Asterisk 1.2.17, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.17 currently running on evpp2 (pid = 5804) Verbosity is at least 25 -- Remote UNIX connection evpp2*CLI> sip debug SIP Debugging enabled REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 204.147.183.18:5060: REGISTER sip:sip.stanaphone.com SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK160c83f2;rport From: ;tag=as522967e1 To: Call-ID: 73bb50220272cd587bc9342a325f9afd@192.168.0.103 CSeq: 712 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="08157027", realm="stanaphone.com", algorithm=MD5, uri="sip:sip.stanaphone.com", nonce="467cd6c68beaebe461ea4ab27000d24e5800acea", response="faeb4d5a9d1805b551aaedfd43c14d04", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- evpp2*CLI> <-- SIP read from 204.147.183.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK160c83f2;rport=5060 From: ;tag=as522967e1 To: ;tag=4e487ad2f2fae333606267674b3a79b1.f7a3 Call-ID: 73bb50220272cd587bc9342a325f9afd@192.168.0.103 CSeq: 712 REGISTER Contact: ;expires=115 Server: Sip EXpress router (0.9.5-pre1 (i386/linux)) Content-Length: 0 Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=29580 req_src_ip=67.161.185.121 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1" --- (10 headers 0 lines) --- Scheduling destruction of call '73bb50220272cd587bc9342a325f9afd@192.168.0.103' in 32000 ms -- Starting simple switch on 'Zap/6-1' REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 204.11.192.22:5060: REGISTER sip:callcentric.com SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK7f9eeb61;rport From: ;tag=as45d872c8 To: Call-ID: 6880133c7f613ea67b07b9363907137c@192.168.0.103 CSeq: 1811 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="17772939780", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="c1b307d390f9be477ebb50fb12e1c808", response="6bb60d2359eeda8cb0db45d878aecf45", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- evpp2*CLI> <-- SIP read from 204.11.192.22:5060: SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK7f9eeb61;rport=5060 f: ;tag=as45d872c8 t: i: 6880133c7f613ea67b07b9363907137c@192.168.0.103 CSeq: 1811 REGISTER Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="5c5dc3dcde1632cc72fbed4ff915c331", opaque="", stale=TRUE, algorithm=MD5 l: 0 --- (8 headers 0 lines) --- Responding to challenge, registration to domain/host name callcentric.com REGISTER 13 headers, 0 lines Reliably Transmitting (NAT) to 204.11.192.22:5060: REGISTER sip:callcentric.com SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK1985aa19;rport From: ;tag=as0d2b0762 To: Call-ID: 6880133c7f613ea67b07b9363907137c@192.168.0.103 CSeq: 1812 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="17772939780", realm="callcentric.com", algorithm=MD5, uri="sip:callcentric.com", nonce="5c5dc3dcde1632cc72fbed4ff915c331", response="751b58e68591731d74bbfaf53a6374a0", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- evpp2*CLI> <-- SIP read from 204.11.192.22:5060: NOTIFY sip:s@67.161.185.121 SIP/2.0 v: SIP/2.0/UDP 204.11.192.22:5060;branch=z9hG4bK-d2cce413e7e1e4ed0ec07ce0df0f3667 f: t: i: 76258d2f8bbc53ea735ef2ebd9a075d6-05f52368f794a3d01c2ea3446313dc66@callcentric.com CSeq: 1 NOTIFY Max-Forwards: 15 m: Event: message-summary c: application/simple-message-summary l: 138 essages-Waiting: no Message-Account: sip:17772939780@callcentric.com Voice-Message: 0/0 (0/0) Fax-Message: 0/0 (0/0) None: 0/0 (0/0) --- (11 headers 5 lines) --- Transmitting (NAT) to 204.11.192.22:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP 204.11.192.22:5060;branch=z9hG4bK-d2cce413e7e1e4ed0ec07ce0df0f3667;received=204.11.192.22 From: To: ;tag=as0f951acc Call-ID: 76258d2f8bbc53ea735ef2ebd9a075d6-05f52368f794a3d01c2ea3446313dc66@callcentric.com CSeq: 1 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- evpp2*CLI> <-- SIP read from 204.11.192.22:5060: SIP/2.0 200 Ok v: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK1985aa19;rport=5060 f: ;tag=as0d2b0762 t: i: 6880133c7f613ea67b07b9363907137c@192.168.0.103 CSeq: 1812 REGISTER m: ;expires=60 l: 0 --- (8 headers 0 lines) --- Scheduling destruction of call '6880133c7f613ea67b07b9363907137c@192.168.0.103' in 32000 ms -- Executing Macro("Zap/6-1", "exten-vm|7240|7240") in new stack -- Executing Macro("Zap/6-1", "user-callerid") in new stack -- Executing NoOp("Zap/6-1", "user-callerid: device 233") in new stack -- Executing GotoIf("Zap/6-1", "0?report") in new stack -- Executing GotoIf("Zap/6-1", "0?start") in new stack -- Executing Set("Zap/6-1", "REALCALLERIDNUM=233") in new stack -- Executing NoOp("Zap/6-1", "REALCALLERIDNUM is 233") in new stack -- Executing Set("Zap/6-1", "AMPUSER=233") in new stack -- Executing Set("Zap/6-1", "AMPUSERCIDNAME=Adeel") in new stack -- Executing GotoIf("Zap/6-1", "0?report") in new stack -- Executing Set("Zap/6-1", "CALLERID(all)=Adeel <233>") in new stack -- Executing Set("Zap/6-1", "REALCALLERIDNUM=233") in new stack -- Executing NoOp("Zap/6-1", "TTL: ARG1: 7240") in new stack -- Executing GotoIf("Zap/6-1", "0?continue") in new stack -- Executing Set("Zap/6-1", "__TTL=64") in new stack -- Executing GotoIf("Zap/6-1", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("Zap/6-1", "Using CallerID "Adeel" <233>") in new stack -- Executing Set("Zap/6-1", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("Zap/6-1", "VMBOX=7240") in new stack -- Executing Set("Zap/6-1", "EXTTOCALL=7240") in new stack -- Executing Set("Zap/6-1", "CFUEXT=") in new stack -- Executing Set("Zap/6-1", "CFBEXT=") in new stack -- Executing Set("Zap/6-1", "RT=15") in new stack -- Executing Macro("Zap/6-1", "record-enable|7240|IN") in new stack -- Executing GotoIf("Zap/6-1", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("Zap/6-1", "recordingcheck|20070623-181142|1182647495.141") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070623-181142|1182647495.141: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("Zap/6-1", "No recording needed") in new stack -- Executing Macro("Zap/6-1", "dial|15|trw|7240") in new stack -- Executing DeadAGI("Zap/6-1", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'Adeel' number is '233' dialparties.agi: Methodology of ring is 'none' > dialparties.agi: USE_CONFIRMATION: 'FALSE' > dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 7240 to extension map -- dialparties.agi: Extension 7240 cf is disabled -- dialparties.agi: Extension 7240 do not disturb is disabled > dialparties.agi: extnum: 7240 > dialparties.agi: exthascw: 1 > dialparties.agi: exthascfb: 0 > dialparties.agi: extcfb: > dialparties.agi: exthascfu: 0 > dialparties.agi: extcfu: -- dialparties.agi: dbset CALLTRACE/7240 to 233 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial("Zap/6-1", "SIP/7240|15|trw") in new stack We're at 192.168.0.103 port 14328 Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 16 lines Reliably Transmitting (NAT) to 192.168.0.104:5060: INVITE sip:7240@67.161.185.121:32776;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport From: "Adeel" ;tag=as08aa317f To: Contact: Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 24 Jun 2007 01:11:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 365 v=0 =root 5804 5804 IN IP4 192.168.0.103 s=session c=IN IP4 192.168.0.103 t=0 0 m=audio 14328 RTP/AVP 18 3 110 97 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 7240 evpp2*CLI> <-- SIP read from 192.168.0.104:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport From: "Adeel" ;tag=as08aa317f To: Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103 CSeq: 102 INVITE User-Agent: Grandstream BT120 1.0.8.33 Warning: 399 67.161.185.121 "detected NAT type is full cone" Content-Length: 0 --- (9 headers 0 lines) --- evpp2*CLI> <-- SIP read from 192.168.0.104:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport From: "Adeel" ;tag=as08aa317f To: ;tag=17e302e1cbe45b3d Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103 CSeq: 102 INVITE User-Agent: Grandstream BT120 1.0.8.33 Warning: 399 67.161.185.121 "detected NAT type is full cone" Content-Length: 0 --- (9 headers 0 lines) --- -- SIP/7240-08218198 is ringing == Manager 'admin' logged off from 127.0.0.1 evpp2*CLI> <-- SIP read from 192.168.0.104:5060: --- (0 headers 0 lines) Nat keepalive --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.0.104:5060: OPTIONS sip:7240@67.161.185.121:32776;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK5e2fd17b;rport From: "Unknown" ;tag=as530907c3 To: Contact: Call-ID: 360f35ea4ff7248a5b1135e72e96488c@192.168.0.103 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 24 Jun 2007 01:11:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- evpp2*CLI> <-- SIP read from 192.168.0.104:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK5e2fd17b;rport From: "Unknown" ;tag=as530907c3 To: ;tag=17e302e1cbe45b3d Call-ID: 360f35ea4ff7248a5b1135e72e96488c@192.168.0.103 CSeq: 102 OPTIONS User-Agent: Grandstream BT120 1.0.8.33 Warning: 399 67.161.185.121 "detected NAT type is full cone" Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '360f35ea4ff7248a5b1135e72e96488c@192.168.0.103' evpp2*CLI> <-- SIP read from 192.168.0.104:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1f5b51c7;rport From: "Adeel" ;tag=as08aa317f To: ;tag=17e302e1cbe45b3d Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103 CSeq: 102 INVITE User-Agent: Grandstream BT120 1.0.8.33 Warning: 399 67.161.185.121 "detected NAT type is full cone" Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 218 v=0 o=7240 8000 8000 IN IP4 67.161.185.121 s=SIP Call c=IN IP4 67.161.185.121 t=0 0 m=audio 56378 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 67.161.185.121:56378 Found description format G729 Found description format telephone-event Capabilities: us - 0x70e (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 67.161.185.121, port 32776 Transmitting (NAT) to 192.168.0.104:5060: ACK sip:7240@67.161.185.121:32776;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK4e79b957;rport From: "Adeel" ;tag=as08aa317f To: ;tag=17e302e1cbe45b3d Contact: Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/7240-08218198 answered Zap/6-1 Scheduling destruction of call '7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 67.161.185.121, port 32776 Reliably Transmitting (NAT) to 192.168.0.104:5060: BYE sip:7240@67.161.185.121:32776;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK7ef46beb;rport From: "Adeel" ;tag=as08aa317f To: ;tag=17e302e1cbe45b3d Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/6-1' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/6-1' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Zap/6-1' -- Executing Macro("Zap/6-1", "hangupcall") in new stack -- Executing ResetCDR("Zap/6-1", "w") in new stack -- Executing NoCDR("Zap/6-1", "") in new stack -- Executing GotoIf("Zap/6-1", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("Zap/6-1", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait("Zap/6-1", "5") in new stack evpp2*CLI> <-- SIP read from 192.168.0.104:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK7ef46beb;rport From: "Adeel" ;tag=as08aa317f To: ;tag=17e302e1cbe45b3d Call-ID: 7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103 CSeq: 103 BYE User-Agent: Grandstream BT120 1.0.8.33 Warning: 399 67.161.185.121 "detected NAT type is full cone" Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '7b9f6c9846f681631d6ba365579b3dbc@192.168.0.103' -- Executing Hangup("Zap/6-1", "") in new stack == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/6-1' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'Zap/6-1' -- Hungup 'Zap/6-1' Destroying call '51274156501c534a5d9ac4f042b62766@192.168.0.103' 12 headers, 0 lines Reliably Transmitting (NAT) to 64.34.176.212:5060: OPTIONS sip:did.voip.les.net SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK2da9b019;rport From: "Unknown" ;tag=as31184c0c To: Contact: Call-ID: 642376b9719e61d6212f844015cc0ab1@67.161.185.121 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 24 Jun 2007 01:11:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- evpp2*CLI> <-- SIP read from 64.34.176.212:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK2da9b019;received=67.161.185.121;rport=5060 From: "Unknown" ;tag=as31184c0c To: ;tag=as55956209 Call-ID: 642376b9719e61d6212f844015cc0ab1@67.161.185.121 CSeq: 102 OPTIONS User-Agent: LES.NET.VoIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 --- (10 headers 0 lines) --- Destroying call '642376b9719e61d6212f844015cc0ab1@67.161.185.121' evpp2*CLI> <-- SIP read from 192.168.0.104:5060: INVITE sip:233@192.168.0.103;user=phone SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK6479bd648933e19c From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: Contact: Supported: replaces Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49459 INVITE User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 387 v=0 o=7240 8000 8000 IN IP4 67.161.185.121 s=SIP Call c=IN IP4 67.161.185.121 t=0 0 m=audio 56378 RTP/AVP 0 8 4 18 2 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 18 lines) --- Using INVITE request as basis request - 48adb4a1972878bc@192.168.0.104 Sending to 67.161.185.121 : 32776 (NAT) Reliably Transmitting (NAT) to 192.168.0.104:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK6479bd648933e19c;received=192.168.0.104 From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: ;tag=as44dd4976 Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49459 INVITE User-Agent: Asterisk PBX llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="57b0c117" Content-Length: 0 --- Scheduling destruction of call '48adb4a1972878bc@192.168.0.104' in 15000 ms Found user '7240' evpp2*CLI> <-- SIP read from 192.168.0.104:5060: ACK sip:233@192.168.0.103;user=phone SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK6479bd648933e19c From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: ;tag=as44dd4976 Contact: Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49459 ACK User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (11 headers 0 lines) --- evpp2*CLI> <-- SIP read from 192.168.0.104:5060: INVITE sip:233@192.168.0.103;user=phone SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bKebe3ef55f1e35855 From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: Contact: Supported: replaces Proxy-Authorization: Digest username="7240", realm="asterisk", algorithm=MD5, uri="sip:233@192.168.0.103;user=phone", nonce="57b0c117", response="6cc9c5fdc8366b049d225aae4ef54451" Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49460 INVITE User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 387 v=0 o=7240 8000 8001 IN IP4 67.161.185.121 s=SIP Call c=IN IP4 67.161.185.121 t=0 0 m=audio 56378 RTP/AVP 0 8 4 18 2 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (14 headers 18 lines) --- Using INVITE request as basis request - 48adb4a1972878bc@192.168.0.104 Sending to 67.161.185.121 : 32776 (NAT) Found user '7240' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 97 Found RTP audio format 9 Found RTP audio format 101 Peer audio RTP is at port 67.161.185.121:56378 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format iLBC Found description format G722 Found description format telephone-event Capabilities: us - 0x70e (gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x50c (ulaw|alaw|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 233 in from-internal (domain 192.168.0.103;user=phone) list_route: hop: Transmitting (NAT) to 192.168.0.104:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bKebe3ef55f1e35855;received=192.168.0.104 From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49460 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/7240-08201ed0", "exten-vm|233|233") in new stack -- Executing Macro("SIP/7240-08201ed0", "user-callerid") in new stack -- Executing NoOp("SIP/7240-08201ed0", "user-callerid: device 7240") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?start") in new stack -- Executing Set("SIP/7240-08201ed0", "REALCALLERIDNUM=7240") in new stack -- Executing NoOp("SIP/7240-08201ed0", "REALCALLERIDNUM is 7240") in new stack -- Executing Set("SIP/7240-08201ed0", "AMPUSER=7240") in new stack -- Executing Set("SIP/7240-08201ed0", "AMPUSERCIDNAME=Nancy") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack -- Executing Set("SIP/7240-08201ed0", "CALLERID(all)=Nancy <7240>") in new stack -- Executing Set("SIP/7240-08201ed0", "REALCALLERIDNUM=7240") in new stack -- Executing NoOp("SIP/7240-08201ed0", "TTL: ARG1: 233") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?continue") in new stack -- Executing Set("SIP/7240-08201ed0", "__TTL=64") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/7240-08201ed0", "Using CallerID "Nancy" <7240>") in new stack -- Executing Set("SIP/7240-08201ed0", "FROMCONTEXT=exten-vm") in new stack -- Executing Set("SIP/7240-08201ed0", "VMBOX=233") in new stack -- Executing Set("SIP/7240-08201ed0", "EXTTOCALL=233") in new stack -- Executing Set("SIP/7240-08201ed0", "CFUEXT=") in new stack -- Executing Set("SIP/7240-08201ed0", "CFBEXT=") in new stack -- Executing Set("SIP/7240-08201ed0", "RT=15") in new stack -- Executing Macro("SIP/7240-08201ed0", "record-enable|233|IN") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/7240-08201ed0", "recordingcheck|20070623-181203|1182647523.143") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck <-- SIP read from 192.168.0.104:5060: --- (0 headers 0 lines) Nat keepalive --- recordingcheck|20070623-181203|1182647523.143: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/7240-08201ed0", "No recording needed") in new stack -- Executing Macro("SIP/7240-08201ed0", "dial|15|trw|233") in new stack -- Executing DeadAGI("SIP/7240-08201ed0", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'Nancy' number is '7240' dialparties.agi: Methodology of ring is 'none' > dialparties.agi: USE_CONFIRMATION: 'FALSE' > dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 233 to extension map -- dialparties.agi: Extension 233 cf is disabled dialparties.agi: Extension 233 has do not disturb enabled, or followme pre-ring returned busy -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp("SIP/7240-08201ed0", "Returned from dialparties with no extensions to call") in new stack -- Executing NoOp("SIP/7240-08201ed0", "DIALSTATUS is BUSY") in new stack -- Executing Set("SIP/7240-08201ed0", "SV_DIALSTATUS=BUSY") in new stack -- Executing GosubIf("SIP/7240-08201ed0", "0?docfu|1") in new stack -- Executing GosubIf("SIP/7240-08201ed0", "0?docfb|1") in new stack -- Executing Set("SIP/7240-08201ed0", "DIALSTATUS=BUSY") in new stack -- Executing NoOp("SIP/7240-08201ed0", "Voicemail is 233") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?s-BUSY|1") in new stack -- Executing NoOp("SIP/7240-08201ed0", "Sending to Voicemail box 233") in new stack -- Executing Macro("SIP/7240-08201ed0", "vm|233|BUSY") in new stack -- Executing Macro("SIP/7240-08201ed0", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/7240-08201ed0", "user-callerid: Nancy 7240") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "1?start") in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp("SIP/7240-08201ed0", "REALCALLERIDNUM is 7240") in new stack -- Executing Set("SIP/7240-08201ed0", "AMPUSER=7240") in new stack -- Executing Set("SIP/7240-08201ed0", "AMPUSERCIDNAME=Nancy") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?report") in new stack -- Executing Set("SIP/7240-08201ed0", "CALLERID(all)=Nancy <7240>") in new stack -- Executing Set("SIP/7240-08201ed0", "REALCALLERIDNUM=7240") in new stack -- Executing NoOp("SIP/7240-08201ed0", "TTL: 64 ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/7240-08201ed0", "Using CallerID "Nancy" <7240>") in new stack -- Executing Set("SIP/7240-08201ed0", "VMGAIN=") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "1?s-BUSY|1") in new stack -- Goto (macro-vm,s-BUSY,1) -- Executing NoOp("SIP/7240-08201ed0", "BUSY voicemail") in new stack -- Executing Macro("SIP/7240-08201ed0", "get-vmcontext|233") in new stack -- Executing Set("SIP/7240-08201ed0", "VMCONTEXT=default") in new stack -- Executing GotoIf("SIP/7240-08201ed0", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing NoOp("SIP/7240-08201ed0", "") in new stack -- Executing VoiceMail("SIP/7240-08201ed0", "233@default|b") in new stack We're at 192.168.0.103 port 16160 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.104:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bKebe3ef55f1e35855;received=192.168.0.104 From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: ;tag=as1feab7b9 Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49460 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 313 v=0 o=root 5804 5804 IN IP4 192.168.0.103 s=session c=IN IP4 192.168.0.103 t=0 0 m=audio 16160 RTP/AVP 18 97 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- == Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/7240-08201ed0' in macro 'vm' == Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/7240-08201ed0' in macro 'exten-vm' == Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/7240-08201ed0' Scheduling destruction of call '48adb4a1972878bc@192.168.0.104' in 32000 ms evpp2*CLI> <-- SIP read from 192.168.0.104:5060: ACK sip:233@192.168.0.103 SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK8d746d15b4a57536 From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: ;tag=as1feab7b9 Contact: Proxy-Authorization: Digest username="7240", realm="asterisk", algorithm=MD5, uri="sip:233@192.168.0.103", nonce="57b0c117", response="69bcdce490eb32d9a650d4548ab2955e" Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49460 ACK User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (12 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 67.161.185.121, port 32776 Reliably Transmitting (NAT) to 192.168.0.104:5060: CANCEL :233@192.168.0.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK662f2add;rport From: To: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '48adb4a1972878bc@192.168.0.104' in 32000 ms == Manager 'admin' logged off from 127.0.0.1 evpp2*CLI> <-- SIP read from 192.168.0.104:5060: SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK662f2add;rport From: To: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 101 CANCEL User-Agent: Grandstream BT120 1.0.8.33 Warning: 399 67.161.185.121 "detected NAT type is full cone" Content-Length: 0 --- (9 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (NAT) to 204.147.183.18:5060: OPTIONS sip:sip.stanaphone.com SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK66820565;rport From: "Unknown" ;tag=as21a81c99 To: Contact: Call-ID: 6026db9e78e91c9371f868b403589171@67.161.185.121 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 24 Jun 2007 01:12:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- evpp2*CLI> <-- SIP read from 204.147.183.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.161.185.121:5060;branch=z9hG4bK66820565;rport=5060 From: "Unknown" ;tag=as21a81c99 To: ;tag=4e487ad2f2fae333606267674b3a79b1.64d6 Call-ID: 6026db9e78e91c9371f868b403589171@67.161.185.121 CSeq: 102 OPTIONS Server: Sip EXpress router (0.9.5-pre1 (i386/linux)) Content-Length: 0 Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=29582 req_src_ip=67.161.185.121 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1" --- (9 headers 0 lines) --- Destroying call '6026db9e78e91c9371f868b403589171@67.161.185.121' Destroying call '73bb50220272cd587bc9342a325f9afd@192.168.0.103' Destroying call '6880133c7f613ea67b07b9363907137c@192.168.0.103' evpp2*CLI> <-- SIP read from 192.168.0.104:5060: BYE sip:233@192.168.0.103 SIP/2.0 Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK420487a531a54536 From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: ;tag=as1feab7b9 Supported: replaces Proxy-Authorization: Digest username="7240", realm="asterisk", algorithm=MD5, uri="sip:233@192.168.0.103", nonce="57b0c117", response="be4e5b317dfdde22c97aa6b588057239" Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49461 BYE User-Agent: Grandstream BT120 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (12 headers 0 lines) --- Sending to 67.161.185.121 : 32776 (NAT) Transmitting (NAT) to 192.168.0.104:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 67.161.185.121:32776;branch=z9hG4bK420487a531a54536;received=192.168.0.104 From: "Nancy Spicer" ;tag=f9eba7edd6dc2d56 To: ;tag=as1feab7b9 Call-ID: 48adb4a1972878bc@192.168.0.104 CSeq: 49461 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Destroying call '48adb4a1972878bc@192.168.0.104'