asterisk-test*CLI> sip debug peer netplex_301 SIP Debugging Enabled for IP: 204.213.176.201:5060 asterisk-test*CLI> <-- SIP read from 204.213.176.201:5060: INVITE sip:391@asterisk-test.ntplx.net:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 204.213.176.201;branch=z9hG4bKb693d818E7CF2B7F From: "301" ;tag=C621F85A-D3C3352B To: CSeq: 1 INVITE Call-ID: c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Supported: 100rel,timer,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 255 v=0 o=- 1154212996 1154212996 IN IP4 204.213.176.201 s=Polycom IP Phone c=IN IP4 204.213.176.201 t=0 0 a=sendrecv m=audio 2250 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 --- (14 headers 11 lines)--- Using INVITE request as basis request - c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 Sending to 204.213.176.201 : 5060 (NAT) Reliably Transmitting (NAT) to 204.213.176.201:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 204.213.176.201;branch=z9hG4bKb693d818E7CF2B7F;received=204.213.176.201 From: "301" ;tag=C621F85A-D3C3352B To: ;tag=as48b8fb91 Call-ID: c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 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="08718b15" Content-Length: 0 --- Scheduling destruction of call 'c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201' in 15000 ms Found user 'netplex_301' asterisk-test*CLI> <-- SIP read from 204.213.176.201:5060: ACK sip:391@asterisk-test.ntplx.net:5060 SIP/2.0 Via: SIP/2.0/UDP 204.213.176.201;branch=z9hG4bKb693d818E7CF2B7F From: "301" ;tag=C621F85A-D3C3352B To: ;tag=as48b8fb91 CSeq: 1 ACK Call-ID: c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- asterisk-test*CLI> <-- SIP read from 204.213.176.201:5060: INVITE sip:391@asterisk-test.ntplx.net:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 204.213.176.201;branch=z9hG4bKd6cb35f1F97AB7B0 From: "301" ;tag=C621F85A-D3C3352B To: CSeq: 2 INVITE Call-ID: c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Supported: 100rel,timer,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="netplex_301", realm="asterisk", nonce="08718b15", uri="sip:391@asterisk-test.ntplx.net:5060;user=phone", response="9f39805ce9ab3030da84fa17527c547a", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 255 v=0 o=- 1154212996 1154212996 IN IP4 204.213.176.201 s=Polycom IP Phone c=IN IP4 204.213.176.201 t=0 0 a=sendrecv m=audio 2250 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 --- (15 headers 11 lines)--- Using INVITE request as basis request - c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 Sending to 204.213.176.201 : 5060 (NAT) Found user 'netplex_301' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 204.213.176.201:2250 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x1f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g722|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for s in netplex-inside (domain user=phone) list_route: hop: Transmitting (NAT) to 204.213.176.201:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 204.213.176.201;branch=z9hG4bKd6cb35f1F97AB7B0;received=204.213.176.201 From: "301" ;tag=C621F85A-D3C3352B To: Call-ID: c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/netplex_301-40d1e818", "TRUNK=ZAP/g1") in new stack -- Executing Set("SIP/netplex_301-40d1e818", "DIALOUT=rnktel") in new stack -- Executing ExecIf("SIP/netplex_301-40d1e818", "1|Set|LocalAC=860") in new stack -- Executing ExecIf("SIP/netplex_301-40d1e818", "1|Set|OutCID="NETPLEX" <8602331111>") in new stack -- Executing ExecIf("SIP/netplex_301-40d1e818", "1|Set|E911ANI="NETPLEX" <8602331111>") in new stack -- Executing SetMusicOnHold("SIP/netplex_301-40d1e818", "netplex") in new stack -- Executing Goto("SIP/netplex_301-40d1e818", "netplex-inside2|s|1") in new stack -- Goto (netplex-inside2,s,1) -- Sent into invalid extension 's' in context 'netplex-inside2' on SIP/netplex_301-40d1e818 -- Executing Answer("SIP/netplex_301-40d1e818", "") in new stack We're at 204.213.176.139 port 28362 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 204.213.176.201:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.213.176.201;branch=z9hG4bKd6cb35f1F97AB7B0;received=204.213.176.201 From: "301" ;tag=C621F85A-D3C3352B To: ;tag=as3a77905e Call-ID: c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 23853 23853 IN IP4 204.213.176.139 s=session c=IN IP4 204.213.176.139 t=0 0 m=audio 28362 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Playback("SIP/netplex_301-40d1e818", "invalid|skip") in new stack -- Playing 'invalid' (language 'en') asterisk-test*CLI> <-- SIP read from 204.213.176.201:5060: ACK sip:s@204.213.176.139 SIP/2.0 Via: SIP/2.0/UDP 204.213.176.201;branch=z9hG4bK385dd93c7CBE5003 From: "301" ;tag=C621F85A-D3C3352B To: ;tag=as3a77905e CSeq: 2 ACK Call-ID: c9b56e8e-59ba1ca4-d4b3e905@204.213.176.201 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Proxy-Authorization: Digest username="netplex_301", realm="asterisk", nonce="08718b15", uri="sip:391@asterisk-test.ntplx.net:5060;user=phone", response="9f39805ce9ab3030da84fa17527c547a", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines)--- Destroying call '6255a275-4362af9b-26cd1894@204.213.176.201' 12 headers, 3 lines Reliably Transmitting (NAT) to 204.213.176.201:5060: NOTIFY sip:netplex_301e@204.213.176.201 SIP/2.0 Via: SIP/2.0/UDP 204.213.176.139:5060;branch=z9hG4bK4b3eb15c;rport From: "asterisk" ;tag=as1821a006 To: Contact: Call-ID: 29ca5d405fc4863c7f1d4027516af2bb@204.213.176.139 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 95 Messages-Waiting: no Message-Account: sip:asterisk@204.213.176.139 Voice-Message: 0/1 (0/0) --- Scheduling destruction of call '29ca5d405fc4863c7f1d4027516af2bb@204.213.176.139' in 15000 ms 12 headers, 3 lines Reliably Transmitting (NAT) to 204.213.176.201:5060: NOTIFY sip:netplex_301d@204.213.176.201 SIP/2.0 Via: SIP/2.0/UDP 204.213.176.139:5060;branch=z9hG4bK77564382;rport From: "asterisk" ;tag=as503ca79f To: Contact: Call-ID: 1441ef800918fb201c067b0e2a7ac1ec@204.213.176.139 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary ontent-Length: 95> Messages-Waiting: no Message-Account: sip:asterisk@204.213.176.139 Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '1441ef800918fb201c067b0e2a7ac1ec@204.213.176.139' in 15000 ms 12 headers, 3 lines Reliably Transmitting (NAT) to 204.213.176.201:5060: NOTIFY sip:netplex_301c@204.213.176.201 SIP/2.0 Via: SIP/2.0/UDP 204.213.176.139:5060;branch=z9hG4bK666b0ff1;rport From: "asterisk" ;tag=as2f240b82 To: Contact: Call-ID: 11bbffd94de705195b1db13d13fff24d@204.213.176.139 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 95 Messages-Waiting: no Message-Account: sip:asterisk@204.213.176.139 Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '11bbffd94de705195b1db13d13fff24d@204.213.176.139' in 15000 ms asterisk-test*CLI> <-- SIP read from 204.213.176.201:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.213.176.139:5060;branch=z9hG4bK4b3eb15c;rport From: "asterisk" ;tag=as1821a006 To: ;tag=9EAD1F49-95CDC772 CSeq: 102 NOTIFY Call-ID: 29ca5d405fc4863c7f1d4027516af2bb@204.213.176.139 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '29ca5d405fc4863c7f1d4027516af2bb@204.213.176.139' asterisk-test*CLI> <-- SIP read from 204.213.176.201:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.213.176.139:5060;branch=z9hG4bK77564382;rport From: "asterisk" ;tag=as503ca79f To: ;tag=8C28C12F-785B148 CSeq: 102 NOTIFY Call-ID: 1441ef800918fb201c067b0e2a7ac1ec@204.213.176.139 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '1441ef800918fb201c067b0e2a7ac1ec@204.213.176.139' asterisk-test*CLI> <-- SIP read from 204.213.176.201:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.213.176.139:5060;branch=z9hG4bK666b0ff1;rport From: "asterisk" ;tag=as2f240b82 To: ;tag=BE951BB5-7D816CBE CSeq: 102 NOTIFY Call-ID: 11bbffd94de705195b1db13d13fff24d@204.213.176.139 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '11bbffd94de705195b1db13d13fff24d@204.213.176.139' Destroying call 'd819ead0-454601c6-896ef77@204.213.176.201' Destroying call '6101eeb9-73d8f19f-8e61d338@204.213.176.201' asterisk-test*CLI> sip no debug SIP Debugging Disabled