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aastra31i-asterisk-hangup

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  1. Connected to Asterisk 1.4.38 currently running on pbx1 (pid = 3273)
  2. Verbosity is at least 50
  3. Really destroying SIP dialog '7ca6fd443ea42ba6' Method: REGISTER
  4.     -- Executing [6994999911@from-trunk:1] Set("SIP/1-pstn-0000056f", "__FROM_DID=6994999911") in new stack
  5.     -- Executing [6994999911@from-trunk:2] Gosub("SIP/1-pstn-0000056f", "app-blacklist-check|s|1") in new stack
  6.     -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/1-pstn-0000056f", "") in new stack
  7.     -- Executing [s@app-blacklist-check:2] GotoIf("SIP/1-pstn-0000056f", "0?blacklisted") in new stack
  8.     -- Executing [s@app-blacklist-check:3] Set("SIP/1-pstn-0000056f", "CALLED_BLACKLIST=1") in new stack
  9.     -- Executing [s@app-blacklist-check:4] Return("SIP/1-pstn-0000056f", "") in new stack
  10.     -- Executing [6994999911@from-trunk:3] ExecIf("SIP/1-pstn-0000056f", "0 |Set|CALLERID(name)=6994998632") in new stack
  11.     -- Executing [6994999911@from-trunk:4] Set("SIP/1-pstn-0000056f", "__CALLINGPRES_SV=allowed_not_screened") in new stack
  12.     -- Executing [6994999911@from-trunk:5] SetCallerPres("SIP/1-pstn-0000056f", "allowed_not_screened") in new stack
  13.     -- Executing [6994999911@from-trunk:6] Goto("SIP/1-pstn-0000056f", "from-did-direct|11|1") in new stack
  14.     -- Goto (from-did-direct,11,1)
  15.     -- Executing [11@from-did-direct:1] GotoIf("SIP/1-pstn-0000056f", "0?ext-local|11|1") in new stack
  16.     -- Executing [11@from-did-direct:2] Macro("SIP/1-pstn-0000056f", "user-callerid|") in new stack
  17.     -- Executing [s@macro-user-callerid:1] Set("SIP/1-pstn-0000056f", "AMPUSER=6994998632") in new stack
  18.     -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1-pstn-0000056f", "0?report") in new stack
  19.     -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1-pstn-0000056f", "1|Set|REALCALLERIDNUM=6994998632") in new stack
  20.     -- Executing [s@macro-user-callerid:4] Set("SIP/1-pstn-0000056f", "AMPUSER=") in new stack
  21.     -- Executing [s@macro-user-callerid:5] Set("SIP/1-pstn-0000056f", "AMPUSERCIDNAME=") in new stack
  22.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1-pstn-0000056f", "1?report") in new stack
  23.     -- Goto (macro-user-callerid,s,10)
  24.     -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1-pstn-0000056f", "0?continue") in new stack
  25.     -- Executing [s@macro-user-callerid:11] Set("SIP/1-pstn-0000056f", "__TTL=64") in new stack
  26.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1-pstn-0000056f", "1?continue") in new stack
  27.     -- Goto (macro-user-callerid,s,19)
  28.     -- Executing [s@macro-user-callerid:19] NoOp("SIP/1-pstn-0000056f", "Using CallerID "Doe John" <6994998632>") in new stack
  29.     -- Executing [11@from-did-direct:3] Set("SIP/1-pstn-0000056f", "__EXTTOCALL=11") in new stack
  30.     -- Executing [11@from-did-direct:4] GotoIf("SIP/1-pstn-0000056f", "1?skipdb") in new stack
  31.     -- Goto (from-did-direct,11,6)
  32.     -- Executing [11@from-did-direct:6] Set("SIP/1-pstn-0000056f", "__NODEST=") in new stack
  33.     -- Executing [11@from-did-direct:7] Set("SIP/1-pstn-0000056f", "__BLKVM_OVERRIDE=BLKVM/11/SIP/1-pstn-0000056f") in new stack
  34.     -- Executing [11@from-did-direct:8] Set("SIP/1-pstn-0000056f", "__BLKVM_BASE=11") in new stack
  35.     -- Executing [11@from-did-direct:9] Set("SIP/1-pstn-0000056f", "DB(BLKVM/11/SIP/1-pstn-0000056f)=TRUE") in new stack
  36.     -- Executing [11@from-did-direct:10] Set("SIP/1-pstn-0000056f", "RRNODEST=") in new stack
  37.     -- Executing [11@from-did-direct:11] Set("SIP/1-pstn-0000056f", "__NODEST=11") in new stack
  38.     -- Executing [11@from-did-direct:12] GosubIf("SIP/1-pstn-0000056f", "0?sub-fmsetcid|s|1") in new stack
  39.     -- Executing [11@from-did-direct:13] Set("SIP/1-pstn-0000056f", "RecordMethod=Group") in new stack
  40.     -- Executing [11@from-did-direct:14] Macro("SIP/1-pstn-0000056f", "record-enable|11-11|Group") in new stack
  41.     -- Executing [s@macro-record-enable:1] GotoIf("SIP/1-pstn-0000056f", "1?check") in new stack
  42.     -- Goto (macro-record-enable,s,4)
  43.     -- Executing [s@macro-record-enable:4] ExecIf("SIP/1-pstn-0000056f", "0|MacroExit|") in new stack
  44.     -- Executing [s@macro-record-enable:5] GotoIf("SIP/1-pstn-0000056f", "1?Group:OUT") in new stack
  45.     -- Goto (macro-record-enable,s,6)
  46.     -- Executing [s@macro-record-enable:6] Set("SIP/1-pstn-0000056f", "LOOPCNT=2") in new stack
  47.     -- Executing [s@macro-record-enable:7] Set("SIP/1-pstn-0000056f", "ITER=1") in new stack
  48.     -- Executing [s@macro-record-enable:8] GotoIf("SIP/1-pstn-0000056f", "1?continue") in new stack
  49.     -- Goto (macro-record-enable,s,13)
  50.     -- Executing [s@macro-record-enable:13] Set("SIP/1-pstn-0000056f", "ITER=2") in new stack
  51.     -- Executing [s@macro-record-enable:14] GotoIf("SIP/1-pstn-0000056f", "1?begin") in new stack
  52.     -- Goto (macro-record-enable,s,8)
  53.     -- Executing [s@macro-record-enable:8] GotoIf("SIP/1-pstn-0000056f", "1?continue") in new stack
  54.     -- Goto (macro-record-enable,s,13)
  55.     -- Executing [s@macro-record-enable:13] Set("SIP/1-pstn-0000056f", "ITER=3") in new stack
  56.     -- Executing [s@macro-record-enable:14] GotoIf("SIP/1-pstn-0000056f", "0?begin") in new stack
  57.     -- Executing [s@macro-record-enable:15] GotoIf("SIP/1-pstn-0000056f", "0?IN") in new stack
  58.     -- Executing [s@macro-record-enable:16] ExecIf("SIP/1-pstn-0000056f", "1|MacroExit|") in new stack
  59.     -- Executing [11@from-did-direct:15] Set("SIP/1-pstn-0000056f", "RingGroupMethod=ringallv2") in new stack
  60.     -- Executing [11@from-did-direct:16] Set("SIP/1-pstn-0000056f", "_FMGRP=11") in new stack
  61.     -- Executing [11@from-did-direct:17] GotoIf("SIP/1-pstn-0000056f", "0?doconfirm") in new stack
  62.     -- Executing [11@from-did-direct:18] Macro("SIP/1-pstn-0000056f", "dial|20|trw|11") in new stack
  63.     -- Executing [s@macro-dial:1] GotoIf("SIP/1-pstn-0000056f", "1?dial") in new stack
  64.     -- Goto (macro-dial,s,3)
  65.     -- Executing [s@macro-dial:3] AGI("SIP/1-pstn-0000056f", "dialparties.agi") in new stack
  66.     -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  67.   dialparties.agi: Starting New Dialparties.agi
  68.   == Parsing '/etc/asterisk/manager.conf': Found
  69.   == Parsing '/etc/asterisk/manager_additional.conf': Found
  70.   == Parsing '/etc/asterisk/manager_custom.conf': Found
  71.   dialparties.agi: Caller ID name is 'Doe John' number is '6994998632'
  72.   dialparties.agi: USE_CONFIRMATION:  'FALSE'
  73.   dialparties.agi: RINGGROUP_INDEX:   ''
  74.   dialparties.agi: Methodology of ring is  'ringallv2'
  75.     --  dialparties.agi: Added extension 11 to extension map
  76.        >  dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 20
  77.        >  dialparties.agi: fmgrp_totalprering: 22
  78.        >  dialparties.agi: found extension in pre-ring and array
  79.        >  dialparties.agi: ringallv2 ring times: REALPRERING: 22, PRERING: 2
  80.        >  dialparties.agi: Extension 11 has call screening off
  81.     --  dialparties.agi: Extension 11 cf is disabled
  82.     --  dialparties.agi: Extension 11 do not disturb is disabled
  83.        >  dialparties.agi: extnum 11 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
  84.     --  dialparties.agi: dbset CALLTRACE/11 to 6994998632
  85.     --  dialparties.agi: Filtered ARG3: 11
  86.        >  dialparties.agi: NODEST: 11 adding M(auto-blkvm) to dialopts: trwM(auto-blkvm)
  87.        >  dialparties.agi: NODEST: 11 blkvm enabled macro already in dialopts: trwM(auto-blkvm)
  88.     -- AGI Script dialparties.agi completed, returning 0
  89.     -- Executing [s@macro-dial:7] Dial("SIP/1-pstn-0000056f", "SIP/11|22|trwM(auto-blkvm)") in new stack
  90. Audio is at 192.168.1.10 port 16960
  91. Adding codec 0x1000 (g722) to SDP
  92. Adding non-codec 0x1 (telephone-event) to SDP
  93. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  94. INVITE sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  95. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3bb35efe;rport
  96. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  97. To: <sip:11@192.168.1.17:5060;transport=udp>
  98. Contact: <sip:6994998632@192.168.1.10>
  99. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  100. CSeq: 102 INVITE
  101. User-Agent: Asterisk PBX
  102. Max-Forwards: 70
  103. Date: Sat, 01 Jan 2011 00:49:12 GMT
  104. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  105. Supported: replaces
  106. Content-Type: application/sdp
  107. Content-Length: 211
  108.  
  109. v=0
  110. o=root 3273 3273 IN IP4 192.168.1.10
  111. s=session
  112. c=IN IP4 192.168.1.10
  113. t=0 0
  114. m=audio 16960 RTP/AVP 9 101
  115. a=rtpmap:9 G722/8000
  116. a=rtpmap:101 telephone-event/8000
  117. a=fmtp:101 0-16
  118. a=ptime:20
  119. a=sendrecv
  120.  
  121. ---
  122.     -- Called 11
  123.  Extension Changed 11[ext-local] new state Ringing for Notify User 12
  124.  Extension Changed 11[ext-local] new state Ringing for Notify User 15
  125.  Extension Changed 11[ext-local] new state Ringing for Notify User 14
  126. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  127. set_destination: set destination to 192.168.1.17, port 5060
  128. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  129. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  130. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK64ad4d91;rport
  131. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  132. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  133. Contact: <sip:11@192.168.1.10>
  134. Call-ID: 289210011a3e4b9a
  135. CSeq: 161 NOTIFY
  136. User-Agent: Asterisk PBX
  137. Max-Forwards: 70
  138. Event: dialog
  139. Content-Type: application/dialog-info+xml
  140. Subscription-State: active
  141. Content-Length: 224
  142.  
  143. <?xml version="1.0"?>
  144. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="59" state="full" entity="sip:11@192.168.1.10:5060">
  145. <dialog id="11" direction="recipient">
  146. <state>early</state>
  147. </dialog>
  148. </dialog-info>
  149.  
  150. ---
  151.  Extension Changed 11[ext-local] new state Ringing for Notify User 11
  152.  Extension Changed 11[ext-local] new state Ringing for Notify User 10
  153.  Extension Changed 11[ext-local] new state Ringing for Notify User 13
  154.  
  155. <--- SIP read from 192.168.1.17:5060 --->
  156. SIP/2.0 180 Ringing
  157. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3bb35efe;rport=5060;received=192.168.1.10
  158. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  159. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  160. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  161. CSeq: 102 INVITE
  162. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  163. Allow-Events: talk, hold, conference, aastra-xml, LocalModeStatus
  164. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  165. Server: Aastra 6731i/2.6.0.2010
  166. Supported: path
  167. Content-Length: 0
  168.  
  169.  
  170. <------------->
  171. --- (12 headers 0 lines) ---
  172.  
  173. <--- SIP read from 192.168.1.17:5060 --->
  174. SIP/2.0 200 OK
  175. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK64ad4d91;rport=5060;received=192.168.1.10
  176. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  177. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  178. Call-ID: 289210011a3e4b9a
  179. CSeq: 161 NOTIFY
  180. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  181. Server: Aastra 6731i/2.6.0.2010
  182. Content-Length: 0
  183.  
  184.  
  185. <------------->
  186. --- (9 headers 0 lines) ---
  187. SIP Response message for INCOMING dialog NOTIFY arrived
  188.     -- SIP/11-00000570 is ringing
  189.  
  190. <--- SIP read from 192.168.1.17:5060 --->
  191. SIP/2.0 200 OK
  192. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3bb35efe;rport=5060;received=192.168.1.10
  193. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  194. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  195. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  196. CSeq: 102 INVITE
  197. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  198. Allow-Events: talk, hold, conference, aastra-xml, LocalModeStatus
  199. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  200. Server: Aastra 6731i/2.6.0.2010
  201. Supported: path, timer, replaces
  202. Content-Type: application/sdp
  203. Content-Length: 206
  204.  
  205. v=0
  206. o=MxSIP 0 0 IN IP4 192.168.1.17
  207. s=SIP Call
  208. c=IN IP4 192.168.1.17
  209. t=0 0
  210. m=audio 3000 RTP/AVP 9 101
  211. a=rtpmap:9 G722/8000
  212. a=rtpmap:101 telephone-event/8000
  213. a=fmtp:101 0-15
  214. a=ptime:20
  215. a=sendrecv
  216.  
  217. <------------->
  218. --- (13 headers 11 lines) ---
  219. Found RTP audio format 9
  220. Found RTP audio format 101
  221. Found audio description format G722 for ID 9
  222. Found audio description format telephone-event for ID 101
  223. Capabilities: us - 0x1000 (g722), peer - audio=0x1000 (g722)/video=0x0 (nothing), combined - 0x1000 (g722)
  224. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  225. Peer audio RTP is at port 192.168.1.17:3000
  226. list_route: hop: <sip:11@192.168.1.17:5060;transport=udp>
  227. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  228. set_destination: set destination to 192.168.1.17, port 5060
  229. Transmitting (NAT) to 192.168.1.17:5060:
  230. ACK sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  231. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK7b93d0a2;rport
  232. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  233. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  234. Contact: <sip:6994998632@192.168.1.10>
  235. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  236. CSeq: 102 ACK
  237. User-Agent: Asterisk PBX
  238. Max-Forwards: 70
  239. Content-Length: 0
  240.  
  241.  
  242. ---
  243.     -- SIP/11-00000570 answered SIP/1-pstn-0000056f
  244.     -- Executing [s@macro-auto-blkvm:1] Set("SIP/11-00000570", "__MACRO_RESULT=") in new stack
  245.  Extension Changed 11[ext-local] new state InUse for Notify User 12
  246.  Extension Changed 11[ext-local] new state InUse for Notify User 15
  247.  Extension Changed 11[ext-local] new state InUse for Notify User 14
  248. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  249. set_destination: set destination to 192.168.1.17, port 5060
  250. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  251. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  252. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK22c3ab1f;rport
  253. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  254. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  255. Contact: <sip:11@192.168.1.10>
  256. Call-ID: 289210011a3e4b9a
  257. CSeq: 162 NOTIFY
  258. User-Agent: Asterisk PBX
  259. Max-Forwards: 70
  260. Event: dialog
  261. Content-Type: application/dialog-info+xml
  262. Subscription-State: active
  263. Content-Length: 206
  264.  
  265. <?xml version="1.0"?>
  266. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="60" state="full" entity="sip:11@192.168.1.10:5060">
  267. <dialog id="11">
  268. <state>confirmed</state>
  269. </dialog>
  270. </dialog-info>
  271.  
  272. ---
  273.  Extension Changed 11[ext-local] new state InUse for Notify User 11
  274.  Extension Changed 11[ext-local] new state InUse for Notify User 10
  275.  Extension Changed 11[ext-local] new state InUse for Notify User 13
  276.     -- Executing [s@macro-auto-blkvm:2] NoOp("SIP/11-00000570", "Deleting: BLKVM/11/SIP/1-pstn-0000056f TRUE") in new stack
  277.  
  278. <--- SIP read from 192.168.1.17:5060 --->
  279. SIP/2.0 200 OK
  280. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK22c3ab1f;rport=5060;received=192.168.1.10
  281. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  282. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  283. Call-ID: 289210011a3e4b9a
  284. CSeq: 162 NOTIFY
  285. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  286. Server: Aastra 6731i/2.6.0.2010
  287. Content-Length: 0
  288.  
  289.  
  290. <------------->
  291. --- (9 headers 0 lines) ---
  292. SIP Response message for INCOMING dialog NOTIFY arrived
  293.  
  294. <--- SIP read from 192.168.1.17:5060 --->
  295. INVITE sip:6994998632@192.168.1.10 SIP/2.0
  296. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc6363d81f39b75c60.fede541c4f84a0ee9
  297. Max-Forwards: 70
  298. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  299. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  300. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  301. CSeq: 3846 INVITE
  302. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  303. Allow-Events: talk, hold, conference, aastra-xml, LocalModeStatus
  304. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  305. Supported: path, timer, 100rel, replaces
  306. User-Agent: Aastra 6731i/2.6.0.2010
  307. Content-Type: application/sdp
  308. Content-Length: 280
  309.  
  310. v=0
  311. o=MxSIP 0 1 IN IP4 192.168.1.17
  312. s=SIP Call
  313. c=IN IP4 192.168.1.17
  314. t=0 0
  315. m=audio 3000 RTP/AVP 9 0 8 101
  316. a=rtpmap:9 G722/8000
  317. a=rtpmap:0 PCMU/8000
  318. a=rtpmap:8 PCMA/8000
  319. a=rtpmap:101 telephone-event/8000
  320. a=silenceSupp:on - - - -
  321. a=fmtp:101 0-15
  322. a=ptime:20
  323. a=sendonly
  324.  
  325. <------------->
  326. --- (14 headers 14 lines) ---
  327. Sending to 192.168.1.17 : 5060 (NAT)
  328. Found RTP audio format 9
  329. Found RTP audio format 0
  330. Found RTP audio format 8
  331. Found RTP audio format 101
  332. Found audio description format G722 for ID 9
  333. Found audio description format PCMU for ID 0
  334. Found audio description format PCMA for ID 8
  335. Found audio description format telephone-event for ID 101
  336. Capabilities: us - 0x1000 (g722), peer - audio=0x100c (ulaw|alaw|g722)/video=0x0 (nothing), combined - 0x1000 (g722)
  337. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  338. Peer audio RTP is at port 192.168.1.17:3000
  339.  
  340. <--- Transmitting (NAT) to 192.168.1.17:5060 --->
  341. SIP/2.0 100 Trying
  342. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc6363d81f39b75c60.fede541c4f84a0ee9;received=192.168.1.17
  343. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  344. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  345. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  346. CSeq: 3846 INVITE
  347. User-Agent: Asterisk PBX
  348. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  349. Supported: replaces
  350. Contact: <sip:6994998632@192.168.1.10>
  351. Content-Length: 0
  352.  
  353.  
  354. <------------>
  355. Audio is at 192.168.1.10 port 16960
  356. Adding codec 0x1000 (g722) to SDP
  357. Adding non-codec 0x1 (telephone-event) to SDP
  358.  
  359. <--- Reliably Transmitting (NAT) to 192.168.1.17:5060 --->
  360. SIP/2.0 200 OK
  361. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bKc6363d81f39b75c60.fede541c4f84a0ee9;received=192.168.1.17
  362. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  363. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  364. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  365. CSeq: 3846 INVITE
  366. User-Agent: Asterisk PBX
  367. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  368. Supported: replaces
  369. Contact: <sip:6994998632@192.168.1.10>
  370. Content-Type: application/sdp
  371. Content-Length: 211
  372.  
  373. v=0
  374. o=root 3273 3274 IN IP4 192.168.1.10
  375. s=session
  376. c=IN IP4 192.168.1.10
  377. t=0 0
  378. m=audio 16960 RTP/AVP 9 101
  379. a=rtpmap:9 G722/8000
  380. a=rtpmap:101 telephone-event/8000
  381. a=fmtp:101 0-16
  382. a=ptime:20
  383. a=recvonly
  384.  
  385. <------------>
  386.     -- Started music on hold, class 'default', on SIP/1-pstn-0000056f
  387.  Extension Changed 11[ext-local] new state Hold for Notify User 12
  388.  Extension Changed 11[ext-local] new state Hold for Notify User 15
  389.  Extension Changed 11[ext-local] new state Hold for Notify User 14
  390. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  391. set_destination: set destination to 192.168.1.17, port 5060
  392. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  393. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  394. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47fd5e4a;rport
  395. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  396. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  397. Contact: <sip:11@192.168.1.10>
  398. Call-ID: 289210011a3e4b9a
  399. CSeq: 163 NOTIFY
  400. User-Agent: Asterisk PBX
  401. Max-Forwards: 70
  402. Event: dialog
  403. Content-Type: application/dialog-info+xml
  404. Subscription-State: active
  405. Content-Length: 317
  406.  
  407. <?xml version="1.0"?>
  408. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="61" state="full" entity="sip:11@192.168.1.10:5060">
  409. <dialog id="11">
  410. <state>confirmed</state>
  411. <local>
  412. <target uri="sip:11@192.168.1.10:5060">
  413. <param pname="+sip.rendering" pvalue="no"/>
  414. </target>
  415. </local>
  416. </dialog>
  417. </dialog-info>
  418.  
  419. ---
  420.  Extension Changed 11[ext-local] new state Hold for Notify User 11
  421.  Extension Changed 11[ext-local] new state Hold for Notify User 10
  422.  Extension Changed 11[ext-local] new state Hold for Notify User 13
  423.  
  424. <--- SIP read from 192.168.1.17:5060 --->
  425. ACK sip:6994998632@192.168.1.10 SIP/2.0
  426. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK4c803405e885ebdd2.095544d6e9ef5f427
  427. Max-Forwards: 70
  428. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  429. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  430. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  431. CSeq: 3846 ACK
  432. User-Agent: Aastra 6731i/2.6.0.2010
  433. Content-Length: 0
  434.  
  435.  
  436. <------------->
  437. --- (9 headers 0 lines) ---
  438.  
  439. <--- SIP read from 192.168.1.17:5060 --->
  440. SIP/2.0 200 OK
  441. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47fd5e4a;rport=5060;received=192.168.1.10
  442. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  443. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  444. Call-ID: 289210011a3e4b9a
  445. CSeq: 163 NOTIFY
  446. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  447. Server: Aastra 6731i/2.6.0.2010
  448. Content-Length: 0
  449.  
  450.  
  451. <------------->
  452. --- (9 headers 0 lines) ---
  453. SIP Response message for INCOMING dialog NOTIFY arrived
  454.  
  455. <--- SIP read from 192.168.1.17:5060 --->
  456. INVITE sip:12@192.168.1.10:5060 SIP/2.0
  457. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK35be83b706ff2cb9a.c7ab6e4d0d1cb57ad
  458. Max-Forwards: 70
  459. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  460. To: "12" <sip:12@192.168.1.10:5060>
  461. Call-ID: 6425e83c54a11bd9
  462. CSeq: 1300 INVITE
  463. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  464. Allow-Events: talk, hold, conference, aastra-xml, LocalModeStatus
  465. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  466. Supported: path, timer, 100rel, replaces
  467. User-Agent: Aastra 6731i/2.6.0.2010
  468. Content-Type: application/sdp
  469. Content-Length: 280
  470.  
  471. v=0
  472. o=MxSIP 0 0 IN IP4 192.168.1.17
  473. s=SIP Call
  474. c=IN IP4 192.168.1.17
  475. t=0 0
  476. m=audio 3002 RTP/AVP 9 0 8 101
  477. a=rtpmap:9 G722/8000
  478. a=rtpmap:0 PCMU/8000
  479. a=rtpmap:8 PCMA/8000
  480. a=rtpmap:101 telephone-event/8000
  481. a=silenceSupp:on - - - -
  482. a=fmtp:101 0-15
  483. a=ptime:20
  484. a=sendrecv
  485.  
  486. <------------->
  487. --- (14 headers 14 lines) ---
  488. Sending to 192.168.1.17 : 5060 (NAT)
  489. Using INVITE request as basis request - 6425e83c54a11bd9
  490.  
  491. <--- Reliably Transmitting (NAT) to 192.168.1.17:5060 --->
  492. SIP/2.0 407 Proxy Authentication Required
  493. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK35be83b706ff2cb9a.c7ab6e4d0d1cb57ad;received=192.168.1.17
  494. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  495. To: "12" <sip:12@192.168.1.10:5060>;tag=as4c30df54
  496. Call-ID: 6425e83c54a11bd9
  497. CSeq: 1300 INVITE
  498. User-Agent: Asterisk PBX
  499. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  500. Supported: replaces
  501. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="515d7af3"
  502. Content-Length: 0
  503.  
  504.  
  505. <------------>
  506. Scheduling destruction of SIP dialog '6425e83c54a11bd9' in 32000 ms (Method: INVITE)
  507. Found user '11'
  508.  
  509. <--- SIP read from 192.168.1.17:5060 --->
  510. ACK sip:12@192.168.1.10:5060 SIP/2.0
  511. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK35be83b706ff2cb9a.c7ab6e4d0d1cb57ad
  512. Max-Forwards: 70
  513. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  514. To: "12" <sip:12@192.168.1.10:5060>;tag=as4c30df54
  515. Call-ID: 6425e83c54a11bd9
  516. CSeq: 1300 ACK
  517. User-Agent: Aastra 6731i/2.6.0.2010
  518. Content-Length: 0
  519.  
  520.  
  521. <------------->
  522. --- (9 headers 0 lines) ---
  523.  
  524. <--- SIP read from 192.168.1.17:5060 --->
  525. INVITE sip:12@192.168.1.10:5060 SIP/2.0
  526. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK715b891c4962c7082.9434695aa7547ae5a
  527. Proxy-Authorization: Digest username="11",realm="asterisk",nonce="515d7af3",uri="sip:12@192.168.1.10:5060",response="602e4bfc47c2b46d4dde5498ef052d71",algorithm=MD5
  528. Max-Forwards: 70
  529. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  530. To: "12" <sip:12@192.168.1.10:5060>
  531. Call-ID: 6425e83c54a11bd9
  532. CSeq: 1301 INVITE
  533. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  534. Allow-Events: talk, hold, conference, aastra-xml, LocalModeStatus
  535. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  536. Supported: path, timer, 100rel, replaces
  537. User-Agent: Aastra 6731i/2.6.0.2010
  538. Content-Type: application/sdp
  539. Content-Length: 280
  540.  
  541. v=0
  542. o=MxSIP 0 0 IN IP4 192.168.1.17
  543. s=SIP Call
  544. c=IN IP4 192.168.1.17
  545. t=0 0
  546. m=audio 3002 RTP/AVP 9 0 8 101
  547. a=rtpmap:9 G722/8000
  548. a=rtpmap:0 PCMU/8000
  549. a=rtpmap:8 PCMA/8000
  550. a=rtpmap:101 telephone-event/8000
  551. a=silenceSupp:on - - - -
  552. a=fmtp:101 0-15
  553. a=ptime:20
  554. a=sendrecv
  555.  
  556. <------------->
  557. --- (15 headers 14 lines) ---
  558. Sending to 192.168.1.17 : 5060 (NAT)
  559. Using INVITE request as basis request - 6425e83c54a11bd9
  560. Found user '11'
  561. Found RTP audio format 9
  562. Found RTP audio format 0
  563. Found RTP audio format 8
  564. Found RTP audio format 101
  565. Found audio description format G722 for ID 9
  566. Found audio description format PCMU for ID 0
  567. Found audio description format PCMA for ID 8
  568. Found audio description format telephone-event for ID 101
  569. Capabilities: us - 0x1000 (g722), peer - audio=0x100c (ulaw|alaw|g722)/video=0x0 (nothing), combined - 0x1000 (g722)
  570. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  571. Peer audio RTP is at port 192.168.1.17:3002
  572. Looking for 12 in from-internal (domain 192.168.1.10)
  573. list_route: hop: <sip:11@192.168.1.17:5060;transport=udp>
  574.  
  575. <--- Transmitting (NAT) to 192.168.1.17:5060 --->
  576. SIP/2.0 100 Trying
  577. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK715b891c4962c7082.9434695aa7547ae5a;received=192.168.1.17
  578. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  579. To: "12" <sip:12@192.168.1.10:5060>
  580. Call-ID: 6425e83c54a11bd9
  581. CSeq: 1301 INVITE
  582. User-Agent: Asterisk PBX
  583. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  584. Supported: replaces
  585. Contact: <sip:12@192.168.1.10>
  586. Content-Length: 0
  587.  
  588.  
  589. <------------>
  590.     -- Executing [12@from-internal:1] GotoIf("SIP/11-00000571", "0?ext-local|12|1") in new stack
  591.     -- Executing [12@from-internal:2] Macro("SIP/11-00000571", "user-callerid|") in new stack
  592.     -- Executing [s@macro-user-callerid:1] Set("SIP/11-00000571", "AMPUSER=11") in new stack
  593.     -- Executing [s@macro-user-callerid:2] GotoIf("SIP/11-00000571", "0?report") in new stack
  594.     -- Executing [s@macro-user-callerid:3] ExecIf("SIP/11-00000571", "1|Set|REALCALLERIDNUM=11") in new stack
  595.     -- Executing [s@macro-user-callerid:4] Set("SIP/11-00000571", "AMPUSER=11") in new stack
  596.     -- Executing [s@macro-user-callerid:5] Set("SIP/11-00000571", "AMPUSERCIDNAME=Dining Room") in new stack
  597.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/11-00000571", "0?report") in new stack
  598.     -- Executing [s@macro-user-callerid:7] Set("SIP/11-00000571", "AMPUSERCID=11") in new stack
  599.     -- Executing [s@macro-user-callerid:8] Set("SIP/11-00000571", "CALLERID(all)="Dining Room" <11>") in new stack
  600.     -- Executing [s@macro-user-callerid:9] ExecIf("SIP/11-00000571", "0|Set|CHANNEL(language)=") in new stack
  601.     -- Executing [s@macro-user-callerid:10] GotoIf("SIP/11-00000571", "0?continue") in new stack
  602.     -- Executing [s@macro-user-callerid:11] Set("SIP/11-00000571", "__TTL=64") in new stack
  603.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/11-00000571", "1?continue") in new stack
  604.     -- Goto (macro-user-callerid,s,19)
  605.     -- Executing [s@macro-user-callerid:19] NoOp("SIP/11-00000571", "Using CallerID "Dining Room" <11>") in new stack
  606.     -- Executing [12@from-internal:3] Set("SIP/11-00000571", "__EXTTOCALL=12") in new stack
  607.     -- Executing [12@from-internal:4] GotoIf("SIP/11-00000571", "1?skipdb") in new stack
  608.     -- Goto (from-internal,12,6)
  609.     -- Executing [12@from-internal:6] Set("SIP/11-00000571", "__NODEST=") in new stack
  610.     -- Executing [12@from-internal:7] Set("SIP/11-00000571", "__BLKVM_OVERRIDE=BLKVM/12/SIP/11-00000571") in new stack
  611.     -- Executing [12@from-internal:8] Set("SIP/11-00000571", "__BLKVM_BASE=12") in new stack
  612.     -- Executing [12@from-internal:9] Set("SIP/11-00000571", "DB(BLKVM/12/SIP/11-00000571)=TRUE") in new stack
  613.     -- Executing [12@from-internal:10] Set("SIP/11-00000571", "RRNODEST=") in new stack
  614.     -- Executing [12@from-internal:11] Set("SIP/11-00000571", "__NODEST=12") in new stack
  615.     -- Executing [12@from-internal:12] GosubIf("SIP/11-00000571", "0?sub-fmsetcid|s|1") in new stack
  616.     -- Executing [12@from-internal:13] Set("SIP/11-00000571", "RecordMethod=Group") in new stack
  617.     -- Executing [12@from-internal:14] Macro("SIP/11-00000571", "record-enable|12-12|Group") in new stack
  618.     -- Executing [s@macro-record-enable:1] GotoIf("SIP/11-00000571", "1?check") in new stack
  619.     -- Goto (macro-record-enable,s,4)
  620.     -- Executing [s@macro-record-enable:4] ExecIf("SIP/11-00000571", "0|MacroExit|") in new stack
  621.     -- Executing [s@macro-record-enable:5] GotoIf("SIP/11-00000571", "1?Group:OUT") in new stack
  622.     -- Goto (macro-record-enable,s,6)
  623.     -- Executing [s@macro-record-enable:6] Set("SIP/11-00000571", "LOOPCNT=2") in new stack
  624.     -- Executing [s@macro-record-enable:7] Set("SIP/11-00000571", "ITER=1") in new stack
  625.     -- Executing [s@macro-record-enable:8] GotoIf("SIP/11-00000571", "1?continue") in new stack
  626.     -- Goto (macro-record-enable,s,13)
  627.     -- Executing [s@macro-record-enable:13] Set("SIP/11-00000571", "ITER=2") in new stack
  628.     -- Executing [s@macro-record-enable:14] GotoIf("SIP/11-00000571", "1?begin") in new stack
  629.     -- Goto (macro-record-enable,s,8)
  630.     -- Executing [s@macro-record-enable:8] GotoIf("SIP/11-00000571", "1?continue") in new stack
  631.     -- Goto (macro-record-enable,s,13)
  632.     -- Executing [s@macro-record-enable:13] Set("SIP/11-00000571", "ITER=3") in new stack
  633.     -- Executing [s@macro-record-enable:14] GotoIf("SIP/11-00000571", "0?begin") in new stack
  634.     -- Executing [s@macro-record-enable:15] GotoIf("SIP/11-00000571", "0?IN") in new stack
  635.     -- Executing [s@macro-record-enable:16] ExecIf("SIP/11-00000571", "1|MacroExit|") in new stack
  636.     -- Executing [12@from-internal:15] Set("SIP/11-00000571", "RingGroupMethod=ringallv2") in new stack
  637.     -- Executing [12@from-internal:16] Set("SIP/11-00000571", "_FMGRP=12") in new stack
  638.     -- Executing [12@from-internal:17] GotoIf("SIP/11-00000571", "0?doconfirm") in new stack
  639.     -- Executing [12@from-internal:18] Macro("SIP/11-00000571", "dial|20|trw|12") in new stack
  640.     -- Executing [s@macro-dial:1] GotoIf("SIP/11-00000571", "1?dial") in new stack
  641.     -- Goto (macro-dial,s,3)
  642.     -- Executing [s@macro-dial:3] AGI("SIP/11-00000571", "dialparties.agi") in new stack
  643.     -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  644.   dialparties.agi: Starting New Dialparties.agi
  645.   == Parsing '/etc/asterisk/manager.conf': Found
  646.   == Parsing '/etc/asterisk/manager_additional.conf': Found
  647.   == Parsing '/etc/asterisk/manager_custom.conf': Found
  648.   dialparties.agi: Caller ID name is 'Dining Room' number is '11'
  649.   dialparties.agi: USE_CONFIRMATION:  'FALSE'
  650.   dialparties.agi: RINGGROUP_INDEX:   ''
  651.   dialparties.agi: Methodology of ring is  'ringallv2'
  652.     --  dialparties.agi: Added extension 12 to extension map
  653.        >  dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 20
  654.        >  dialparties.agi: fmgrp_totalprering: 22
  655.        >  dialparties.agi: found extension in pre-ring and array
  656.        >  dialparties.agi: ringallv2 ring times: REALPRERING: 22, PRERING: 2
  657.     --  dialparties.agi: Extension 12 cf is disabled
  658.     --  dialparties.agi: Extension 12 do not disturb is disabled
  659.        >  dialparties.agi: extnum 12 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
  660.     --  dialparties.agi: dbset CALLTRACE/12 to 11
  661.     --  dialparties.agi: Filtered ARG3: 12
  662.        >  dialparties.agi: NODEST: 12 adding M(auto-blkvm) to dialopts: trwM(auto-blkvm)
  663.        >  dialparties.agi: NODEST: 12 blkvm enabled macro already in dialopts: trwM(auto-blkvm)
  664.     -- AGI Script dialparties.agi completed, returning 0
  665.     -- Executing [s@macro-dial:7] Dial("SIP/11-00000571", "SIP/12|22|trwM(auto-blkvm)") in new stack
  666.     -- Called 12
  667.  Extension Changed 12[ext-local] new state Ringing for Notify User 12
  668.  
  669. <--- Transmitting (NAT) to 192.168.1.17:5060 --->
  670. SIP/2.0 180 Ringing
  671. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK715b891c4962c7082.9434695aa7547ae5a;received=192.168.1.17
  672. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  673. To: "12" <sip:12@192.168.1.10:5060>;tag=as5818ccd2
  674. Call-ID: 6425e83c54a11bd9
  675. CSeq: 1301 INVITE
  676. User-Agent: Asterisk PBX
  677. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  678. Supported: replaces
  679. Contact: <sip:12@192.168.1.10>
  680. Content-Length: 0
  681.  
  682.  
  683. <------------>
  684.  Extension Changed 12[ext-local] new state Ringing for Notify User 15
  685.  Extension Changed 12[ext-local] new state Ringing for Notify User 14
  686. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  687. set_destination: set destination to 192.168.1.17, port 5060
  688. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  689. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  690. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2219a690;rport
  691. From: "" <sip:12@192.168.1.10:5060>;tag=as01d3df4a
  692. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=90f820dbe6
  693. Contact: <sip:12@192.168.1.10>
  694. Call-ID: 49f8da1dc614c738
  695. CSeq: 131 NOTIFY
  696. User-Agent: Asterisk PBX
  697. Max-Forwards: 70
  698. Event: dialog
  699. Content-Type: application/dialog-info+xml
  700. Subscription-State: active
  701. Content-Length: 224
  702.  
  703. <?xml version="1.0"?>
  704. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="29" state="full" entity="sip:12@192.168.1.10:5060">
  705. <dialog id="12" direction="recipient">
  706. <state>early</state>
  707. </dialog>
  708. </dialog-info>
  709.  
  710. ---
  711.  Extension Changed 12[ext-local] new state Ringing for Notify User 11
  712.  Extension Changed 12[ext-local] new state Ringing for Notify User 10
  713.  Extension Changed 12[ext-local] new state Ringing for Notify User 13
  714.  
  715. <--- SIP read from 192.168.1.17:5060 --->
  716. SIP/2.0 200 OK
  717. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2219a690;rport=5060;received=192.168.1.10
  718. From: "" <sip:12@192.168.1.10:5060>;tag=as01d3df4a
  719. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=90f820dbe6
  720. Call-ID: 49f8da1dc614c738
  721. CSeq: 131 NOTIFY
  722. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  723. Server: Aastra 6731i/2.6.0.2010
  724. Content-Length: 0
  725.  
  726.  
  727. <------------->
  728. --- (9 headers 0 lines) ---
  729. SIP Response message for INCOMING dialog NOTIFY arrived
  730.   == Parsing '/etc/asterisk/manager.conf': Found
  731.   == Parsing '/etc/asterisk/manager_additional.conf': Found
  732.   == Parsing '/etc/asterisk/manager_custom.conf': Found
  733.     -- SIP/12-00000572 is ringing
  734.   == Parsing '/etc/asterisk/manager.conf': Found
  735.   == Parsing '/etc/asterisk/manager_additional.conf': Found
  736.   == Parsing '/etc/asterisk/manager_custom.conf': Found
  737.   == Parsing '/etc/asterisk/manager.conf': Found
  738.   == Parsing '/etc/asterisk/manager_additional.conf': Found
  739.   == Parsing '/etc/asterisk/manager_custom.conf': Found
  740.   == Parsing '/etc/asterisk/manager.conf': Found
  741.   == Parsing '/etc/asterisk/manager_additional.conf': Found
  742.   == Parsing '/etc/asterisk/manager_custom.conf': Found
  743.  
  744. <--- SIP read from 192.168.1.17:5060 --->
  745. CANCEL sip:12@192.168.1.10:5060 SIP/2.0
  746. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK715b891c4962c7082.9434695aa7547ae5a
  747. Max-Forwards: 70
  748. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  749. To: "12" <sip:12@192.168.1.10:5060>
  750. Call-ID: 6425e83c54a11bd9
  751. CSeq: 1301 CANCEL
  752. Supported: path
  753. User-Agent: Aastra 6731i/2.6.0.2010
  754. Content-Length: 0
  755.  
  756.  
  757. <------------->
  758. --- (10 headers 0 lines) ---
  759. Sending to 192.168.1.17 : 5060 (NAT)
  760.  
  761. <--- Reliably Transmitting (NAT) to 192.168.1.17:5060 --->
  762. SIP/2.0 487 Request Terminated
  763. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK715b891c4962c7082.9434695aa7547ae5a;received=192.168.1.17
  764. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  765. To: "12" <sip:12@192.168.1.10:5060>;tag=as5818ccd2
  766. Call-ID: 6425e83c54a11bd9
  767. CSeq: 1301 INVITE
  768. User-Agent: Asterisk PBX
  769. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  770. Supported: replaces
  771. Content-Length: 0
  772.  
  773.  
  774. <------------>
  775.  
  776. <--- Transmitting (NAT) to 192.168.1.17:5060 --->
  777. SIP/2.0 200 OK
  778. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK715b891c4962c7082.9434695aa7547ae5a;received=192.168.1.17
  779. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  780. To: "12" <sip:12@192.168.1.10:5060>;tag=as5818ccd2
  781. Call-ID: 6425e83c54a11bd9
  782. CSeq: 1301 CANCEL
  783. User-Agent: Asterisk PBX
  784. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  785. Supported: replaces
  786. Content-Length: 0
  787.  
  788.  
  789. <------------>
  790.  
  791. <--- SIP read from 192.168.1.17:5060 --->
  792. REFER sip:6994998632@192.168.1.10 SIP/2.0
  793. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK02ad271945f3d591e.ab872f99efc2cbb8a
  794. Max-Forwards: 70
  795. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  796. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  797. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  798. CSeq: 3847 REFER
  799. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  800. Allow-Events: talk, hold, conference, aastra-xml, LocalModeStatus
  801. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  802. Refer-To: 12 <sip:12@192.168.1.10:5060>
  803. Referred-By: <sip:11@192.168.1.10>
  804. Supported: path, timer
  805. User-Agent: Aastra 6731i/2.6.0.2010
  806. Content-Length: 0
  807.  
  808.  
  809. <------------->
  810. --- (15 headers 0 lines) ---
  811. Call 596d687d420f456e190086aa2434a05e@192.168.1.10 got a SIP call transfer from caller: (REFER)!
  812.   == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/11-00000571' in macro 'dial'
  813. SIP transfer to extension 12@from-internal-xfer by 11@192.168.1.10
  814.   == Spawn extension (from-internal, 12, 18) exited non-zero on 'SIP/11-00000571'
  815.  
  816. <--- Transmitting (NAT) to 192.168.1.17:5060 --->
  817. SIP/2.0 202 Accepted
  818. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK02ad271945f3d591e.ab872f99efc2cbb8a;received=192.168.1.17
  819. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  820. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  821. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  822. CSeq: 3847 REFER
  823. User-Agent: Asterisk PBX
  824. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  825. Supported: replaces
  826. Contact: <sip:6994998632@192.168.1.10>
  827. Content-Length: 0
  828.  
  829.  
  830. <------------>
  831.     -- Executing [h@macro-dial:1] Macro("SIP/11-00000571", "hangupcall") in new stack
  832. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  833.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/11-00000571", "1?skiprg") in new stack
  834. set_destination: set destination to 192.168.1.17, port 5060
  835.     -- Goto (macro-hangupcall,s,4)
  836.     -- Stopped music on hold on SIP/1-pstn-0000056f
  837. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  838. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  839. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0d3f4a98;rport
  840. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  841. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  842. Contact: <sip:6994998632@192.168.1.10>
  843. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  844. CSeq: 103 NOTIFY
  845. User-Agent: Asterisk PBX
  846. Max-Forwards: 70
  847. Event: refer;id=3847
  848. Subscription-state: active
  849. Content-Type: message/sipfrag;version=2.0
  850. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  851. Supported: replaces
  852. Content-Length: 21
  853.  
  854. SIP/2.0 183 Ringing
  855.  
  856. ---
  857.     -- Executing [s@macro-hangupcall:4] GotoIf("SIP/11-00000571", "0?skipblkvm") in new stack
  858.     -- Executing [s@macro-hangupcall:5] NoOp("SIP/11-00000571", "Cleaning Up Block VM Flag: BLKVM/12/SIP/11-00000571") in new stack
  859. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  860. set_destination: set destination to 192.168.1.17, port 5060
  861. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  862. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  863. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1f1f9743;rport
  864. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  865. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  866. Contact: <sip:6994998632@192.168.1.10>
  867. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  868. CSeq: 104 NOTIFY
  869. User-Agent: Asterisk PBX
  870. Max-Forwards: 70
  871. Event: refer;id=3847
  872. Subscription-state: terminated;reason=noresource
  873. Content-Type: message/sipfrag;version=2.0
  874. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  875. Supported: replaces
  876. Content-Length: 16
  877.  
  878. SIP/2.0 200 Ok
  879.  
  880. ---
  881.     -- Executing [h@from-internal-xfer:1] Macro("SIP/1-pstn-0000056f", "hangupcall") in new stack
  882.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1-pstn-0000056f", "1?skiprg") in new stack
  883.  Extension Changed 12[ext-local] new state Idle for Notify User 12
  884.     -- Goto (macro-hangupcall,s,4)
  885.     -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1-pstn-0000056f", "0?skipblkvm") in new stack
  886.  Extension Changed 12[ext-local] new state Idle for Notify User 15
  887.     -- Executing [s@macro-hangupcall:5] NoOp("SIP/1-pstn-0000056f", "Cleaning Up Block VM Flag: BLKVM/11/SIP/1-pstn-0000056f") in new stack
  888.  Extension Changed 12[ext-local] new state Idle for Notify User 14
  889. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  890. set_destination: set destination to 192.168.1.17, port 5060
  891. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  892. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  893. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0bcf4d35;rport
  894. From: "" <sip:12@192.168.1.10:5060>;tag=as01d3df4a
  895. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=90f820dbe6
  896. Contact: <sip:12@192.168.1.10>
  897. Call-ID: 49f8da1dc614c738
  898. CSeq: 132 NOTIFY
  899. User-Agent: Asterisk PBX
  900. Max-Forwards: 70
  901. Event: dialog
  902. Content-Type: application/dialog-info+xml
  903. Subscription-State: active
  904. Content-Length: 207
  905.  
  906. <?xml version="1.0"?>
  907. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="30" state="full" entity="sip:12@192.168.1.10:5060">
  908. <dialog id="12">
  909. <state>terminated</state>
  910. </dialog>
  911. </dialog-info>
  912.  
  913. ---
  914.  Extension Changed 12[ext-local] new state Idle for Notify User 11
  915.  Extension Changed 12[ext-local] new state Idle for Notify User 10
  916.  Extension Changed 12[ext-local] new state Idle for Notify User 13
  917.     -- Executing [s@macro-hangupcall:6] NoOp("SIP/11-00000571", "Deleting: BLKVM/12/SIP/11-00000571 TRUE") in new stack
  918.     -- Executing [s@macro-hangupcall:6] NoOp("SIP/1-pstn-0000056f", "Deleting: BLKVM/11/SIP/1-pstn-0000056f ") in new stack
  919.     -- Executing [s@macro-hangupcall:7] GotoIf("SIP/11-00000571", "1?theend") in new stack
  920.     -- Goto (macro-hangupcall,s,9)
  921.     -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1-pstn-0000056f", "1?theend") in new stack
  922.     -- Executing [s@macro-hangupcall:9] Hangup("SIP/11-00000571", "") in new stack
  923.     -- Goto (macro-hangupcall,s,9)
  924.   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/11-00000571' in macro 'hangupcall'
  925.   == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/11-00000571'
  926.     -- Executing [s@macro-hangupcall:9] Hangup("SIP/1-pstn-0000056f", "") in new stack
  927.   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1-pstn-0000056f' in macro 'hangupcall'
  928.   == Spawn h extension (from-internal-xfer, s, 1) exited non-zero on 'SIP/1-pstn-0000056f'
  929. Scheduling destruction of SIP dialog '596d687d420f456e190086aa2434a05e@192.168.1.10' in 6400 ms (Method: REFER)
  930.   == Spawn extension (from-internal-xfer, 12, 0) exited non-zero on 'SIP/1-pstn-0000056f' in macro 'dial'
  931.   == Spawn extension (from-internal-xfer, 12, 0) exited non-zero on 'SIP/1-pstn-0000056f'
  932.  Extension Changed 11[ext-local] new state Idle for Notify User 12
  933.  Extension Changed 11[ext-local] new state Idle for Notify User 15
  934.  Extension Changed 11[ext-local] new state Idle for Notify User 14
  935. set_destination: Parsing <sip:11@192.168.1.17:5060;transport=udp> for address/port to send to
  936. set_destination: set destination to 192.168.1.17, port 5060
  937. Reliably Transmitting (NAT) to 192.168.1.17:5060:
  938. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  939. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK74415e27;rport
  940. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  941. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  942. Contact: <sip:11@192.168.1.10>
  943. Call-ID: 289210011a3e4b9a
  944. CSeq: 164 NOTIFY
  945. User-Agent: Asterisk PBX
  946. Max-Forwards: 70
  947. Event: dialog
  948. Content-Type: application/dialog-info+xml
  949. Subscription-State: active
  950. Content-Length: 207
  951.  
  952. <?xml version="1.0"?>
  953. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="62" state="full" entity="sip:11@192.168.1.10:5060">
  954. <dialog id="11">
  955. <state>terminated</state>
  956. </dialog>
  957. </dialog-info>
  958.  
  959. ---
  960.  Extension Changed 11[ext-local] new state Idle for Notify User 11
  961.  Extension Changed 11[ext-local] new state Idle for Notify User 10
  962.  Extension Changed 11[ext-local] new state Idle for Notify User 13
  963. Retransmitting #1 (NAT) to 192.168.1.17:5060:
  964. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  965. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0d3f4a98;rport
  966. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  967. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  968. Contact: <sip:6994998632@192.168.1.10>
  969. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  970. CSeq: 103 NOTIFY
  971. User-Agent: Asterisk PBX
  972. Max-Forwards: 70
  973. Event: refer;id=3847
  974. Subscription-state: active
  975. Content-Type: message/sipfrag;version=2.0
  976. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  977. Supported: replaces
  978. Content-Length: 21
  979.  
  980. SIP/2.0 183 Ringing
  981.  
  982. ---
  983. Retransmitting #1 (NAT) to 192.168.1.17:5060:
  984. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  985. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1f1f9743;rport
  986. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  987. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  988. Contact: <sip:6994998632@192.168.1.10>
  989. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  990. CSeq: 104 NOTIFY
  991. User-Agent: Asterisk PBX
  992. Max-Forwards: 70
  993. Event: refer;id=3847
  994. Subscription-state: terminated;reason=noresource
  995. Content-Type: message/sipfrag;version=2.0
  996. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  997. Supported: replaces
  998. Content-Length: 16
  999.  
  1000. SIP/2.0 200 Ok
  1001.  
  1002. ---
  1003. Retransmitting #1 (NAT) to 192.168.1.17:5060:
  1004. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  1005. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0bcf4d35;rport
  1006. From: "" <sip:12@192.168.1.10:5060>;tag=as01d3df4a
  1007. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=90f820dbe6
  1008. Contact: <sip:12@192.168.1.10>
  1009. Call-ID: 49f8da1dc614c738
  1010. CSeq: 132 NOTIFY
  1011. User-Agent: Asterisk PBX
  1012. Max-Forwards: 70
  1013. Event: dialog
  1014. Content-Type: application/dialog-info+xml
  1015. Subscription-State: active
  1016. Content-Length: 207
  1017.  
  1018. ?xml version="1.0"?>
  1019. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="30" state="full" entity="sip:12@192.168.1.10:5060">
  1020. <dialog id="12">
  1021. <state>terminated</state>
  1022. </dialog>
  1023. </dialog-info>
  1024.  
  1025. ---
  1026. Retransmitting #1 (NAT) to 192.168.1.17:5060:
  1027. NOTIFY sip:11@192.168.1.17:5060;transport=udp SIP/2.0
  1028. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK74415e27;rport
  1029. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  1030. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  1031. Contact: <sip:11@192.168.1.10>
  1032. Call-ID: 289210011a3e4b9a
  1033. CSeq: 164 NOTIFY
  1034. User-Agent: Asterisk PBX
  1035. Max-Forwards: 70
  1036. Event: dialog
  1037. Content-Type: application/dialog-info+xml
  1038. Subscription-State: active
  1039. Content-Length: 207
  1040.  
  1041. ?xml version="1.0"?>
  1042. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="62" state="full" entity="sip:11@192.168.1.10:5060">
  1043. <dialog id="11">
  1044. <state>terminated</state>
  1045. </dialog>
  1046. </dialog-info>
  1047.  
  1048. ---
  1049.  
  1050. <--- SIP read from 192.168.1.17:5060 --->
  1051. ACK sip:12@192.168.1.10:5060 SIP/2.0
  1052. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK715b891c4962c7082.9434695aa7547ae5a
  1053. Max-Forwards: 70
  1054. From: "Dining Room" <sip:11@192.168.1.10:5060>;tag=7e26487395
  1055. To: "12" <sip:12@192.168.1.10:5060>;tag=as5818ccd2
  1056. Call-ID: 6425e83c54a11bd9
  1057. CSeq: 1301 ACK
  1058. User-Agent: Aastra 6731i/2.6.0.2010
  1059. Content-Length: 0
  1060.  
  1061.  
  1062. <------------->
  1063. --- (9 headers 0 lines) ---
  1064.  
  1065. <--- SIP read from 192.168.1.17:5060 --->
  1066. SIP/2.0 200 OK
  1067. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0d3f4a98;rport=5060;received=192.168.1.10
  1068. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  1069. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  1070. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  1071. CSeq: 103 NOTIFY
  1072. Server: Aastra 6731i/2.6.0.2010
  1073. Content-Length: 0
  1074.  
  1075.  
  1076. <------------->
  1077. --- (8 headers 0 lines) ---
  1078. SIP Response message for INCOMING dialog NOTIFY arrived
  1079. Really destroying SIP dialog '6425e83c54a11bd9' Method: ACK
  1080.  
  1081. <--- SIP read from 192.168.1.17:5060 --->
  1082. SIP/2.0 200 OK
  1083. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1f1f9743;rport=5060;received=192.168.1.10
  1084. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  1085. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  1086. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  1087. CSeq: 104 NOTIFY
  1088. Server: Aastra 6731i/2.6.0.2010
  1089. Content-Length: 0
  1090.  
  1091.  
  1092. <------------->
  1093. --- (8 headers 0 lines) ---
  1094. SIP Response message for INCOMING dialog NOTIFY arrived
  1095.  
  1096. <--- SIP read from 192.168.1.17:5060 --->
  1097. BYE sip:6994998632@192.168.1.10 SIP/2.0
  1098. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK9b0933bbec7b39bf1.e63e2e108ca9714fb
  1099. Max-Forwards: 70
  1100. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  1101. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  1102. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  1103. CSeq: 3848 BYE
  1104. Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
  1105. Allow-Events: talk, hold, conference, aastra-xml, LocalModeStatus
  1106. Supported: path, timer
  1107. User-Agent: Aastra 6731i/2.6.0.2010
  1108. Content-Length: 0
  1109.  
  1110.  
  1111. <------------->
  1112. --- (12 headers 0 lines) ---
  1113. Sending to 192.168.1.17 : 5060 (NAT)
  1114. Scheduling destruction of SIP dialog '596d687d420f456e190086aa2434a05e@192.168.1.10' in 6400 ms (Method: BYE)
  1115.  
  1116. <--- Transmitting (NAT) to 192.168.1.17:5060 --->
  1117. SIP/2.0 200 OK
  1118. Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK9b0933bbec7b39bf1.e63e2e108ca9714fb;received=192.168.1.17
  1119. From: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  1120. To: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  1121. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  1122. CSeq: 3848 BYE
  1123. User-Agent: Asterisk PBX
  1124. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1125. Supported: replaces
  1126. Content-Length: 0
  1127.  
  1128.  
  1129. <------------>
  1130.  
  1131. <--- SIP read from 192.168.1.17:5060 --->
  1132. SIP/2.0 200 OK
  1133. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0d3f4a98;rport=5060;received=192.168.1.10
  1134. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  1135. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  1136. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  1137. CSeq: 103 NOTIFY
  1138. Server: Aastra 6731i/2.6.0.2010
  1139. Content-Length: 0
  1140.  
  1141.  
  1142. <------------->
  1143. --- (8 headers 0 lines) ---
  1144.  
  1145. <--- SIP read from 192.168.1.17:5060 --->
  1146. SIP/2.0 200 OK
  1147. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1f1f9743;rport=5060;received=192.168.1.10
  1148. From: "Doe John" <sip:6994998632@192.168.1.10>;tag=as5c36d89c
  1149. To: <sip:11@192.168.1.17:5060;transport=udp>;tag=1348055138
  1150. Call-ID: 596d687d420f456e190086aa2434a05e@192.168.1.10
  1151. CSeq: 104 NOTIFY
  1152. Server: Aastra 6731i/2.6.0.2010
  1153. Content-Length: 0
  1154.  
  1155.  
  1156. <------------->
  1157. --- (8 headers 0 lines) ---
  1158.  
  1159. <--- SIP read from 192.168.1.17:5060 --->
  1160. SIP/2.0 200 OK
  1161. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0bcf4d35;rport=5060;received=192.168.1.10
  1162. From: "" <sip:12@192.168.1.10:5060>;tag=as01d3df4a
  1163. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=90f820dbe6
  1164. Call-ID: 49f8da1dc614c738
  1165. CSeq: 132 NOTIFY
  1166. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  1167. Server: Aastra 6731i/2.6.0.2010
  1168. Content-Length: 0
  1169.  
  1170.  
  1171. <------------->
  1172. --- (9 headers 0 lines) ---
  1173. SIP Response message for INCOMING dialog NOTIFY arrived
  1174.  
  1175. <--- SIP read from 192.168.1.17:5060 --->
  1176. SIP/2.0 200 OK
  1177. Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK74415e27;rport=5060;received=192.168.1.10
  1178. From: "" <sip:11@192.168.1.10:5060>;tag=as6aab04ec
  1179. To: "Dining Room" <sip:11@192.168.1.10:5060>;tag=8cbdfec015
  1180. Call-ID: 289210011a3e4b9a
  1181. CSeq: 164 NOTIFY
  1182. Contact: "Dining Room" <sip:11@192.168.1.17:5060;transport=udp>
  1183. Server: Aastra 6731i/2.6.0.2010
  1184. Content-Length: 0
  1185.  
  1186.  
  1187. <------------->
  1188. --- (9 headers 0 lines) ---
  1189. SIP Response message for INCOMING dialog NOTIFY arrived
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