*CLI> sip debug SIP Debugging Enabled *CLI> Sip read: INVITE sip:9700000004239@asterisk.private.ip.address;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-7s6rqpuohc15 From: "mack" ;tag=plv0ymk595 To: Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 1 INVITE Max-Forwards: 70 Contact: User-Agent: snom100-2.03o Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 283 v=0 o=root 695249894 695249894 IN IP4 192.168.10.251 s=call c=IN IP4 192.168.10.251 t=0 0 m=audio 10144 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=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 16 headers, 13 lines Using latest request as basis request Sending to 192.168.10.251 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.251:10144 Found description format pcmu Found description format pcma Found description format gsm Found description format g729 Found description format telephone-event Capabilities: us - 0x6(GSM|ULAW), peer - audio=0x10e(GSM|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x6(GSM|ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-7s6rqpuohc15 From: "mack" ;tag=plv0ymk595 To: ;tag=as72cbef5d Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5865516e" Content-Length: 0 to 192.168.10.251:5060 Scheduling destruction of call '3c359e091559-id47fwa0oxx2@192-168-10-251' in 15000 ms Found user 'mack' Sip read: ACK sip:9700000004239@asterisk.private.ip.address;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-7s6rqpuohc15 From: "mack" ;tag=plv0ymk595 To: ;tag=as72cbef5d Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 1 ACK Max-Forwards: 70 Contact: Content-Length: 0 9 headers, 0 lines Sip read: INVITE sip:9700000004239@asterisk.private.ip.address;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-0ldkuv1wouk0 From: "mack" ;tag=plv0ymk595 To: Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 2 INVITE Max-Forwards: 70 Contact: User-Agent: snom100-2.03o Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Supported: timer, 100rel, replaces Session-Expires: 3600 Proxy-Authorization: Digest username="mack",realm="asterisk",nonce="5865516e",uri="sip:9700000004239@asterisk.private.ip.address;user=phone",response="c88fcd2b00076577169c fa8ffba948e0",algorithm=md5 Content-Type: application/sdp Content-Length: 283 v=0 o=root 695249894 695249894 IN IP4 192.168.10.251 s=call c=IN IP4 192.168.10.251 t=0 0 m=audio 10144 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=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 17 headers, 13 lines Using latest request as basis request Sending to 192.168.10.251 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.251:10144 Found description format pcmu Found description format pcma Found description format gsm Found description format g729 Found description format telephone-event Capabilities: us - 0x6(GSM|ULAW), peer - audio=0x10e(GSM|ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x6(GSM|ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user 'mack' Looking for 9700000004239 in default list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-0ldkuv1wouk0 From: "mack" ;tag=plv0ymk595 To: ;tag=as16a3d311 Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.10.251:5060 -- Executing SetCallerID("SIP/mack-e0d1", "telconumber") in new stack -- Executing SetCIDName("SIP/mack-e0d1", "telconumber") in new stack -- Executing Dial("SIP/mack-e0d1", "sip/00000004239@aTelcoCompany") in new stack We're at 210.194.200.154 port 18388 Answering/Requesting with root capability 4 12 headers, 8 lines Reliably Transmitting: INVITE sip:00000004239@telco.uas.ip.address SIP/2.0 Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK2252aebc From: "telconumber" ;tag=as76828056 To: Contact: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 17 Nov 2004 04:40:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 5203 5203 IN IP4 210.194.200.154 s=session c=IN IP4 210.194.200.154 t=0 0 m=audio 18388 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - (no NAT) to telco.uas.ip.address:5060 -- Called 00000004239@aTelcoCompany Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK2252aebc From: "telconumber" ;tag=as76828056 To: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 102 INVITE 6 headers, 0 lines Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK2252aebc From: "telconumber" ;tag=as76828056 To: ;tag=SD40e6399-253022465 Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 102 INVITE Content-Length: 0 Server: SMAP-ShortBoard Inc. Proxy-Authenticate: Digest realm="Registered Users",qop="auth",opaque="1091041000331317",nonce="1091041000331317" 9 headers, 0 lines Transmitting: ACK sip:00000004239@telco.uas.ip.address SIP/2.0 Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK2252aebc From: "telconumber" ;tag=as76828056 To: ;tag=SD40e6399-253022465 Contact: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to telco.uas.ip.address:5060 We're at 210.194.200.154 port 18388 Answering/Requesting with root capability 4 Reliably Transmitting: INVITE sip:00000004239@telco.uas.ip.address SIP/2.0 Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK670d0c8f From: "telconumber" ;tag=as76828056 To: Contact: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="telconumber", realm="Registered Users", algorithm=MD5, uri="sip:00000004239@telco.uas.ip.address", nonce="10910410003313 17", response="a5295bd6dafc48593a4c368e2c514cf0", opaque="1091041000331317", qop="auth", cnonce="083ba9d0", nc=00000001 Date: Wed, 17 Nov 2004 04:40:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 5203 5204 IN IP4 210.194.200.154 s=session c=IN IP4 210.194.200.154 t=0 0 m=audio 18388 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - (no NAT) to telco.uas.ip.address:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK670d0c8f From: "telconumber" ;tag=as76828056 To: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 103 INVITE 6 headers, 0 lines Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK670d0c8f From: "telconumber" ;tag=as76828056 To: ;tag=SD40e6399-50388663 Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 103 INVITE Content-Length: 0 Server: SMAP-ShortBoard Inc. Proxy-Authenticate: Digest realm="Registered Users",qop="auth",opaque="1101041001331317",nonce="1101041001331317" 9 headers, 0 lines Transmitting: ACK sip:00000004239@telco.uas.ip.address SIP/2.0 Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK670d0c8f From: "telconumber" ;tag=as76828056 To: ;tag=SD40e6399-50388663 Contact: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to telco.uas.ip.address:5060 We're at 210.194.200.154 port 18388 Answering/Requesting with root capability 4 Reliably Transmitting: INVITE sip:00000004239@telco.uas.ip.address SIP/2.0 Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK08c8e293 From: "telconumber" ;tag=as76828056 To: Contact: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 104 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="telconumber", realm="Registered Users", algorithm=MD5, uri="sip:00000004239@telco.uas.ip.address", nonce="11010410013313 17", response="1148e3da5659b393c8f2288fa5eb1825", opaque="1101041001331317", qop="auth", cnonce="2585a93b", nc=00000001 Date: Wed, 17 Nov 2004 04:40:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 5203 5205 IN IP4 210.194.200.154 s=session c=IN IP4 210.194.200.154 t=0 0 m=audio 18388 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - (no NAT) to telco.uas.ip.address:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK08c8e293 From: "telconumber" ;tag=as76828056 To: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 104 INVITE 6 headers, 0 lines Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK08c8e293 From: "telconumber" ;tag=as76828056 To: ;tag=SD40e6399-131734833 Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 104 INVITE Content-Length: 0 Server: SMAP-ShortBoard Inc. Proxy-Authenticate: Digest realm="Registered Users",qop="auth",opaque="1101041001331317",nonce="1101041001331317" 9 headers, 0 lines Transmitting: ACK sip:00000004239@telco.uas.ip.address SIP/2.0 Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK08c8e293 From: "telconumber" ;tag=as76828056 To: ;tag=SD40e6399-131734833 Contact: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to telco.uas.ip.address:5060 Nov 17 13:40:54 NOTICE[5211]: chan_sip.c:6761 handle_response: Failed to authenticate on INVITE to '"telconumber" ;tag= as76828056' Sip read: CANCEL sip:9700000004239@asterisk.private.ip.address;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-0ldkuv1wouk0 From: "mack" ;tag=plv0ymk595 To: Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 2 CANCEL Max-Forwards: 70 Contact: Content-Length: 0 9 headers, 0 lines Sending to 192.168.10.251 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-0ldkuv1wouk0 From: "mack" ;tag=plv0ymk595 To: ;tag=as16a3d311 Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.10.251:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-0ldkuv1wouk0 From: "mack" ;tag=plv0ymk595 To: ;tag=as16a3d311 Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.10.251:5060 Reliably Transmitting: CANCEL sip:00000004239@telco.uas.ip.address SIP/2.0 Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK08c8e293 From: "telconumber" ;tag=as76828056 To: Contact: Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 104 CANCEL User-Agent: Asterisk PBX Proxy-Authorization: Digest username="telconumber", realm="Registered Users", algorithm=MD5, uri="sip:00000004239@telco.uas.ip.address", nonce="11010410013313 17", response="6009fee0c9199d4fc3868eee48617087", opaque="1101041001331317", qop="auth", cnonce="794c9c7f", nc=00000001 Content-Length: 0 (no NAT) to telco.uas.ip.address:5060 Scheduling destruction of call '367d2f9868330091549e280a6636da67@telco.uas.ip.address' in 15000 ms == Spawn extension (default, 9700000004239, 3) exited non-zero on 'SIP/mack-e0d1' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 210.194.200.154:5060;branch=z9hG4bK08c8e293 From: "telconumber" ;tag=as76828056 To: ;tag=SD40e6399-131734833 Call-ID: 367d2f9868330091549e280a6636da67@telco.uas.ip.address CSeq: 104 CANCEL 6 headers, 0 lines Sip read: ACK sip:9700000004239@asterisk.private.ip.address;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-0ldkuv1wouk0 From: "mack" ;tag=plv0ymk595 To: ;tag=as16a3d311 Call-ID: 3c359e091559-id47fwa0oxx2@192-168-10-251 CSeq: 2 ACK Max-Forwards: 70 Contact: Content-Length: 0 9 headers, 0 lines Destroying call '3c359e091559-id47fwa0oxx2@192-168-10-251'