WITH CALLER ID DISABLED pbx1*CLI> pbx1*CLI> <-- SIP read from 10.10.10.39:5060: INVITE sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK1084469646 From: 3000;tag=3815248908 To: Call-ID: 1112494159@10.10.10.39 CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.) Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER Content-Type: application/sdp Content-Length: 195 v=0 o=3000 123456 654321 IN IP4 10.10.10.39 s=A conversation c=IN IP4 10.10.10.39 t=0 0 m=audio 12080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000/1 --- (12 headers 9 lines)--- Using INVITE request as basis request - 1112494159@10.10.10.39 Sending to 10.10.10.39 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.10.10.39:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK1084469646;received=10.10.10.39 From: 3000;tag=3815248908 To: ;tag=as6f5cbdc6 Call-ID: 1112494159@10.10.10.39 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3eb3c174" Content-Length: 0 --- Scheduling destruction of call '1112494159@10.10.10.39' in 15000 ms Found user '3000' pbx1*CLI> <-- SIP read from 10.10.10.39:5060: ACK sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK1084469646 From: 3000;tag=3815248908 To: ;tag=as6f5cbdc6 Call-ID: 1112494159@10.10.10.39 CSeq: 1 ACK Content-Length: 0 --- (7 headers 0 lines)--- pbx1*CLI> <-- SIP read from 10.10.10.39:5060: INVITE sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3334702713 From: 3000;tag=3815248908 To: Call-ID: 1112494159@10.10.10.39 CSeq: 2 INVITE Contact: Proxy-Authorization: Digest username="3000", realm="asterisk", nonce="3eb3c174", uri="sip:s@10.10.10.10", response="43f1c0cfdfe2bd6457f93696da56a557", algorithm=MD5 Max-Forwards: 70 User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.) Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER Content-Type: application/sdp Content-Length: 195 v=0 o=3000 123456 654321 IN IP4 10.10.10.39 s=A conversation c=IN IP4 10.10.10.39 t=0 0 m=audio 12080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000/1 --- (13 headers 9 lines)--- Using INVITE request as basis request - 1112494159@10.10.10.39 Sending to 10.10.10.39 : 5060 (non-NAT) Found user '3000' Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.39:12080 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in inbound (domain 10.10.10.10) list_route: hop: Transmitting (no NAT) to 10.10.10.39:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3334702713;received=10.10.10.39 From: 3000;tag=3815248908 To: Call-ID: 1112494159@10.10.10.39 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Wait("SIP/3000-0815a048", "2") in new stack -- Executing Answer("SIP/3000-0815a048", "") in new stack We're at 10.10.10.10 port 12508 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.10.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3334702713;received=10.10.10.39 From: 3000;tag=3815248908 To: ;tag=as06e5c823 Call-ID: 1112494159@10.10.10.39 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 214 v=0 o=root 32372 32372 IN IP4 10.10.10.10 s=session c=IN IP4 10.10.10.10 t=0 0 m=audio 12508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing NoOp("SIP/3000-0815a048", ""3000" <3000> 3000 3000") in new stack -- Executing BackGround("SIP/3000-0815a048", "726_welcome") in new stack -- Playing '726_welcome' (language 'en') pbx1*CLI> <-- SIP read from 10.10.10.39:5060: ACK sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK398548864 From: 3000;tag=3815248908 To: ;tag=as06e5c823 Call-ID: 1112494159@10.10.10.39 CSeq: 2 ACK Max-Forwards: 70 User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.) Content-Length: 0 --- (9 headers 0 lines)--- pbx1*CLI> <-- SIP read from 10.10.10.39:5060: BYE sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3381027283 From: 3000;tag=3815248908 To: ;tag=as06e5c823 Call-ID: 1112494159@10.10.10.39 CSeq: 3 BYE Max-Forwards: 70 User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.) Content-Length: 0 --- (9 headers 0 lines)--- Sending to 10.10.10.39 : 5060 (non-NAT) Transmitting (no NAT) to 10.10.10.39:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK3381027283;received=10.10.10.39 From: 3000;tag=3815248908 To: ;tag=as06e5c823 Call-ID: 1112494159@10.10.10.39 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- == Spawn extension (inbound, s, 4) exited non-zero on 'SIP/3000-0815a048' -- Executing Hangup("SIP/3000-0815a048", "") in new stack == Spawn extension (inbound, h, 1) exited non-zero on 'SIP/3000-0815a048' Destroying call '1112494159@10.10.10.39' pbx1*CLI> ========================================================================================= ========================================================================================= ========================================================================================= ========================================================================================= ========================================================================================= WITH CALLER ID ENABLED pbx1*CLI> <-- SIP read from 10.10.10.39:5060: INVITE sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK654133672 From: 3000;tag=249097462 To: Call-ID: 1390462977@10.10.10.39 CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.) Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER Content-Type: application/sdp Content-Length: 195 v=0 o=3000 123456 654321 IN IP4 10.10.10.39 s=A conversation c=IN IP4 10.10.10.39 t=0 0 m=audio 12080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000/1 --- (12 headers 9 lines)--- Using INVITE request as basis request - 1390462977@10.10.10.39 Sending to 10.10.10.39 : 5060 (non-NAT) Found peer '3030' Reliably Transmitting (no NAT) to 10.10.10.39:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK654133672;received=10.10.10.39 From: 3000;tag=249097462 To: ;tag=as0b3678cf Call-ID: 1390462977@10.10.10.39 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1aecd999" Content-Length: 0 --- Scheduling destruction of call '1390462977@10.10.10.39' in 15000 ms pbx1*CLI> <-- SIP read from 10.10.10.39:5060: ACK sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK654133672 From: 3000;tag=249097462 To: ;tag=as0b3678cf Call-ID: 1390462977@10.10.10.39 CSeq: 1 ACK Content-Length: 0 --- (7 headers 0 lines)--- pbx1*CLI> <-- SIP read from 10.10.10.39:5060: INVITE sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK2531879195 From: 3000;tag=249097462 To: Call-ID: 1390462977@10.10.10.39 CSeq: 2 INVITE Contact: Proxy-Authorization: Digest username="3000", realm="asterisk", nonce="1aecd999", uri="sip:s@10.10.10.10", response="2c121175fb4b01fd34373e934ade9792", algorithm=MD5 Max-Forwards: 70 User-Agent: Clipcomm CG-410 IP175C v1.2.12 (2005.09.13.) Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,INFO,REGISTER Content-Type: application/sdp Content-Length: 195 v=0 o=3000 123456 654321 IN IP4 10.10.10.39 s=A conversation c=IN IP4 10.10.10.39 t=0 0 m=audio 12080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=ptime:20 a=rtpmap:101 telephone-event/8000/1 --- (13 headers 9 lines)--- Using INVITE request as basis request - 1390462977@10.10.10.39 Sending to 10.10.10.39 : 5060 (non-NAT) Found peer '3030' Nov 3 14:03:00 NOTICE[3415]: chan_sip.c:10468 handle_request_invite: Failed to authenticate user 3000;tag=249097462 Reliably Transmitting (no NAT) to 10.10.10.39:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK2531879195;received=10.10.10.39 From: 3000;tag=249097462 To: ;tag=as0b3678cf Call-ID: 1390462977@10.10.10.39 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- pbx1*CLI> <-- SIP read from 10.10.10.39:5060: ACK sip:s@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.39:5060;branch=z9hG4bK2531879195 From: 3000;tag=249097462 To: ;tag=as0b3678cf Call-ID: 1390462977@10.10.10.39 CSeq: 2 ACK Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '1390462977@10.10.10.39'