Connected to Asterisk 1.6.1.0 currently running on vspbx1 (pid = 31936) vspbx1*CLI> sip set debug on SIP Debugging enabled vspbx1*CLI> <--- SIP read from UDP://10.104.230.41:5060 ---> INVITE sip:111@vspbx1.devint.marchex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.104.230.41:5060 From: Cisco7905g#2 ;tag=459288036 To: Call-ID: 49237876@10.104.230.41 CSeq: 1 INVITE Contact: Cisco7905g#2 User-Agent: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Expires: 300 Content-Length: 258 Content-Type: application/sdp v=0 o=112 42908 42908 IN IP4 10.104.230.41 s=Cisco 7905 SIP Call c=IN IP4 10.104.230.41 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (12 headers 11 lines) --- Sending to 10.104.230.41 : 5060 (no NAT) Using INVITE request as basis request - 49237876@10.104.230.41 Found peer '112' for '112' from 10.104.230.41:5060 vspbx1*CLI> <--- Reliably Transmitting (no NAT) to 10.104.230.41:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.104.230.41:5060;received=10.104.230.41 From: Cisco7905g#2 ;tag=459288036 To: ;tag=as72d3067c Call-ID: 49237876@10.104.230.41 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7373e290" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '49237876@10.104.230.41' in 32000 ms (Method: INVITE) vspbx1*CLI> <--- SIP read from UDP://10.104.230.41:5060 ---> ACK sip:111@vspbx1.devint.marchex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.104.230.41:5060;received=10.104.230.41 From: Cisco7905g#2 ;tag=459288036 To: ;tag=as72d3067c Call-ID: 49237876@10.104.230.41 CSeq: 1 ACK User-Agent: Cisco-CP7905/1.01-030807A Content-Length: 0 <-------------> --- (8 headers 0 lines) --- vspbx1*CLI> <--- SIP read from UDP://10.104.230.41:5060 ---> INVITE sip:111@vspbx1.devint.marchex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.104.230.41:5060 From: Cisco7905g#2 ;tag=459288036 To: Call-ID: 49237876@10.104.230.41 CSeq: 2 INVITE Contact: Cisco7905g#2 User-Agent: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Authorization: Digest username="112",realm="asterisk",nonce="7373e290",uri="sip:111@vspbx1.devint.marchex.com",response="ba72eb81bda5fca7b332dfff1e301bc8" Expires: 300 Content-Length: 258 Content-Type: application/sdp v=0 o=112 42911 42911 IN IP4 10.104.230.41 s=Cisco 7905 SIP Call c=IN IP4 10.104.230.41 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- Sending to 10.104.230.41 : 5060 (no NAT) Using INVITE request as basis request - 49237876@10.104.230.41 Found peer '112' for '112' from 10.104.230.41:5060 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.104.230.41:16384 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=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 10.104.230.41:16384 Looking for 111 in sip_phones (domain vspbx1.devint.marchex.com) list_route: hop: vspbx1*CLI> <--- Transmitting (no NAT) to 10.104.230.41:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.104.230.41:5060;received=10.104.230.41 From: Cisco7905g#2 ;tag=459288036 To: Call-ID: 49237876@10.104.230.41 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 10.108.166.23 port 6216 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP vspbx1*CLI> <--- Reliably Transmitting (no NAT) to 10.104.230.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.104.230.41:5060;received=10.104.230.41 From: Cisco7905g#2 ;tag=459288036 To: ;tag=as650fad84 Call-ID: 49237876@10.104.230.41 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 286 v=0 o=root 862236946 862236946 IN IP4 10.108.166.23 s=Asterisk PBX 1.6.1.0 c=IN IP4 10.108.166.23 t=0 0 m=audio 6216 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> vspbx1*CLI> <--- SIP read from UDP://10.104.230.41:5060 ---> ACK sip:111@10.108.166.23 SIP/2.0 Via: SIP/2.0/UDP 10.104.230.41:5060 From: Cisco7905g#2 ;tag=459288036 To: ;tag=as650fad84 Call-ID: 49237876@10.104.230.41 CSeq: 2 ACK User-Agent: Cisco-CP7905/1.01-030807A Authorization: Digest username="112",realm="asterisk",nonce="7373e290",uri="sip:111@vspbx1.devint.marchex.com",response="ba72eb81bda5fca7b332dfff1e301bc8" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Audio is at 10.108.166.23 port 17260 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.104.230.40:5060: INVITE sip:111@10.104.230.40:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.108.166.23:5060;branch=z9hG4bK7e15981d;rport Max-Forwards: 70 From: "mAdkins 2/2" ;tag=as4cdaedd6 To: Contact: Call-ID: 6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0 Date: Fri, 12 Jun 2009 16:29:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1397962272 1397962272 IN IP4 10.108.166.23 s=Asterisk PBX 1.6.1.0 c=IN IP4 10.108.166.23 t=0 0 m=audio 17260 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- vspbx1*CLI> <--- SIP read from UDP://10.104.230.40:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.108.166.23:5060;branch=z9hG4bK7e15981d;rport From: "mAdkins 2/2" ;tag=as4cdaedd6 To: ;tag=3294042348 Call-ID: 6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23 CSeq: 102 INVITE Server: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 <-------------> --- (9 headers 0 lines) --- vspbx1*CLI> <--- SIP read from UDP://10.104.230.40:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.108.166.23:5060;branch=z9hG4bK7e15981d;rport From: "mAdkins 2/2" ;tag=as4cdaedd6 To: ;tag=3294042348 Call-ID: 6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23 CSeq: 102 INVITE Server: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Jun 12 09:29:07] WARNING[11689]: channel.c:3075 ast_indicate_data: Unable to handle indication 3 for 'SIP/112-00737f18' vspbx1*CLI> <--- SIP read from UDP://10.104.230.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.108.166.23:5060;branch=z9hG4bK7e15981d;rport From: "mAdkins 2/2" ;tag=as4cdaedd6 To: ;tag=3294042348 Call-ID: 6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23 CSeq: 102 INVITE Contact: Cisco7905g#1 Server: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 204 Content-Type: application/sdp v=0 o=111 43259 43259 IN IP4 10.104.230.40 s=Cisco 7905 SIP Call c=IN IP4 10.104.230.40 t=0 0 m=audio 16384 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.104.230.40:16384 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.104.230.40:16384 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.104.230.40, port 5060 Transmitting (no NAT) to 10.104.230.40:5060: ACK sip:111@10.104.230.40:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.108.166.23:5060;branch=z9hG4bK33cd1169;rport Max-Forwards: 70 From: "mAdkins 2/2" ;tag=as4cdaedd6 To: ;tag=3294042348 Contact: Call-ID: 6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0 Content-Length: 0 --- [Jun 12 09:29:10] NOTICE[11689]: rtp.c:1091 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.104.230.41 vspbx1*CLI> <--- SIP read from UDP://10.104.230.40:5060 ---> BYE sip:112@10.108.166.23 SIP/2.0 Via: SIP/2.0/UDP 10.104.230.40:5060 From: ;tag=3294042348 To: "mAdkins 2/2" ;tag=as4cdaedd6 Call-ID: 6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23 CSeq: 1 BYE User-Agent: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.104.230.40 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.104.230.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.104.230.40:5060;received=10.104.230.40 From: ;tag=3294042348 To: "mAdkins 2/2" ;tag=as4cdaedd6 Call-ID: 6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23 CSeq: 1 BYE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '49237876@10.104.230.41' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.104.230.41, port 5060 Reliably Transmitting (no NAT) to 10.104.230.41:5060: BYE sip:112@10.104.230.41:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.108.166.23:5060;branch=z9hG4bK1e5d4943;rport Max-Forwards: 70 From: ;tag=as650fad84 To: Cisco7905g#2 ;tag=459288036 Call-ID: 49237876@10.104.230.41 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.1.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- vspbx1*CLI> <--- SIP read from UDP://10.104.230.41:5060 ---> BYE sip:111@10.108.166.23 SIP/2.0 Via: SIP/2.0/UDP 10.104.230.41:5060 From: Cisco7905g#2 ;tag=459288036 To: ;tag=as650fad84 Call-ID: 49237876@10.104.230.41 CSeq: 3 BYE User-Agent: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Authorization: Digest username="112",realm="asterisk",nonce="7373e290",uri="sip:111@vspbx1.devint.marchex.com",response="d1b338e77f53d3e9f9ff653e5bcfc873" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.104.230.41 : 5060 (no NAT) Scheduling destruction of SIP dialog '49237876@10.104.230.41' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 10.104.230.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.104.230.41:5060;received=10.104.230.41 From: Cisco7905g#2 ;tag=459288036 To: ;tag=as650fad84 Call-ID: 49237876@10.104.230.41 CSeq: 3 BYE Server: Asterisk PBX 1.6.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '6ae95d7e46ee6eed4d287c1a159f39e4@10.108.166.23' Method: BYE vspbx1*CLI> <--- SIP read from UDP://10.104.230.41:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.108.166.23:5060;branch=z9hG4bK1e5d4943;rport From: ;tag=as650fad84 To: Cisco7905g#2 ;tag=459288036 Call-ID: 49237876@10.104.230.41 CSeq: 102 BYE Server: Cisco-CP7905/1.01-030807A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '49237876@10.104.230.41' Method: BYE vspbx1*CLI> quit