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  1. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
  2. <--- SIP read from UDP:192.168.1.158:5060 --->
  3. INVITE sip:5098556587@192.168.1.12;user=phone SIP/2.0
  4. Via: SIP/2.0/UDP 192.168.1.158;branch=z9hG4bK9f763194460023C5
  5. From: "204" <sip:204@192.168.1.12>;tag=F94F8C7A-CB1BBD3B
  6. To: <sip:5098556587@192.168.1.12;user=phone>
  7. CSeq: 1 INVITE
  8. Call-ID: 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
  9. Contact: <sip:204@192.168.1.158>
  10. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  11. User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.1.0933
  12. Accept-Language: en
  13. Supported: 100rel,replaces
  14. Allow-Events: talk,hold,conference
  15. Max-Forwards: 70
  16. Content-Type: application/sdp
  17. Content-Length: 296
  18.  
  19. v=0
  20. o=- 1168133393 1168133393 IN IP4 192.168.1.158
  21. s=Polycom IP Phone
  22. c=IN IP4 192.168.1.158
  23. t=0 0
  24. a=sendrecv
  25. m=audio 2222 RTP/AVP 9 0 8 18 127
  26. a=rtpmap:9 G722/8000
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:18 G729/8000
  30. a=fmtp:18 annexb=no
  31. a=rtpmap:127 telephone-event/8000
  32. <------------->
  33. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: --- (15 headers 13 lines) ---
  34. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Sending to 192.168.1.158:5060 (NAT)
  35. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Using INVITE request as basis request - 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
  36. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found peer '204' for '204' from 192.168.1.158:5060
  37. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 9
  38. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 0
  39. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 8
  40. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 18
  41. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 127
  42. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format G722 for ID 9
  43. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format PCMU for ID 0
  44. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format PCMA for ID 8
  45. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format G729 for ID 18
  46. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format telephone-event for ID 127
  47. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  48. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
  49. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Peer audio RTP is at port 192.168.1.158:2222
  50. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Looking for 5098556587 in from-internal (domain 192.168.1.12)
  51. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: list_route: hop: <sip:204@192.168.1.158>
  52. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
  53. <--- Transmitting (no NAT) to 192.168.1.158:5060 --->
  54. SIP/2.0 100 Trying
  55. Via: SIP/2.0/UDP 192.168.1.158;branch=z9hG4bK9f763194460023C5;received=192.168.1.158
  56. From: "204" <sip:204@192.168.1.12>;tag=F94F8C7A-CB1BBD3B
  57. To: <sip:5098556587@192.168.1.12;user=phone>
  58. Call-ID: 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
  59. CSeq: 1 INVITE
  60. Server: FPBX-2.9.0(1.8.3.2)
  61. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  62. Supported: replaces, timer
  63. Contact: <sip:5098556587@192.168.1.12:5060>
  64. Content-Length: 0
  65.  
  66.  
  67. <------------>
  68. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  69. [2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Audio is at 5060
  70. [2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  71. [2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  72. [2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding codec 0x2 (gsm) to SDP
  73. [2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  74. [2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Reliably Transmitting (NAT) to 64.2.142.29:5060:
  75. INVITE sip:5098556587@outbound.vitelity.net SIP/2.0
  76. Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK542211aa;rport
  77. Max-Forwards: 70
  78. From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
  79. To: <sip:5098556587@outbound.vitelity.net>
  80. Contact: <sip:njfragent@67.158.158.25:5060>
  81. Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
  82. CSeq: 102 INVITE
  83. User-Agent: FPBX-2.9.0(1.8.3.2)
  84. Date: Wed, 14 Sep 2011 17:21:16 GMT
  85. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  86. Supported: replaces, timer
  87. Remote-Party-ID: "Hvac corp" <sip:5097682249@67.158.158.25>;party=calling;privacy=off;screen=no
  88. Content-Type: application/sdp
  89. Content-Length: 287
  90.  
  91. v=0
  92. o=root 1760196305 1760196305 IN IP4 67.158.158.25
  93. s=Asterisk PBX 1.8.3.2
  94. c=IN IP4 67.158.158.25
  95. t=0 0
  96. m=audio 11272 RTP/AVP 0 8 3 101
  97. a=rtpmap:0 PCMU/8000
  98. a=rtpmap:8 PCMA/8000
  99. a=rtpmap:3 GSM/8000
  100. a=rtpmap:101 telephone-event/8000
  101. a=fmtp:101 0-16
  102. a=ptime:20
  103. a=sendrecv
  104.  
  105. ---
  106. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  107. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  108. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  109. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  110. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  111. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
  112. <--- SIP read from UDP:64.2.142.29:5060 --->
  113. SIP/2.0 407 Proxy Authentication Required
  114. Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK542211aa;received=67.158.158.25;rport=37041
  115. From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
  116. To: <sip:5098556587@outbound.vitelity.net>;tag=as02c943ec
  117. Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
  118. CSeq: 102 INVITE
  119. User-Agent: Asterisk PBX
  120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  121. Supported: replaces
  122. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50fb97fb"
  123. Content-Length: 0
  124.  
  125. <------------->
  126. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: --- (11 headers 0 lines) ---
  127. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Transmitting (NAT) to 64.2.142.29:5060:
  128. ACK sip:5098556587@outbound.vitelity.net SIP/2.0
  129. Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK542211aa;rport
  130. Max-Forwards: 70
  131. From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
  132. To: <sip:5098556587@outbound.vitelity.net>;tag=as02c943ec
  133. Contact: <sip:njfragent@67.158.158.25:5060>
  134. Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
  135. CSeq: 102 ACK
  136. User-Agent: FPBX-2.9.0(1.8.3.2)
  137. Content-Length: 0
  138.  
  139.  
  140. ---
  141. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Audio is at 5060
  142. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  143. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  144. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding codec 0x2 (gsm) to SDP
  145. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  146. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Reliably Transmitting (NAT) to 64.2.142.29:5060:
  147. INVITE sip:5098556587@outbound.vitelity.net SIP/2.0
  148. Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK47af3aca;rport
  149. Max-Forwards: 70
  150. From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
  151. To: <sip:5098556587@outbound.vitelity.net>
  152. Contact: <sip:njfragent@67.158.158.25:5060>
  153. Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
  154. CSeq: 103 INVITE
  155. User-Agent: FPBX-2.9.0(1.8.3.2)
  156. Proxy-Authorization: Digest username="njfragent", realm="asterisk", algorithm=MD5, uri="sip:5098556587@outbound.vitelity.net", nonce="50fb97fb", response="dd061e97a3dfd2010b5c32d0f4d078b3"
  157. Date: Wed, 14 Sep 2011 17:21:16 GMT
  158. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  159. Supported: replaces, timer
  160. Remote-Party-ID: "Hvac corp" <sip:5097682249@67.158.158.25>;party=calling;privacy=off;screen=no
  161. Content-Type: application/sdp
  162. Content-Length: 287
  163.  
  164. v=0
  165. o=root 1760196305 1760196306 IN IP4 67.158.158.25
  166. s=Asterisk PBX 1.8.3.2
  167. c=IN IP4 67.158.158.25
  168. t=0 0
  169. m=audio 11272 RTP/AVP 0 8 3 101
  170. a=rtpmap:0 PCMU/8000
  171. a=rtpmap:8 PCMA/8000
  172. a=rtpmap:3 GSM/8000
  173. a=rtpmap:101 telephone-event/8000
  174. a=fmtp:101 0-16
  175. a=ptime:20
  176. a=sendrecv
  177.  
  178. ---
  179. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  180. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  181. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  182. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  183. [2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
  184. [2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
  185. <--- SIP read from UDP:64.2.142.29:5060 --->
  186. SIP/2.0 100 Trying
  187. Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK47af3aca;received=67.158.158.25;rport=37041
  188. From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
  189. To: <sip:5098556587@outbound.vitelity.net>
  190. Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
  191. CSeq: 103 INVITE
  192. User-Agent: Asterisk PBX
  193. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  194. Supported: replaces
  195. Contact: <sip:5098556587@64.2.142.29>
  196. Content-Length: 0
  197. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c:
  198. <--- SIP read from UDP:64.2.142.29:5060 --->
  199. SIP/2.0 183 Session Progress
  200. Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK47af3aca;received=67.158.158.25;rport=37041
  201. From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
  202. To: <sip:5098556587@outbound.vitelity.net>;tag=as2fff8ffc
  203. Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
  204. CSeq: 103 INVITE
  205. User-Agent: Asterisk PBX
  206. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  207. Supported: replaces
  208. Contact: <sip:5098556587@64.2.142.29>
  209. Content-Type: application/sdp
  210. Content-Length: 283
  211.  
  212. v=0
  213. o=root 3697 3697 IN IP4 64.2.142.29
  214. s=session
  215. c=IN IP4 64.2.142.29
  216. t=0 0
  217. m=audio 12786 RTP/AVP 0 8 3 101
  218. a=rtpmap:0 PCMU/8000
  219. a=rtpmap:8 PCMA/8000
  220. a=rtpmap:3 GSM/8000
  221. a=rtpmap:101 telephone-event/8000
  222. a=fmtp:101 0-16
  223. a=silenceSupp:off - - - -
  224. a=ptime:20
  225. a=sendrecv
  226. <------------->
  227. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: --- (12 headers 14 lines) ---
  228. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 0
  229. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 8
  230. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 3
  231. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 101
  232. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format PCMU for ID 0
  233. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format PCMA for ID 8
  234. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format GSM for ID 3
  235. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format telephone-event for ID 101
  236. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  237. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  238. [2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Peer audio RTP is at port 64.2.142.29:12786
  239. [2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c: Audio is at 5060
  240. [2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  241. [2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  242. [2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c:
  243. <--- Transmitting (no NAT) to 192.168.1.158:5060 --->
  244. SIP/2.0 183 Session Progress
  245. Via: SIP/2.0/UDP 192.168.1.158;branch=z9hG4bK9f763194460023C5;received=192.168.1.158
  246. From: "204" <sip:204@192.168.1.12>;tag=F94F8C7A-CB1BBD3B
  247. To: <sip:5098556587@192.168.1.12;user=phone>;tag=as58e86aa0
  248. Call-ID: 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
  249. CSeq: 1 INVITE
  250. Server: FPBX-2.9.0(1.8.3.2)
  251. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  252. Supported: replaces, timer
  253. Contact: <sip:5098556587@192.168.1.12:5060>
  254. Content-Type: application/sdp
  255. Content-Length: 204
  256.  
  257. v=0
  258. o=root 1113813564 1113813564 IN IP4 192.168.1.12
  259. s=Asterisk PBX 1.8.3.2
  260. c=IN IP4 192.168.1.12
  261. t=0 0
  262. m=audio 19212 RTP/AVP 0 8
  263. a=rtpmap:0 PCMU/8000
  264. a=rtpmap:8 PCMA/8000
  265. a=ptime:20
  266. a=sendrecv
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