<-- SIP read from 10.131.2.1:5060: INVITE sip:*2@10.131.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacMOVCIHw Max-Forwards: 70 From: ;tag=1c594231441 To: Call-ID: 532314835OUbZ@10.131.2.1 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Type: application/sdp Content-Length: 242 v=0 o=AudiocodesGW 775665 222110 IN IP4 10.131.2.1 s=Phone-Call c=IN IP4 10.131.2.1 t=0 0 m=audio 6000 RTP/AVP 8 0 96 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv --- (13 headers 12 lines)--- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:3101 find_call: **** Creating new SIP dialog for incoming request 532314835OUbZ@10.131.2.1 Using INVITE request as basis request - 532314835OUbZ@10.131.2.1 Sending to 10.131.2.1 : 5060 (non-NAT) Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6917 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 10.131.2.1:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacMOVCIHw;received=10.131.2.1 From: ;tag=1c594231441 To: ;tag=as78d40c02 Call-ID: 532314835OUbZ@10.131.2.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="02a3afc1" Content-Length: 0 --- Scheduling destruction of call '532314835OUbZ@10.131.2.1' in 15000 ms Found user '070001' localhost*CLI> <-- SIP read from 10.131.2.1:5060: ACK sip:*2@10.131.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacMOVCIHw Max-Forwards: 70 From: ;tag=1c594231441 To: ;tag=as78d40c02 Call-ID: 532314835OUbZ@10.131.2.1 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Length: 0 --- (12 headers 0 lines)--- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:3094 find_call: **** Found existing SIP dialog for incoming request 532314835OUbZ@10.131.2.1 Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:1347 __sip_ack: Stopping retransmission on '532314835OUbZ@10.131.2.1' of Response 1: Match Found localhost*CLI> <-- SIP read from 10.131.2.1:5060: INVITE sip:*2@10.131.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacppvcjhW Max-Forwards: 70 From: ;tag=1c594231441 To: Call-ID: 532314835OUbZ@10.131.2.1 CSeq: 2 INVITE Proxy-Authorization: Digest username="070001",realm="asterisk", nonce="02a3afc1" ",uri="sip:*2@10.131.0.1",algorithm=MD5,response="9cea14946198c51398606b416d28b1f0" Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Type: application/sdp Content-Length: 242 v=0 o=AudiocodesGW 775665 222110 IN IP4 10.131.2.1 s=Phone-Call c=IN IP4 10.131.2.1 t=0 0 m=audio 6000 RTP/AVP 8 0 96 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv --- (14 headers 12 lines)--- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:3094 find_call: **** Found existing SIP dialog for incoming request 532314835OUbZ@10.131.2.1 Using INVITE request as basis request - 532314835OUbZ@10.131.2.1 Sending to 10.131.2.1 : 5060 (non-NAT) Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6917 check_user_full: Setting NAT on RTP to 0 Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6096 check_auth: ********** Testar: -Digest username="070001",realm="asterisk",nonce="02a3afc1" ",uri="sip:*2@10.131.0.1",algorithm=MD5,response="9cea14946198c51398606b416d28b1f0"- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6096 check_auth: ********** Testar: -realm="asterisk",nonce="02a3afc1" ",uri="sip:*2@10.131.0.1",algorithm=MD5,response="9cea14946198c51398606b416d28b1f0"- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6096 check_auth: ********** Testar: -nonce="02a3afc1" ",uri="sip:*2@10.131.0.1",algorithm=MD5,response="9cea14946198c51398606b416d28b1f0"- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6138 check_auth: ********** Nonce fas 1: -- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6143 check_auth: ********** Nonce fas 2a: -02a3afc1- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:6150 check_auth: ********** Nonce fas 3: -02a3afc1- Found user '070001' Oct 5 09:04:08 NOTICE[2599]: chan_sip.c:10164 handle_request_invite: Failed to authenticate user ;tag=1c594231441 Reliably Transmitting (no NAT) to 10.131.2.1:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacppvcjhW;received=10.131.2.1 From: ;tag=1c594231441 To: ;tag=as78d40c02 Call-ID: 532314835OUbZ@10.131.2.1 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- localhost*CLI> <-- SIP read from 10.131.2.1:5060: ACK sip:*2@10.131.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacppvcjhW Max-Forwards: 70 From: ;tag=1c594231441 To: ;tag=as78d40c02 Call-ID: 532314835OUbZ@10.131.2.1 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Length: 0 --- (12 headers 0 lines)--- Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:3094 find_call: **** Found existing SIP dialog for incoming request 532314835OUbZ@10.131.2.1 Oct 5 09:04:08 DEBUG[2599]: chan_sip.c:1347 __sip_ack: Stopping retransmission on '532314835OUbZ@10.131.2.1' of Response 2: Match Found Destroying call '532314835OUbZ@10.131.2.1'