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Jun 14th, 2011
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  1. Then I reload asterisk from CLI and do the following
  2.  
  3.  
  4. Asterisk18*CLI> sip show peers
  5. Name/username Host Dyn Forcerport ACL Port Status Realtime
  6. 173.xx.xx.63 173.xx.xx.63 5060 Unmonitored
  7. 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline]
  8.  
  9. Asterisk18*CLI> core show channels
  10. Channel Location State Application(Data)
  11. 0 active channels
  12. 0 active calls
  13. 36 calls processed
  14.  
  15. Asterisk18*CLI> queue show
  16. irock.com has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime, 172s talktime), W:0, C:14, A:16, SL:0.0% within 0s
  17. Members:
  18. SIP/9013XX9XX8 with penalty 10 (dynamic) (Not in use) has taken 14 calls (last was 118 secs ago)
  19. No Callers
  20.  
  21.  
  22.  
  23.  
  24.  
  25. Then I call from PSTN phone and call is GOOD and the agent gets the INVITE
  26.  
  27. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  28. INVITE sip:[email protected] SIP/2.0
  29. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=da.26eb33d4;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  30. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKd7f.570082f.0
  31. Via: SIP/2.0/UDP 173.XX.XX.88;branch=z9hG4bKd7f.97ec4c72.0
  32. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  33. To: <sip:[email protected]>
  34. CSeq: 103 INVITE
  35. Call-ID: B2B.365.4070722
  36. Content-Length: 334
  37. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux))
  38. Content-Type: application/sdp
  39. Supported: replaces
  40. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  41. Max-Forwards: 68
  42. Contact: <sip:173.XX.XX.88:5060;transport=udp>
  43. P-hint: outbound->inbound
  44. P-hint: Route[6]: mediaproxy
  45.  
  46. v=0
  47. o=root 15753 15753 IN IP4 64.2.142.15
  48. s=session
  49. c=IN IP4 173.xx.xx.111
  50. t=0 0
  51. m=audio 2822 RTP/AVP 0 8 3 18 101
  52. a=rtpmap:0 PCMU/8000
  53. a=rtpmap:8 PCMA/8000
  54. a=rtpmap:3 GSM/8000
  55. a=rtpmap:18 G729/8000
  56. a=fmtp:18 annexb=no
  57. a=rtpmap:101 telephone-event/8000
  58. a=fmtp:101 0-16
  59. a=silenceSupp:off - - - -
  60. a=ptime:20
  61. a=sendrecv
  62. <------------->
  63. --- (17 headers 16 lines) ---
  64. Sending to 173.xx.xx.107:5060 (no NAT)
  65. Using INVITE request as basis request - B2B.365.4070722
  66. No matching peer for '9014XX7XX9' from '173.xx.xx.107:5060'
  67. == Using SIP RTP CoS mark 5
  68. Found RTP audio format 0
  69. Found RTP audio format 8
  70. Found RTP audio format 3
  71. Found RTP audio format 18
  72. Found RTP audio format 101
  73. Found audio description format PCMU for ID 0
  74. Found audio description format PCMA for ID 8
  75. Found audio description format GSM for ID 3
  76. Found audio description format G729 for ID 18
  77. Found audio description format telephone-event for ID 101
  78. Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  79. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  80. Peer audio RTP is at port 173.xx.xx.111:2822
  81. Looking for 9012XX1XX1 in default (domain irock.com)
  82. list_route: hop: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=da.26eb33d4;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  83.  
  84. <--- Transmitting (no NAT) to 173.xx.xx.107:5060 --->
  85. SIP/2.0 100 Trying
  86. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKd7f.570082f.0;received=173.xx.xx.107
  87. Via: SIP/2.0/UDP 173.XX.XX.88;branch=z9hG4bKd7f.97ec4c72.0
  88. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=da.26eb33d4;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  89. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  90. To: <sip:[email protected]>
  91. Call-ID: B2B.365.4070722
  92. CSeq: 103 INVITE
  93. Server: Asterisk PBX 1.8.4.2
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  95. Supported: replaces, timer
  96. Contact: <sip:[email protected]:5060>
  97. Content-Length: 0
  98.  
  99.  
  100. <------------>
  101. -- Executing [9012XX1XX1@default:1] Answer("SIP/64.2.142.15-00000032", "") in new stack
  102. Audio is at 5060
  103. Adding codec 0x2 (gsm) to SDP
  104. Adding codec 0x4 (ulaw) to SDP
  105. Adding codec 0x8 (alaw) to SDP
  106. Adding non-codec 0x1 (telephone-event) to SDP
  107.  
  108. <--- Reliably Transmitting (no NAT) to 173.xx.xx.107:5060 --->
  109. SIP/2.0 200 OK
  110. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKd7f.570082f.0;received=173.xx.xx.107
  111. Via: SIP/2.0/UDP 173.XX.XX.88;branch=z9hG4bKd7f.97ec4c72.0
  112. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=da.26eb33d4;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  113. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  114. To: <sip:[email protected]>;tag=as6aec3303
  115. Call-ID: B2B.365.4070722
  116. CSeq: 103 INVITE
  117. Server: Asterisk PBX 1.8.4.2
  118. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  119. Supported: replaces, timer
  120. Contact: <sip:[email protected]:5060>
  121. Content-Type: application/sdp
  122. Content-Length: 310
  123.  
  124. v=0
  125. o=root 244130491 244130491 IN IP4 173.xx.xx.63
  126. s=Asterisk PBX 1.8.4.2
  127. c=IN IP4 173.xx.xx.63
  128. t=0 0
  129. m=audio 18066 RTP/AVP 3 0 8 101
  130. a=rtpmap:3 GSM/8000
  131. a=rtpmap:0 PCMU/8000
  132. a=rtpmap:8 PCMA/8000
  133. a=rtpmap:101 telephone-event/8000
  134. a=fmtp:101 0-16
  135. a=silenceSupp:off - - - -
  136. a=ptime:20
  137. a=sendrecv
  138.  
  139. <------------>
  140.  
  141. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  142. ACK sip:[email protected]:5060 SIP/2.0
  143. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b>
  144. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKd7f.570082f.2
  145. Via: SIP/2.0/UDP 173.XX.XX.88;branch=z9hG4bKd7f.a7ec4c72.0
  146. To: <sip:[email protected]>;tag=as6aec3303
  147. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  148. CSeq: 103 ACK
  149. Call-ID: B2B.365.4070722
  150. Content-Length: 0
  151. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux))
  152. Max-Forwards: 69
  153. Contact: <sip:173.XX.XX.88:5060;transport=udp>
  154.  
  155. <------------->
  156. --- (12 headers 0 lines) ---
  157. -- Executing [9012XX1XX1@default:2] Queue("SIP/64.2.142.15-00000032", "irock.com,tT,,,300") in new stack
  158. -- Started music on hold, class 'default', on SIP/64.2.142.15-00000032
  159. > doing dnsmgr_lookup for 'ae.com'
  160. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: NOTIFY)
  161. Reliably Transmitting (NAT) to 173.xx.xx.107:5060:
  162. NOTIFY sip:[email protected]:5060 SIP/2.0
  163. Via: SIP/2.0/UDP 173.xx.xx.63:5060;branch=z9hG4bK141fe0e4;rport
  164. Max-Forwards: 70
  165. From: "asterisk" <sip:[email protected]>;tag=as2f10523c
  166. To: <sip:[email protected]:5060>
  167. Contact: <sip:[email protected]:5060>
  168. Call-ID: [email protected]
  169. CSeq: 102 NOTIFY
  170. User-Agent: Asterisk PBX 1.8.4.2
  171. Event: message-summary
  172. Content-Type: application/simple-message-summary
  173. Content-Length: 89
  174.  
  175. Messages-Waiting: no
  176. Message-Account: sip:[email protected]
  177. Voice-Message: 0/0 (0/0)
  178.  
  179. ---
  180. == Using SIP RTP CoS mark 5
  181. Audio is at 5060
  182.  
  183. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  184. SIP/2.0 405 Method Not Allowed
  185. Via: SIP/2.0/UDP 173.xx.xx.63:5060;branch=z9hG4bK141fe0e4;rport=5060
  186. From: "asterisk" <sip:[email protected]>;tag=as2f10523c
  187. To: <sip:[email protected]:5060>;tag=90ecd6ebfaa7dfa0f1f8f06360e22e19.902c
  188. Call-ID: [email protected]
  189. CSeq: 102 NOTIFY
  190. Server: ae SIP Proxy
  191. Content-Length: 0
  192.  
  193. <------------->
  194. --- (8 headers 0 lines) ---
  195. [Jun 14 17:04:10] WARNING[1390]: chan_sip.c:20140 handle_response: Host '173.xx.xx.107:5060' does not implement 'NOTIFY'
  196. Adding codec 0x4 (ulaw) to SDP
  197. Adding codec 0x2 (gsm) to SDP
  198. Adding codec 0x8 (alaw) to SDP
  199. Adding codec 0x800000000000 (testlaw) to SDP
  200. Adding non-codec 0x1 (telephone-event) to SDP
  201. Reliably Transmitting (NAT) to 173.xx.xx.107:5060:
  202. INVITE sip:[email protected]:5060 SIP/2.0
  203. Via: SIP/2.0/UDP 173.xx.xx.63:5060;branch=z9hG4bK551af5a5;rport
  204. Max-Forwards: 70
  205. From: "9014XX7XX9" <sip:[email protected]>;tag=as43504b15
  206. To: <sip:[email protected]:5060>
  207. Contact: <sip:[email protected]:5060>
  208. Call-ID: [email protected]
  209. CSeq: 102 INVITE
  210. User-Agent: Asterisk PBX 1.8.4.2
  211. Date: Tue, 14 Jun 2011 22:04:10 GMT
  212. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  213. Supported: replaces, timer
  214. Content-Type: application/sdp
  215. Content-Length: 310
  216.  
  217. v=0
  218. o=root 839867384 839867384 IN IP4 173.xx.xx.63
  219. s=Asterisk PBX 1.8.4.2
  220. c=IN IP4 173.xx.xx.63
  221. t=0 0
  222. m=audio 16658 RTP/AVP 0 3 8 101
  223. a=rtpmap:0 PCMU/8000
  224. a=rtpmap:3 GSM/8000
  225. a=rtpmap:8 PCMA/8000
  226. a=rtpmap:101 telephone-event/8000
  227. a=fmtp:101 0-16
  228. a=silenceSupp:off - - - -
  229. a=ptime:20
  230. a=sendrecv
  231.  
  232. ---
  233.  
  234. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  235. SIP/2.0 100 Giving a try
  236. Via: SIP/2.0/UDP 173.xx.xx.63:5060;branch=z9hG4bK551af5a5;rport=5060
  237. From: "9014XX7XX9" <sip:[email protected]>;tag=as43504b15
  238. To: <sip:[email protected]:5060>
  239. Call-ID: [email protected]
  240. CSeq: 102 INVITE
  241. Server: ae SIP Proxy
  242. Content-Length: 0
  243.  
  244. <------------->
  245. --- (8 headers 0 lines) ---
  246.  
  247. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  248. SIP/2.0 180 Ringing
  249. Via: SIP/2.0/UDP 173.xx.xx.63:5060;rport=5060;received=173.xx.xx.63;branch=z9hG4bK551af5a5
  250. Record-Route: <sip:173.xx.xx.107;lr;ftag=as43504b15;did=263.96e1cc7;vst=AAAAAAAAAAAAAAAAAAAACBcbCw5cAAAAbXVuaWNhdGlvbnMuY29tOjUwNjA->
  251. Call-ID: [email protected]
  252. From: "9014XX7XX9" <sip:[email protected]>;tag=as43504b15
  253. To: <sip:[email protected]:5060>;tag=732c1eb145cc4313972b46bf012cd9aa
  254. CSeq: 102 INVITE
  255. Server: Blink 0.2.7 (Windows)
  256. Contact: <sip:[email protected]:57111>
  257. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
  258. Content-Length: 0
  259. P-hint: Onreply-route - fixcontact
  260.  
  261. <------------->
  262. --- (12 headers 0 lines) ---
  263. -- SIP/9013XX9XX8-00000033 is ringing
  264.  
  265. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  266. SIP/2.0 200 OK
  267. Via: SIP/2.0/UDP 173.xx.xx.63:5060;rport=5060;received=173.xx.xx.63;branch=z9hG4bK551af5a5
  268. Record-Route: <sip:173.xx.xx.107;lr;ftag=as43504b15;did=263.96e1cc7;vst=AAAAAAAAAAAAAAAAAAAACBcbCw5cAAAAbXVuaWNhdGlvbnMuY29tOjUwNjA->
  269. Call-ID: [email protected]
  270. From: "9014XX7XX9" <sip:[email protected]>;tag=as43504b15
  271. To: <sip:[email protected]:5060>;tag=732c1eb145cc4313972b46bf012cd9aa
  272. CSeq: 102 INVITE
  273. Server: Blink 0.2.7 (Windows)
  274. Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
  275. Contact: <sip:[email protected]:57111>
  276. Supported: 100rel, norefersub
  277. Content-Type: application/sdp
  278. Content-Length: 238
  279. P-hint: Onreply-route - fixcontact
  280.  
  281. v=0
  282. o=- 3517059799 3517059800 IN IP4 64.132.245.122
  283. s=Blink 0.2.7 (Windows)
  284. c=IN IP4 173.xx.xx.111
  285. t=0 0
  286. m=audio 2824 RTP/AVP 0 101
  287. a=rtcp:2825
  288. a=rtpmap:0 PCMU/8000
  289. a=rtpmap:101 telephone-event/8000
  290. a=fmtp:101 0-15
  291. a=sendrecv
  292. <------------->
  293. --- (14 headers 11 lines) ---
  294. Found RTP audio format 0
  295. Found RTP audio format 101
  296. Found audio description format PCMU for ID 0
  297. Found audio description format telephone-event for ID 101
  298. Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  299. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  300. Peer audio RTP is at port 173.xx.xx.111:2824
  301. list_route: hop: <sip:173.xx.xx.107;lr;ftag=as43504b15;did=263.96e1cc7;vst=AAAAAAAAAAAAAAAAAAAACBcbCw5cAAAAbXVuaWNhdGlvbnMuY29tOjUwNjA->
  302. set_destination: Parsing <sip:173.xx.xx.107;lr;ftag=as43504b15;did=263.96e1cc7;vst=AAAAAAAAAAAAAAAAAAAACBcbCw5cAAAAbXVuaWNhdGlvbnMuY29tOjUwNjA-> for address/port to send to
  303. set_destination: set destination to 173.xx.xx.107:5060
  304. Transmitting (NAT) to 173.xx.xx.107:5060:
  305. ACK sip:[email protected]:57111 SIP/2.0
  306. Via: SIP/2.0/UDP 173.xx.xx.63:5060;branch=z9hG4bK50c9b88f;rport
  307. Route: <sip:173.xx.xx.107;lr;ftag=as43504b15;did=263.96e1cc7;vst=AAAAAAAAAAAAAAAAAAAACBcbCw5cAAAAbXVuaWNhdGlvbnMuY29tOjUwNjA->
  308. Max-Forwards: 70
  309. From: "9014XX7XX9" <sip:[email protected]>;tag=as43504b15
  310. To: <sip:[email protected]:5060>;tag=732c1eb145cc4313972b46bf012cd9aa
  311. Contact: <sip:[email protected]:5060>
  312. Call-ID: [email protected]
  313. CSeq: 102 ACK
  314. User-Agent: Asterisk PBX 1.8.4.2
  315. Content-Length: 0
  316.  
  317.  
  318. ---
  319. -- SIP/9013XX9XX8-00000033 connected line has changed. Saving it until answer for SIP/64.2.142.15-00000032
  320. -- SIP/9013XX9XX8-00000033 answered SIP/64.2.142.15-00000032
  321. -- Stopped music on hold on SIP/64.2.142.15-00000032
  322.  
  323.  
  324.  
  325.  
  326.  
  327.  
  328. While the agent is on the call with the caller I do the following commands on the CLI
  329.  
  330. Asterisk18*CLI> sip show peers
  331. Name/username Host Dyn Forcerport ACL Port Status Realtime
  332. 173.xx.xx.63 173.xx.xx.63 5060 Unmonitored
  333. 9013XX9XX8/9013XX9XX8 173.xx.xx.107 N 5060 Unmonitored Cached RT
  334. 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
  335. Asterisk18*CLI> core show channels
  336. Channel Location State Application(Data)
  337. SIP/9013XX9XX8-00000 [email protected] Up AppQueue((Outgoing Line))
  338. SIP/64.2.142.15-0000 9012XX1XX1@default:2 Up Queue(irock.com,tT,,,300)
  339. 2 active channels
  340. 1 active call
  341. 38 calls processed
  342. Asterisk18*CLI> queue show
  343. irock.com has 0 calls (max unlimited) in 'ringall' strategy (3s holdtime, 141s talktime), W:0, C:15, A:16, SL:0.0% within 0s
  344. Members:
  345. SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 15 calls (last was 117 secs ago)
  346. No Callers
  347.  
  348.  
  349.  
  350.  
  351.  
  352.  
  353.  
  354. Now after I hang up and do the CLI commands again it looks like
  355.  
  356. Asterisk18*CLI> queue show
  357. irock.com has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime, 141s talktime), W:0, C:15, A:16, SL:0.0% within 0s
  358. Members:
  359. SIP/9013XX9XX8 with penalty 10 (dynamic) (Not in use) has taken 15 calls (last was 6 secs ago)
  360. No Callers
  361.  
  362. Asterisk18*CLI> core show channels
  363. Channel Location State Application(Data)
  364. 0 active channels
  365. 0 active calls
  366. 37 calls processed
  367.  
  368. Asterisk18*CLI> sip show peers
  369. Name/username Host Dyn Forcerport ACL Port Status Realtime
  370. 173.xx.xx.63 173.xx.xx.63 5060 Unmonitored
  371. 9013XX9XX8/9013XX9XX8 173.xx.xx.107 N 5060 Unmonitored Cached RT
  372. 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
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