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  1. <--- Transmitting (NAT) to 84.43.174.154:1024 --->
  2. SIP/2.0 100 Trying
  3. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  4. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  5. To: <sip:7985271@91.215.56.15>
  6. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  7. CSeq: 271 INVITE
  8. Server: Asterisk PBX 1.8.23.1
  9. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  10. Supported: replaces, timer
  11. Session-Expires: 1800;refresher=uas
  12. Contact: <sip:7985271@91.215.56.15:5060>
  13. Content-Length: 0
  14.  
  15.  
  16. <------------>
  17. -- Executing [7985271@remote-mg:1] Set("SIP/7400505-0006c784", "CALLERID(num)=487400505") in new stack
  18. -- Executing [7985271@remote-mg:2] Dial("SIP/7400505-0006c784", "dahdi/g0/7985271,90,t") in new stack
  19. -- Requested transfer capability: 0x00 - SPEECH
  20. -- Called dahdi/g0/7985271
  21. -- DAHDI/i1/7985271-7fba9 is proceeding passing it to SIP/7400505-0006c784
  22.  
  23. <--- Transmitting (NAT) to 84.43.174.154:1024 --->
  24. SIP/2.0 100 Trying
  25. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  26. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  27. To: <sip:7985271@91.215.56.15>
  28. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  29. CSeq: 271 INVITE
  30. Server: Asterisk PBX 1.8.23.1
  31. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  32. Supported: replaces, timer
  33. Session-Expires: 1800;refresher=uas
  34. Contact: <sip:7985271@91.215.56.15:5060>
  35. Content-Length: 0
  36.  
  37.  
  38. <------------>
  39.  
  40. <--- SIP read from UDP:84.43.174.154:1024 --->
  41. ACK sip:7985271@91.215.56.15 SIP/2.0
  42. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK779521023;rport
  43. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  44. To: <sip:7985271@91.215.56.15>;tag=as64ade5e0
  45. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  46. CSeq: 270 ACK
  47. Content-Length: 0
  48.  
  49.  
  50. <------------->
  51. --- (7 headers 0 lines) ---
  52. -- Accepting call from '487874717' to '7070018' on channel 0/5, span 1
  53. -- Executing [7070018@from-pstn:1] Set("DAHDI/i1/487874717-7fbaa", "CALLERID(num)=0487874717") in new stack
  54. -- Executing [7070018@from-pstn:2] Dial("DAHDI/i1/487874717-7fbaa", "SIP/7070018,30,tr") in new stack
  55. == Using SIP RTP CoS mark 5
  56. -- Called SIP/7070018
  57. -- SIP/7070018-0006c785 is ringing
  58. -- DAHDI/i1/7985271-7fba9 is ringing
  59.  
  60. <--- Transmitting (NAT) to 84.43.174.154:1024 --->
  61. SIP/2.0 180 Ringing
  62. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  63. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  64. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  65. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  66. CSeq: 271 INVITE
  67. Server: Asterisk PBX 1.8.23.1
  68. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  69. Supported: replaces, timer
  70. Content-Length: 0
  71.  
  72.  
  73. <------------>
  74. -- Accepting call from '630385245' to '7070007' on channel 0/20, span 1
  75. -- Executing [7070007@from-pstn:1] Set("DAHDI/i1/630385245-7fbab", "CALLERID(num)=0630385245") in new stack
  76. -- Executing [7070007@from-pstn:2] Dial("DAHDI/i1/630385245-7fbab", "SIP/7070007,30,tr") in new stack
  77. == Using SIP RTP CoS mark 5
  78. -- Called SIP/7070007
  79. -- SIP/7070007-0006c786 is making progress passing it to DAHDI/i1/630385245-7fbab
  80. -- SIP/7070007-0006c786 answered DAHDI/i1/630385245-7fbab
  81. -- Hungup 'DAHDI/i1/0683848642-7fba8'
  82. == Spawn extension (remote-mgmn, 0683848642, 2) exited non-zero on 'SIP/7070007-0006c783'
  83. -- DAHDI/i1/7985271-7fba9 answered SIP/7400505-0006c784
  84. Audio is at 10248
  85. Adding codec 0x8 (alaw) to SDP
  86. Adding codec 0x4 (ulaw) to SDP
  87. Adding codec 0x100 (g729) to SDP
  88.  
  89. <--- Reliably Transmitting (NAT) to 84.43.174.154:1024 --->
  90. SIP/2.0 200 OK
  91. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  92. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  93. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  94. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  95. CSeq: 271 INVITE
  96. Server: Asterisk PBX 1.8.23.1
  97. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  98. Supported: replaces, timer
  99. Content-Type: application/sdp
  100. Content-Length: 250
  101.  
  102. v=0
  103. o=root 277977711 277977711 IN IP4 91.215.56.15
  104. s=Asterisk PBX 1.8.23.1
  105. c=IN IP4 91.215.56.15
  106. t=0 0
  107. m=audio 10248 RTP/AVP 8 0 18
  108. a=rtpmap:8 PCMA/8000
  109. a=rtpmap:0 PCMU/8000
  110. a=rtpmap:18 G729/8000
  111. a=fmtp:18 annexb=no
  112. a=ptime:20
  113. a=sendrecv
  114.  
  115. <------------>
  116. Retransmitting #1 (NAT) to 84.43.174.154:1024:
  117. SIP/2.0 200 OK
  118. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  119. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  120. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  121. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  122. CSeq: 271 INVITE
  123. Server: Asterisk PBX 1.8.23.1
  124. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  125. Supported: replaces, timer
  126. Content-Type: application/sdp
  127. Content-Length: 250
  128.  
  129. v=0
  130. o=root 277977711 277977711 IN IP4 91.215.56.15
  131. s=Asterisk PBX 1.8.23.1
  132. c=IN IP4 91.215.56.15
  133. t=0 0
  134. m=audio 10248 RTP/AVP 8 0 18
  135. a=rtpmap:8 PCMA/8000
  136. a=rtpmap:0 PCMU/8000
  137. a=rtpmap:18 G729/8000
  138. a=fmtp:18 annexb=no
  139. a=ptime:20
  140. a=sendrecv
  141.  
  142. ---
  143. Retransmitting #2 (NAT) to 84.43.174.154:1024:
  144. SIP/2.0 200 OK
  145. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  146. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  147. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  148. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  149. CSeq: 271 INVITE
  150. Server: Asterisk PBX 1.8.23.1
  151. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  152. Supported: replaces, timer
  153. Content-Type: application/sdp
  154. Content-Length: 250
  155.  
  156. v=0
  157. o=root 277977711 277977711 IN IP4 91.215.56.15
  158. s=Asterisk PBX 1.8.23.1
  159. c=IN IP4 91.215.56.15
  160. t=0 0
  161. m=audio 10248 RTP/AVP 8 0 18
  162. a=rtpmap:8 PCMA/8000
  163. a=rtpmap:0 PCMU/8000
  164. a=rtpmap:18 G729/8000
  165. a=fmtp:18 annexb=no
  166. a=ptime:20
  167. a=sendrecv
  168.  
  169. ---
  170. Retransmitting #3 (NAT) to 84.43.174.154:1024:
  171. SIP/2.0 200 OK
  172. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  173. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  174. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  175. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  176. CSeq: 271 INVITE
  177. Server: Asterisk PBX 1.8.23.1
  178. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  179. Supported: replaces, timer
  180. Content-Type: application/sdp
  181. Content-Length: 250
  182.  
  183. v=0
  184. o=root 277977711 277977711 IN IP4 91.215.56.15
  185. s=Asterisk PBX 1.8.23.1
  186. c=IN IP4 91.215.56.15
  187. t=0 0
  188. m=audio 10248 RTP/AVP 8 0 18
  189. a=rtpmap:8 PCMA/8000
  190. a=rtpmap:0 PCMU/8000
  191. a=rtpmap:18 G729/8000
  192. a=fmtp:18 annexb=no
  193. a=ptime:20
  194. a=sendrecv
  195.  
  196. ---
  197. Retransmitting #4 (NAT) to 84.43.174.154:1024:
  198. SIP/2.0 200 OK
  199. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  200. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  201. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  202. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  203. CSeq: 271 INVITE
  204. Server: Asterisk PBX 1.8.23.1
  205. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  206. Supported: replaces, timer
  207. Content-Type: application/sdp
  208. Content-Length: 250
  209.  
  210. v=0
  211. o=root 277977711 277977711 IN IP4 91.215.56.15
  212. s=Asterisk PBX 1.8.23.1
  213. c=IN IP4 91.215.56.15
  214. t=0 0
  215. m=audio 10248 RTP/AVP 8 0 18
  216. a=rtpmap:8 PCMA/8000
  217. a=rtpmap:0 PCMU/8000
  218. a=rtpmap:18 G729/8000
  219. a=fmtp:18 annexb=no
  220. a=ptime:20
  221. a=sendrecv
  222.  
  223. ---
  224. Retransmitting #5 (NAT) to 84.43.174.154:1024:
  225. SIP/2.0 200 OK
  226. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  227. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  228. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  229. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  230. CSeq: 271 INVITE
  231. Server: Asterisk PBX 1.8.23.1
  232. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  233. Supported: replaces, timer
  234. Content-Type: application/sdp
  235. Content-Length: 250
  236.  
  237. v=0
  238. o=root 277977711 277977711 IN IP4 91.215.56.15
  239. s=Asterisk PBX 1.8.23.1
  240. c=IN IP4 91.215.56.15
  241. t=0 0
  242. m=audio 10248 RTP/AVP 8 0 18
  243. a=rtpmap:8 PCMA/8000
  244. a=rtpmap:0 PCMU/8000
  245. a=rtpmap:18 G729/8000
  246. a=fmtp:18 annexb=no
  247. a=ptime:20
  248. a=sendrecv
  249.  
  250. ---
  251. Retransmitting #6 (NAT) to 84.43.174.154:1024:
  252. SIP/2.0 200 OK
  253. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK897431282;received=84.43.174.154;rport=1024
  254. From: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  255. To: <sip:7985271@91.215.56.15>;tag=as789bc733
  256. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  257. CSeq: 271 INVITE
  258. Server: Asterisk PBX 1.8.23.1
  259. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  260. Supported: replaces, timer
  261. Content-Type: application/sdp
  262. Content-Length: 250
  263.  
  264. v=0
  265. o=root 277977711 277977711 IN IP4 91.215.56.15
  266. s=Asterisk PBX 1.8.23.1
  267. c=IN IP4 91.215.56.15
  268. t=0 0
  269. m=audio 10248 RTP/AVP 8 0 18
  270. a=rtpmap:8 PCMA/8000
  271. a=rtpmap:0 PCMU/8000
  272. a=rtpmap:18 G729/8000
  273. a=fmtp:18 annexb=no
  274. a=ptime:20
  275. a=sendrecv
  276.  
  277. ---
  278. [Jul 12 13:00:50] WARNING[1802]: chan_sip.c:3982 retrans_pkt: Retransmission timeout reached on transmission 309822855-5060-28@BJC.BGI.A.BAC for seqno 271 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  279. Packet timed out after 7295ms with no response
  280. [Jul 12 13:00:50] WARNING[1802]: chan_sip.c:4011 retrans_pkt: Hanging up call 309822855-5060-28@BJC.BGI.A.BAC - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  281. -- Hungx855-5060-28@BJC.BGI.A.BAC' in 7296 ms (Method: ACK)
  282. set_destination: Parsing <sip:7400505@192.168.0.102:5060> for address/port to send to
  283. set_destination: set destination to 192.168.0.102:5060
  284. Reliably Transmitting (NAT) to 84.43.174.154:1024:
  285. BYE sip:7400505@192.168.0.102:5060 SIP/2.0
  286. Via: SIP/2.0/UDP 91.215.56.15:5060;branch=z9hG4bK667a9374;rport
  287. Max-Forwards: 70
  288. From: <sip:7985271@91.215.56.15>;tag=as789bc733
  289. To: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  290. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  291. CSeq: 102 BYE
  292. User-Agent: Asterisk PBX 1.8.23.1
  293. Proxy-Authorization: Digest username="7400505", realm="asterisk", algorithm=MD5, uri="sip:91.215.56.15", nonce="", response="d1563cca565d558b679dd4a1b6af5453"
  294. X-Asterisk-HangupCause: No user responding
  295. X-Asterisk-HangupCauseCode: 18
  296. Content-Length: 0
  297.  
  298.  
  299. ---
  300. Retransmitting #1 (NAT) to 84.43.174.154:1024:
  301. BYE sip:7400505@192.168.0.102:5060 SIP/2.0
  302. Via: SIP/2.0/UDP 91.215.56.15:5060;branch=z9hG4bK667a9374;rport
  303. Max-Forwards: 70
  304. From: <sip:7985271@91.215.56.15>;tag=as789bc733
  305. To: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  306. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  307. CSeq: 102 BYE
  308. User-Agent: Asterisk PBX 1.8.23.1
  309. Proxy-Authorization: Digest username="7400505", realm="asterisk", algorithm=MD5, uri="sip:91.215.56.15", nonce="", response="d1563cca565d558b679dd4a1b6af5453"
  310. X-Asterisk-HangupCause: No user responding
  311. X-Asterisk-HangupCauseCode: 18
  312. Content-Length: 0
  313.  
  314.  
  315. ---
  316.  
  317. <--- SIP read from UDP:84.43.174.154:1024 --->
  318. SIP/2.0 481 Call Leg/Transaction Does Not Exist
  319. Via: SIP/2.0/UDP 91.215.56.15:5060;branch=z9hG4bK667a9374;rport=5060
  320. From: <sip:7985271@91.215.56.15>;tag=as789bc733
  321. To: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  322. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  323. CSeq: 102 BYE
  324. Supported: replaces, path, timer, eventlist
  325. User-Agent: Grandstream DP715 1.0.0.5
  326. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  327. Content-Length: 0
  328.  
  329.  
  330. <------------->
  331. --- (10 headers 0 lines) ---
  332. Really destroying SIP dialog '309822855-5060-28@BJC.BGI.A.BAC' Method: ACK
  333.  
  334. <--- SIP read from UDP:84.43.174.154:1024 --->
  335. SIP/2.0 481 Call Leg/Transaction Does Not Exist
  336. Via: SIP/2.0/UDP 91.215.56.15:5060;branch=z9hG4bK667a9374;rport=5060
  337. From: <sip:7985271@91.215.56.15>;tag=as789bc733
  338. To: "vols0505" <sip:7400505@91.215.56.15>;tag=2025457088
  339. Call-ID: 309822855-5060-28@BJC.BGI.A.BAC
  340. CSeq: 102 BYE
  341. Supported: replaces, path, timer, eventlist
  342. User-Agent: Grandstream DP715 1.0.0.5
  343. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  344. Content-Length: 0
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