agia...@192.168.1.100> >;tag=db428348 To: "12345"<sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100>> Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100>> Content-Length: 0* *<------------> -- Executing [12...@tutorial:1] Dial("SIP/gianca-088b96e0", "SIP|giusy") in new stack == Spawn extension (tutorial, 12345, 1) exited non-zero on 'SIP/gianca-088b96e0' Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)* *<--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862 From: "gianca"<sip:gia...@192.168.1.100 <sip%3agia...@192.168.1.100> >;tag=db428348 To: "12345"<sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> >;tag=as12cbf532 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0* *<------------> dhcppc0*CLI> <--- SIP read from 192.168.1.116:14862 ---> ACK sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport To: "12345"<sip:12...@192.168.1.100 <sip%3a12...@192.168.1.100> >;tag=as12cbf532 From: "gianca"<sip:gia...@192.168.1.100 <sip%3agia...@192.168.1.100> >;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 ACK Content-Length: 0 * ** -- Giancarlo Lombardo
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