[Jun 17 10:25:59] Connected to Asterisk 1.8.4.2 currently running on lab (pid = 10898) Verbosity is at least 3 Core debug is at least 3 [2011-06-17 10:26:00] <--- SIP read from UDP:192.168.30.193:5060 ---> INVITE sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK9dac335ab Max-Forwards: 70 Content-Length: 283 To: 6010 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497899 INVITE Supported: timer Session-Expires: 30 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: 6009 Supported: replaces User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1857615529 IN IP4 192.168.30.193 s=SIP Call c=IN IP4 192.168.30.193 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> [2011-06-17 10:26:00] --- (16 headers 13 lines) --- [2011-06-17 10:26:00] == Using UDPTL TOS bits 184 [2011-06-17 10:26:00] == Using UDPTL CoS mark 5 [2011-06-17 10:26:00] Sending to 192.168.30.193:5060 (no NAT) [2011-06-17 10:26:00] Using INVITE request as basis request - edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 [2011-06-17 10:26:00] Found peer '6009' for '6009' from 192.168.30.193:5060 [2011-06-17 10:26:00] <--- Reliably Transmitting (no NAT) to 192.168.30.193:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK9dac335ab;received=192.168.30.193 From: 6009 ;tag=e196699673da571 To: 6010 ;tag=as2a2b7168 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497899 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d822aea" Content-Length: 0 <------------> [2011-06-17 10:26:00] Scheduling destruction of SIP dialog 'edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193' in 6400 ms (Method: INVITE) [2011-06-17 10:26:00] <--- SIP read from UDP:192.168.30.193:5060 ---> ACK sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK9dac335ab Max-Forwards: 70 Content-Length: 0 To: 6010 ;tag=as2a2b7168 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497899 ACK User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> [2011-06-17 10:26:00] --- (9 headers 0 lines) --- [2011-06-17 10:26:00] <--- SIP read from UDP:192.168.30.193:5060 ---> INVITE sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK0e8607be6 Max-Forwards: 70 Content-Length: 283 To: 6010 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497900 INVITE Supported: timer Session-Expires: 30 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: 6009 Content-Type: application/sdp Supported: replaces Authorization:Digest response="2b4bb810a0b49da50fd5d7cdcb63f690",username="6009",realm="asterisk",nonce="5d822aea",algorithm=MD5,uri="sip:6010@192.168.30.254:5060" User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1857615529 IN IP4 192.168.30.193 s=SIP Call c=IN IP4 192.168.30.193 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> [2011-06-17 10:26:00] --- (17 headers 13 lines) --- [2011-06-17 10:26:00] Sending to 192.168.30.193:5060 (no NAT) [2011-06-17 10:26:00] Using INVITE request as basis request - edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 [2011-06-17 10:26:00] Found peer '6009' for '6009' from 192.168.30.193:5060 [2011-06-17 10:26:00] == Using SIP RTP TOS bits 184 [2011-06-17 10:26:00] == Using SIP RTP CoS mark 5 [2011-06-17 10:26:00] Found RTP audio format 0 [2011-06-17 10:26:00] Found RTP audio format 8 [2011-06-17 10:26:00] Found RTP audio format 18 [2011-06-17 10:26:00] Found RTP audio format 101 [2011-06-17 10:26:00] Found audio description format PCMU for ID 0 [2011-06-17 10:26:00] Found audio description format PCMA for ID 8 [2011-06-17 10:26:00] Found audio description format G729 for ID 18 [2011-06-17 10:26:00] Found audio description format telephone-event for ID 101 [2011-06-17 10:26:00] Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2011-06-17 10:26:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2011-06-17 10:26:00] Peer audio RTP is at port 192.168.30.193:3000 [2011-06-17 10:26:00] Looking for 6010 in default-super (domain 192.168.30.254:5060) [2011-06-17 10:26:00] list_route: hop: [2011-06-17 10:26:00] <--- Transmitting (no NAT) to 192.168.30.193:5060 ---> SIP/2.0 422 Session Interval Too Small Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK0e8607be6;received=192.168.30.193 From: 6009 ;tag=e196699673da571 To: 6010 ;tag=as6b754259 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497900 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Date: Fri, 17 Jun 2011 14:26:00 GMT Min-SE: 90 Content-Length: 0 <------------> [2011-06-17 10:26:00] Scheduling destruction of SIP dialog 'edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193' in 6400 ms (Method: INVITE) [2011-06-17 10:26:00] <--- SIP read from UDP:192.168.30.193:5060 ---> ACK sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK0e8607be6 Max-Forwards: 70 Content-Length: 0 To: 6010 ;tag=as6b754259 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497900 ACK Authorization:Digest response="2b4bb810a0b49da50fd5d7cdcb63f690",username="6009",realm="asterisk",nonce="5d822aea",algorithm=MD5,uri="sip:6010@192.168.30.254:5060" User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> [2011-06-17 10:26:00] --- (10 headers 0 lines) --- [2011-06-17 10:26:00] <--- SIP read from UDP:192.168.30.193:5060 ---> INVITE sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK3eca93505 Max-Forwards: 70 Content-Length: 283 To: 6010 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 INVITE Supported: timer Min-SE: 90 Session-Expires: 90 Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: 6009 Content-Type: application/sdp Supported: replaces Authorization:Digest response="2b4bb810a0b49da50fd5d7cdcb63f690",username="6009",realm="asterisk",nonce="5d822aea",algorithm=MD5,uri="sip:6010@192.168.30.254:5060" User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 1857615529 IN IP4 192.168.30.193 s=SIP Call c=IN IP4 192.168.30.193 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> [2011-06-17 10:26:00] --- (18 headers 13 lines) --- [2011-06-17 10:26:00] Sending to 192.168.30.193:5060 (no NAT) [2011-06-17 10:26:00] <--- Transmitting (no NAT) to 192.168.30.193:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK3eca93505;received=192.168.30.193 From: 6009 ;tag=e196699673da571 To: 6010 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [2011-06-17 10:26:00] Audio is at 5060 [2011-06-17 10:26:00] Adding codec 0x4 (ulaw) to SDP [2011-06-17 10:26:00] Adding non-codec 0x1 (telephone-event) to SDP [2011-06-17 10:26:00] <--- Reliably Transmitting (no NAT) to 192.168.30.193:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK3eca93505;received=192.168.30.193 From: 6009 ;tag=e196699673da571 To: 6010 ;tag=as6b754259 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 648643579 648643579 IN IP4 192.168.30.254 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.30.254 t=0 0 m=audio 12226 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [2011-06-17 10:26:00] Retransmitting #1 (no NAT) to 192.168.30.193:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK3eca93505;received=192.168.30.193 From: 6009 ;tag=e196699673da571 To: 6010 ;tag=as6b754259 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 648643579 648643579 IN IP4 192.168.30.254 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.30.254 t=0 0 m=audio 12226 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-06-17 10:26:01] Retransmitting #2 (no NAT) to 192.168.30.193:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK3eca93505;received=192.168.30.193 From: 6009 ;tag=e196699673da571 To: 6010 ;tag=as6b754259 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 648643579 648643579 IN IP4 192.168.30.254 s=Asterisk PBX 1.8.4.2 c=IN IP4 192.168.30.254 t=0 0 m=audio 12226 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [2011-06-17 10:26:01] <--- SIP read from UDP:192.168.30.193:5060 ---> ACK sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK6af7a8f3b Max-Forwards: 70 Content-Length: 0 To: 6010 ;tag=as6b754259 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 ACK Contact: 6009 Authorization:Digest response="2b4bb810a0b49da50fd5d7cdcb63f690",username="6009",realm="asterisk",nonce="5d822aea",algorithm=MD5,uri="sip:6010@192.168.30.254:5060" User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> [2011-06-17 10:26:01] --- (11 headers 0 lines) --- [2011-06-17 10:26:01] <--- SIP read from UDP:192.168.30.193:5060 ---> ACK sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK6af7a8f3b Max-Forwards: 70 Content-Length: 0 To: 6010 ;tag=as6b754259 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 ACK Contact: 6009 Authorization:Digest response="2b4bb810a0b49da50fd5d7cdcb63f690",username="6009",realm="asterisk",nonce="5d822aea",algorithm=MD5,uri="sip:6010@192.168.30.254:5060" User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> [2011-06-17 10:26:01] --- (11 headers 0 lines) --- [2011-06-17 10:26:01] <--- SIP read from UDP:192.168.30.193:5060 ---> ACK sip:6010@192.168.30.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.193:5060;branch=z9hG4bK6af7a8f3b Max-Forwards: 70 Content-Length: 0 To: 6010 ;tag=as6b754259 From: 6009 ;tag=e196699673da571 Call-ID: edf1d00162b2f9c53cbefcc68e15a569@192.168.30.193 CSeq: 1015497901 ACK Contact: 6009 Authorization:Digest response="2b4bb810a0b49da50fd5d7cdcb63f690",username="6009",realm="asterisk",nonce="5d822aea",algorithm=MD5,uri="sip:6010@192.168.30.254:5060" User-Agent: Aastra 480i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> [2011-06-17 10:26:01] --- (11 headers 0 lines) --- [2011-06-17 10:26:02] Really destroying SIP dialog '85d77e8d3c2e4eff' Method: SUBSCRIBE lab*CLI>