API tools faq
paste
Advertisement
Guest User

Asterisk Log

a guest
Aug 15th, 2021
108
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 19.19 KB | None | 0 0
raw download clone embed print report
  1. <--- Received SIP request (2553 bytes) from WSS: SOFTPHONE-WEBRTC-IP:28654 --->
  2. INVITE sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP SIP/2.0
  3. Via: SIP/2.0/WSS g53op9fbfst2.invalid;branch=z9hG4bK375721
  4. Max-Forwards: 69
  5. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>
  6. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  7. Call-ID: 7820gq85pou8l1jh168d
  8. CSeq: 3620 INVITE
  9. Contact: <sip:iu12r47f@g53op9fbfst2.invalid;transport=ws;ob>
  10. Content-Type: application/sdp
  11. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
  12. Supported: ice,replaces,outbound
  13. User-Agent: JsSIP 3.8.0
  14. Content-Length: 2023
  15. v=0
  16. o=- 7562586642801005627 2 IN IP4 127.0.0.1
  17. s=-
  18. t=0 0
  19. a=group:BUNDLE 0
  20. a=extmap-allow-mixed
  21. a=msid-semantic: WMS PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY
  22. m=audio 28657 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  23. c=IN IP4 SOFTPHONE-WEBRTC-IP
  24. a=rtcp:9 IN IP4 0.0.0.0
  25. a=candidate:427294303 1 udp 2122260223 192.168.100.12 50012 typ host generation 0 network-id 1
  26. a=candidate:3589980620 1 udp 1686052607 SOFTPHONE-WEBRTC-IP 28657 typ srflx raddr 192.168.100.12 rport 50012 generation 0 network-id 1
  27. a=candidate:1475609263 1 tcp 1518280447 192.168.100.12 9 typ host tcptype active generation 0 network-id 1
  28. a=ice-ufrag:rP9l
  29. a=ice-pwd:HTjR79Vi+IYYOEeKv5jDE6wZ
  30. a=ice-options:trickle
  31. a=fingerprint:sha-256 68:F2:1D:3B:18:AC:66:D2:D4:FD:1F:FF:84:59:4A:2E:7A:14:F6:37:7C:23:7E:9D:50:1A:B9:CE:6C:1C:60:99
  32. a=setup:actpass
  33. a=mid:0
  34. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  35. a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  36. a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
  37. a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
  38. a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
  39. a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
  40. a=sendrecv
  41. a=msid:PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY 8c098991-e0a1-4fd0-becb-bdda1c27161e
  42. a=rtcp-mux
  43. a=rtpmap:111 opus/48000/2
  44. a=rtcp-fb:111 transport-cc
  45. a=fmtp:111 minptime=10;useinbandfec=1
  46. a=rtpmap:103 ISAC/16000
  47. a=rtpmap:104 ISAC/32000
  48. a=rtpmap:9 G722/8000
  49. a=rtpmap:0 PCMU/8000
  50. a=rtpmap:8 PCMA/8000
  51. a=rtpmap:106 CN/32000
  52. a=rtpmap:105 CN/16000
  53. a=rtpmap:13 CN/8000
  54. a=rtpmap:110 telephone-event/48000
  55. a=rtpmap:112 telephone-event/32000
  56. a=rtpmap:113 telephone-event/16000
  57. a=rtpmap:126 telephone-event/8000
  58. a=ssrc:3650847381 cname:qVS8s/nc/RaBxHaX
  59. a=ssrc:3650847381 msid:PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY 8c098991-e0a1-4fd0-becb-bdda1c27161e
  60. a=ssrc:3650847381 mslabel:PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY
  61. a=ssrc:3650847381 label:8c098991-e0a1-4fd0-becb-bdda1c27161e
  62.  
  63. <--- Transmitting SIP response (492 bytes) to WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  64. SIP/2.0 401 Unauthorized
  65. Via: SIP/2.0/WSS g53op9fbfst2.invalid;rport=28654;received=SOFTPHONE-WEBRTC-IP;branch=z9hG4bK375721
  66. Call-ID: 7820gq85pou8l1jh168d
  67. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  68. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=z9hG4bK375721
  69. CSeq: 3620 INVITE
  70. WWW-Authenticate: Digest realm="asterisk",nonce="1629007576/2c863e3d90b53066c363ec4fae462477",opaque="3d305a040099ece0",algorithm=md5,qop="auth"
  71. Server: FPBX-15.0.17.43(16.19.1)
  72. Content-Length: 0
  73.  
  74. <--- Received SIP request (433 bytes) from WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  75. ACK sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP SIP/2.0
  76. Via: SIP/2.0/WSS g53op9fbfst2.invalid;branch=z9hG4bK375721
  77. Max-Forwards: 69
  78. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=z9hG4bK375721
  79. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  80. Call-ID: 7820gq85pou8l1jh168d
  81. CSeq: 3620 ACK
  82. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
  83. Supported: outbound
  84. User-Agent: JsSIP 3.8.0
  85. Content-Length: 0
  86.  
  87. <--- Received SIP request (2843 bytes) from WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  88. INVITE sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP SIP/2.0
  89. Via: SIP/2.0/WSS g53op9fbfst2.invalid;branch=z9hG4bK7183629
  90. Max-Forwards: 69
  91. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>
  92. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  93. Call-ID: 7820gq85pou8l1jh168d
  94. CSeq: 3621 INVITE
  95. Authorization: Digest algorithm=MD5, username="1000", realm="asterisk", nonce="1629007576/2c863e3d90b53066c363ec4fae462477", uri="sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP", response="5182fdcfeec18618d2d9fba6b6522bd8", opaque="3d305a040099ece0", qop
  96. =auth, cnonce="uc3c7von9nj6", nc=00000001
  97. Contact: <sip:iu12r47f@g53op9fbfst2.invalid;transport=ws;ob>
  98. Content-Type: application/sdp
  99. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
  100. Supported: ice,replaces,outbound
  101. User-Agent: JsSIP 3.8.0
  102. Content-Length: 2023
  103. v=0
  104. o=- 7562586642801005627 2 IN IP4 127.0.0.1
  105. s=-
  106. t=0 0
  107. a=group:BUNDLE 0
  108. a=extmap-allow-mixed
  109. a=msid-semantic: WMS PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY
  110. m=audio 28657 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  111. c=IN IP4 SOFTPHONE-WEBRTC-IP
  112. a=rtcp:9 IN IP4 0.0.0.0
  113. a=candidate:427294303 1 udp 2122260223 192.168.100.12 50012 typ host generation 0 network-id 1
  114. a=candidate:3589980620 1 udp 1686052607 SOFTPHONE-WEBRTC-IP 28657 typ srflx raddr 192.168.100.12 rport 50012 generation 0 network-id 1
  115. a=candidate:1475609263 1 tcp 1518280447 192.168.100.12 9 typ host tcptype active generation 0 network-id 1
  116. a=ice-ufrag:rP9l
  117. a=ice-pwd:HTjR79Vi+IYYOEeKv5jDE6wZ
  118. a=ice-options:trickle
  119. a=fingerprint:sha-256 68:F2:1D:3B:18:AC:66:D2:D4:FD:1F:FF:84:59:4A:2E:7A:14:F6:37:7C:23:7E:9D:50:1A:B9:CE:6C:1C:60:99
  120. a=setup:actpass
  121. a=mid:0
  122. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  123. a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  124. a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
  125. a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
  126. a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
  127. a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
  128. a=sendrecv
  129. a=msid:PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY 8c098991-e0a1-4fd0-becb-bdda1c27161e
  130. a=rtcp-mux
  131. a=rtpmap:111 opus/48000/2
  132. a=rtcp-fb:111 transport-cc
  133. a=fmtp:111 minptime=10;useinbandfec=1
  134. a=rtpmap:103 ISAC/16000
  135. a=rtpmap:104 ISAC/32000
  136. a=rtpmap:9 G722/8000
  137. a=rtpmap:0 PCMU/8000
  138. a=rtpmap:8 PCMA/8000
  139. a=rtpmap:106 CN/32000
  140. a=rtpmap:105 CN/16000
  141. a=rtpmap:13 CN/8000
  142. a=rtpmap:110 telephone-event/48000
  143. a=rtpmap:112 telephone-event/32000
  144. a=rtpmap:113 telephone-event/16000
  145. a=rtpmap:126 telephone-event/8000
  146. a=ssrc:3650847381 cname:qVS8s/nc/RaBxHaX
  147. a=ssrc:3650847381 msid:PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY 8c098991-e0a1-4fd0-becb-bdda1c27161e
  148. a=ssrc:3650847381 mslabel:PlAHENaiUpezYU34DFdGhHHvLhnriljx1sEY
  149. a=ssrc:3650847381 label:8c098991-e0a1-4fd0-becb-bdda1c27161e
  150.  
  151. <--- Transmitting SIP response (323 bytes) to WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  152. SIP/2.0 100 Trying
  153. Via: SIP/2.0/WSS g53op9fbfst2.invalid;rport=28654;received=SOFTPHONE-WEBRTC-IP;branch=z9hG4bK7183629
  154. Call-ID: 7820gq85pou8l1jh168d
  155. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  156. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>
  157. CSeq: 3621 INVITE
  158. Server: FPBX-15.0.17.43(16.19.1)
  159. Content-Length: 0
  160.  
  161. [2021-08-15 06:06:16] ERROR[7260]: res_pjsip_header_funcs.c:410 remove_header: No headers had been previously added to this session.
  162.  
  163. <--- Transmitting SIP request (987 bytes) to UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  164. INVITE sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP:5060 SIP/2.0
  165. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;rport;branch=z9hG4bKPj6521ba42-164f-4287-9628-e2fb11a10cb1
  166. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  167. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>
  168. Contact: <sip:asterisk@ASTERISK-SERVER-IP:5060>
  169. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  170. CSeq: 4919 INVITE
  171. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  172. Supported: 100rel, timer, replaces, norefersub, histinfo
  173. Session-Expires: 1800
  174. Min-SE: 90
  175. Max-Forwards: 70
  176. User-Agent: FPBX-15.0.17.43(16.19.1)
  177. Content-Type: application/sdp
  178. Content-Length: 263
  179. v=0
  180. o=- 529780655 529780655 IN IP4 ASTERISK-SERVER-IP
  181. s=Asterisk
  182. c=IN IP4 ASTERISK-SERVER-IP
  183. t=0 0
  184. m=audio 19564 RTP/AVP 8 0 101
  185. a=rtpmap:8 PCMA/8000
  186. a=rtpmap:0 PCMU/8000
  187. a=rtpmap:101 telephone-event/8000
  188. a=fmtp:101 0-16
  189. a=ptime:20
  190. a=maxptime:150
  191. a=sendrecv
  192.  
  193. <--- Received SIP response (394 bytes) from UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  194. SIP/2.0 100 Giving a try
  195. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;received=ASTERISK-SERVER-IP;rport=5060;branch=z9hG4bKPj6521ba42-164f-4287-9628-e2fb11a10cb1
  196. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  197. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>
  198. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  199. CSeq: 4919 INVITE
  200. Server: Balancer
  201. Content-Length: 0
  202.  
  203. <--- Received SIP response (797 bytes) from UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  204. SIP/2.0 180 Ringing
  205. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;received=ASTERISK-SERVER-IP;rport=5060;branch=z9hG4bKPj6521ba42-164f-4287-9628-e2fb11a10cb1
  206. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  207. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>;tag=fe5AEeGxNRE8E8bhsTURKBPEi2rMmQAq
  208. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  209. CSeq: 4919 INVITE
  210. Contact: <sip:924515541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP;did=5e9.a14e1113>
  211. Content-Type: application/sdp
  212. Server: TS-v4.8.1-17
  213. Content-Length: 265
  214. v=0
  215. o=- 1629007576 1629007576 IN IP4 177.53.17.27
  216. s=-
  217. c=IN IP4 177.53.17.27
  218. t=0 0
  219. m=audio 14000 RTP/AVP 8 0 101
  220. a=rtpmap:8 PCMA/8000
  221. a=rtpmap:0 PCMU/8000
  222. a=rtpmap:101 telephone-event/8000
  223. a=fmtp:101 0-15
  224. a=ptime:20
  225. a=sendrecv
  226. a=silenceSupp:off - - - -
  227.  
  228. <--- Transmitting SIP response (1481 bytes) to WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  229. SIP/2.0 183 Session Progress
  230. Via: SIP/2.0/WSS g53op9fbfst2.invalid;rport=28654;received=SOFTPHONE-WEBRTC-IP;branch=z9hG4bK7183629
  231. Call-ID: 7820gq85pou8l1jh168d
  232. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  233. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=6d5ab8a4-1023-40c3-9e7f-3b0d14cc259e
  234. CSeq: 3621 INVITE
  235. Server: FPBX-15.0.17.43(16.19.1)
  236. Contact: <sip:10.158.0.6:8089;transport=ws>
  237. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  238. P-Asserted-Identity: "CID:1000" <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>
  239. Content-Type: application/sdp
  240. Content-Length: 843
  241. v=0
  242. o=- 3141833787 4 IN IP4 10.158.0.6
  243. s=Asterisk
  244. c=IN IP4 10.158.0.6
  245. t=0 0
  246. a=group:BUNDLE 0
  247. m=audio 17662 UDP/TLS/RTP/SAVPF 8 0 126
  248. a=connection:new
  249. a=setup:active
  250. a=fingerprint:SHA-256 18:1D:D6:5E:B0:08:73:F5:1F:7E:73:3C:F3:FA:CE:4B:E3:A3:80:3F:9E:E1:34:67:C3:85:7B:09:77:3F:55:C7
  251. a=ice-ufrag:78d24d67080e64bb15abd11d03b7761d
  252. a=ice-pwd:38a401ea7cb430eb1a56068d0104b7e3
  253. a=candidate:Ha9e0006 1 UDP 2130706431 10.158.0.6 17662 typ host
  254. a=candidate:H47c9d1ea 1 UDP 2130706431 fe80::4001:aff:fe9e:6 17662 typ host
  255. a=candidate:S23d7f9d5 1 UDP 1694498815 ASTERISK-SERVER-IP 17662 typ srflx raddr 10.158.0.6 rport 17662
  256. a=rtpmap:8 PCMA/8000
  257. a=rtpmap:0 PCMU/8000
  258. a=rtpmap:126 telephone-event/8000
  259. a=fmtp:126 0-16
  260. a=ptime:20
  261. a=maxptime:150
  262. a=sendrecv
  263. a=rtcp-mux
  264. a=ssrc:510838881 cname:235af283-b171-4f11-8bed-86672ad6ee0e
  265. a=mid:0
  266.  
  267. <--- Received SIP response (497 bytes) from UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  268. SIP/2.0 180 Ringing
  269. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;received=ASTERISK-SERVER-IP;rport=5060;branch=z9hG4bKPj6521ba42-164f-4287-9628-e2fb11a10cb1
  270. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  271. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>;tag=fe5AEeGxNRE8E8bhsTURKBPEi2rMmQAq
  272. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  273. CSeq: 4919 INVITE
  274. Contact: <sip:924515541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP;did=5e9.a14e1113>
  275. Server: TS-v4.8.1-17
  276. Content-Length: 0
  277.  
  278. <--- Transmitting SIP response (1481 bytes) to WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  279. SIP/2.0 183 Session Progress
  280. Via: SIP/2.0/WSS g53op9fbfst2.invalid;rport=28654;received=SOFTPHONE-WEBRTC-IP;branch=z9hG4bK7183629
  281. Call-ID: 7820gq85pou8l1jh168d
  282. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  283. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=6d5ab8a4-1023-40c3-9e7f-3b0d14cc259e
  284. CSeq: 3621 INVITE
  285. Server: FPBX-15.0.17.43(16.19.1)
  286. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  287. Contact: <sip:10.158.0.6:8089;transport=ws>
  288. P-Asserted-Identity: "CID:1000" <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>
  289. Content-Type: application/sdp
  290. Content-Length: 843
  291. v=0
  292. o=- 3141833787 4 IN IP4 10.158.0.6
  293. s=Asterisk
  294. c=IN IP4 10.158.0.6
  295. t=0 0
  296. a=group:BUNDLE 0
  297. m=audio 17662 UDP/TLS/RTP/SAVPF 8 0 126
  298. a=connection:new
  299. a=setup:active
  300. a=fingerprint:SHA-256 18:1D:D6:5E:B0:08:73:F5:1F:7E:73:3C:F3:FA:CE:4B:E3:A3:80:3F:9E:E1:34:67:C3:85:7B:09:77:3F:55:C7
  301. a=ice-ufrag:78d24d67080e64bb15abd11d03b7761d
  302. a=ice-pwd:38a401ea7cb430eb1a56068d0104b7e3
  303. a=candidate:Ha9e0006 1 UDP 2130706431 10.158.0.6 17662 typ host
  304. a=candidate:H47c9d1ea 1 UDP 2130706431 fe80::4001:aff:fe9e:6 17662 typ host
  305. a=candidate:S23d7f9d5 1 UDP 1694498815 ASTERISK-SERVER-IP 17662 typ srflx raddr 10.158.0.6 rport 17662
  306. a=rtpmap:8 PCMA/8000
  307. a=rtpmap:0 PCMU/8000
  308. a=rtpmap:126 telephone-event/8000
  309. a=fmtp:126 0-16
  310. a=ptime:20
  311. a=maxptime:150
  312. a=sendrecv
  313. a=rtcp-mux
  314. a=ssrc:510838881 cname:235af283-b171-4f11-8bed-86672ad6ee0e
  315. a=mid:0
  316.  
  317. <--- Received SIP response (865 bytes) from UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  318. SIP/2.0 200 OK
  319. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;received=ASTERISK-SERVER-IP;rport=5060;branch=z9hG4bKPj6521ba42-164f-4287-9628-e2fb11a10cb1
  320. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  321. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>;tag=fe5AEeGxNRE8E8bhsTURKBPEi2rMmQAq
  322. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  323. CSeq: 4919 INVITE
  324. Contact: <sip:924515541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP;did=5e9.a14e1113>
  325. Content-Type: application/sdp
  326. Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE
  327. Server: TS-v4.8.1-17
  328. Content-Length: 265
  329. v=0
  330. o=- 1629007576 1629007576 IN IP4 177.53.17.27
  331. s=-
  332. c=IN IP4 177.53.17.27
  333. t=0 0
  334. m=audio 14000 RTP/AVP 8 0 101
  335. a=rtpmap:8 PCMA/8000
  336. a=rtpmap:0 PCMU/8000
  337. a=rtpmap:101 telephone-event/8000
  338. a=fmtp:101 0-15
  339. a=ptime:20
  340. a=sendrecv
  341. a=silenceSupp:off - - - -
  342.  
  343. <--- Transmitting SIP request (480 bytes) to UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  344. ACK sip:924515541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP;did=5e9.a14e1113 SIP/2.0
  345. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;rport;branch=z9hG4bKPj2fa9d7d3-04fc-41f6-9167-096e7fd2fa82
  346. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  347. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>;tag=fe5AEeGxNRE8E8bhsTURKBPEi2rMmQAq
  348. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  349. CSeq: 4919 ACK
  350. Max-Forwards: 70
  351. User-Agent: FPBX-15.0.17.43(16.19.1)
  352. Content-Length: 0
  353.  
  354. <--- Transmitting SIP response (1515 bytes) to WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  355. SIP/2.0 200 OK
  356. Via: SIP/2.0/WSS g53op9fbfst2.invalid;rport=28654;received=SOFTPHONE-WEBRTC-IP;branch=z9hG4bK7183629
  357. Call-ID: 7820gq85pou8l1jh168d
  358. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  359. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=6d5ab8a4-1023-40c3-9e7f-3b0d14cc259e
  360. CSeq: 3621 INVITE
  361. Server: FPBX-15.0.17.43(16.19.1)
  362. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  363. Contact: <sip:10.158.0.6:8089;transport=ws>
  364. Supported: 100rel, timer, replaces, norefersub
  365. P-Asserted-Identity: "CID:1000" <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>
  366. Content-Type: application/sdp
  367. Content-Length: 843
  368. v=0
  369. o=- 3141833787 4 IN IP4 10.158.0.6
  370. s=Asterisk
  371. c=IN IP4 10.158.0.6
  372. t=0 0
  373. a=group:BUNDLE 0
  374. m=audio 17662 UDP/TLS/RTP/SAVPF 8 0 126
  375. a=connection:new
  376. a=setup:active
  377. a=fingerprint:SHA-256 18:1D:D6:5E:B0:08:73:F5:1F:7E:73:3C:F3:FA:CE:4B:E3:A3:80:3F:9E:E1:34:67:C3:85:7B:09:77:3F:55:C7
  378. a=ice-ufrag:78d24d67080e64bb15abd11d03b7761d
  379. a=ice-pwd:38a401ea7cb430eb1a56068d0104b7e3
  380. a=candidate:Ha9e0006 1 UDP 2130706431 10.158.0.6 17662 typ host
  381. a=candidate:H47c9d1ea 1 UDP 2130706431 fe80::4001:aff:fe9e:6 17662 typ host
  382. a=candidate:S23d7f9d5 1 UDP 1694498815 ASTERISK-SERVER-IP 17662 typ srflx raddr 10.158.0.6 rport 17662
  383. a=rtpmap:8 PCMA/8000
  384. a=rtpmap:0 PCMU/8000
  385. a=rtpmap:126 telephone-event/8000
  386. a=fmtp:126 0-16
  387. a=ptime:20
  388. a=maxptime:150
  389. a=sendrecv
  390. a=rtcp-mux
  391. a=ssrc:510838881 cname:235af283-b171-4f11-8bed-86672ad6ee0e
  392. a=mid:0
  393.  
  394.  
  395. <--- Received SIP request (451 bytes) from WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  396. ACK sip:10.158.0.6:8089;transport=ws SIP/2.0
  397. Via: SIP/2.0/WSS g53op9fbfst2.invalid;branch=z9hG4bK4121085
  398. Max-Forwards: 69
  399. To: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=6d5ab8a4-1023-40c3-9e7f-3b0d14cc259e
  400. From: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  401. Call-ID: 7820gq85pou8l1jh168d
  402. CSeq: 3621 ACK
  403. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
  404. Supported: outbound
  405. User-Agent: JsSIP 3.8.0
  406. Content-Length: 0
  407.  
  408. <--- Transmitting SIP request (504 bytes) to UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  409. BYE sip:924515541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP;did=5e9.a14e1113 SIP/2.0
  410. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;rport;branch=z9hG4bKPjbc65a0ef-c417-43e1-ad98-4beaa5597d24
  411. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  412. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>;tag=fe5AEeGxNRE8E8bhsTURKBPEi2rMmQAq
  413. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  414. CSeq: 4920 BYE
  415. Reason: Q.850;cause=16
  416. Max-Forwards: 70
  417. User-Agent: FPBX-15.0.17.43(16.19.1)
  418. Content-Length: 0
  419.  
  420.  
  421. <--- Received SIP response (489 bytes) from UDP:SIP-OUTBUND-TRUNK-IP:5060 --->
  422. SIP/2.0 200 OK
  423. Via: SIP/2.0/UDP ASTERISK-SERVER-IP:5060;received=ASTERISK-SERVER-IP;rport=5060;branch=z9hG4bKPjbc65a0ef-c417-43e1-ad98-4beaa5597d24
  424. From: "Kelvin - webrtc" <sip:1000@10.158.0.6>;tag=4784162e-4872-41f0-83d9-10c3c30a6996
  425. To: <sip:10005541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP>;tag=fe5AEeGxNRE8E8bhsTURKBPEi2rMmQAq
  426. Call-ID: 1bfb0256-2c31-4703-bef2-ccbc62300244
  427. CSeq: 4920 BYE
  428. Contact: <sip:924515541xxxxxxxxxNUMBER@SIP-OUTBUND-TRUNK-IP;did=5e9.a14e1113>
  429. Server: TS-v4.8.1-17
  430. Content-Length: 0
  431.  
  432. [2021-08-15 06:06:36] ERROR[7412]: manager.c:6632 process_message: Unable to process manager action 'login'. Asterisk is shutting down.
  433. [2021-08-15 06:06:36] ERROR[7412]: manager.c:6632 process_message: Unable to process manager action 'Logoff'. Asterisk is shutting down.
  434.  
  435. <--- Transmitting SIP request (453 bytes) to WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  436. BYE sip:iu12r47f@SOFTPHONE-WEBRTC-IP:28654;transport=ws;ob SIP/2.0
  437. Via: SIP/2.0/WSS 10.158.0.6:8089;rport;branch=z9hG4bKPj2c487d3a-0c8a-409c-b455-41856568f154;alias
  438. From: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=6d5ab8a4-1023-40c3-9e7f-3b0d14cc259e
  439. To: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  440. Call-ID: 7820gq85pou8l1jh168d
  441. CSeq: 12628 BYE
  442. Reason: Q.850;cause=16
  443. Max-Forwards: 70
  444. User-Agent: FPBX-15.0.17.43(16.19.1)
  445. Content-Length: 0
  446.  
  447. <--- Received SIP response (346 bytes) from WSS:SOFTPHONE-WEBRTC-IP:28654 --->
  448. SIP/2.0 200 OK
  449. Via: SIP/2.0/WSS 10.158.0.6:8089;rport;branch=z9hG4bKPj2c487d3a-0c8a-409c-b455-41856568f154;alias
  450. To: <sip:1000@ASTERISK-SERVER-IP>;tag=90aqr98670
  451. From: <sip:5541xxxxxxxxxNUMBER@ASTERISK-SERVER-IP>;tag=6d5ab8a4-1023-40c3-9e7f-3b0d14cc259e
  452. Call-ID: 7820gq85pou8l1jh168d
  453. CSeq: 12628 BYE
  454. Supported: outbound
  455. Content-Length: 0
  456.  
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement
Public Pastes
Advertisement
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!