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  1. [Feb 9 13:42:31] VERBOSE[1642] logger.c: [Feb 9 13:42:31] Asterisk Event Logger restarted
  2. [Feb 9 13:42:31] VERBOSE[1642] logger.c: [Feb 9 13:42:31] Asterisk Queue Logger restarted
  3. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  4. <--- SIP read from 192.168.10.252:5060 --->
  5. INVITE sip:[email protected] SIP/2.0
  6. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport
  7. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  8. To: <sip:[email protected]>
  9. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  10. CSeq: 1 INVITE
  11. Max-Forwards: 70
  12. Contact: <sip:[email protected]:5060>;reg-id=1
  13. X-Serialnumber: 0004134110AC
  14. P-Key-Flags: resolution="31x13", keys="4"
  15. User-Agent: snom870/8.3.6
  16. Accept: application/sdp
  17. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
  18. Allow-Events: talk, hold, refer, call-info
  19. Supported: timer, 100rel, replaces, from-change
  20. Session-Expires: 3600;refresher=uas
  21. Min-SE: 90
  22. Content-Type: application/sdp
  23. Content-Length: 374
  24.  
  25. v=0
  26. o=root 1564119033 1564119033 IN IP4 192.168.10.252
  27. s=call
  28. c=IN IP4 192.168.10.252
  29. t=0 0
  30. m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101
  31. a=rtpmap:0 pcmu/8000
  32. a=rtpmap:8 pcma/8000
  33. a=rtpmap:9 g722/8000
  34. a=rtpmap:99 g726-32/8000
  35. a=rtpmap:3 gsm/8000
  36. a=rtpmap:18 g729/8000
  37. a=rtpmap:4 g723/8000
  38. a=rtpmap:101 telephone-event/8000
  39. a=fmtp:101 0-16
  40. a=ptime:20
  41. a=sendrecv
  42.  
  43. <------------->
  44. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:[email protected] SIP/2.0 (38)
  45. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport (70)
  46. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=ac9v354jjd (54)
  47. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]> (29)
  48. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
  49. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 INVITE (14)
  50. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  51. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  52. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: X-Serialnumber: 0004134110AC (28)
  53. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
  54. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom870/8.3.6 (25)
  55. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
  56. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE (96)
  57. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
  58. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
  59. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
  60. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
  61. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 17: Content-Type: application/sdp (29)
  62. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 18: Content-Length: 374 (19)
  63. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 19: (0)
  64. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: v=0 (3)
  65. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: o=root 1564119033 1564119033 IN IP4 192.168.10.252 (50)
  66. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: s=call (6)
  67. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.252 (23)
  68. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
  69. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101 (41)
  70. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
  71. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
  72. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
  73. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
  74. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
  75. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
  76. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
  77. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
  78. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
  79. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
  80. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
  81. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (19 headers 17 lines) ---
  82. [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 192.168.10.0
  83. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
  84. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Allocating new SIP dialog for f1882b3c0d38-xtxggv3ikaiv - INVITE (With RTP)
  85. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  86. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change"
  87. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -timer-
  88. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Matched SIP option: timer
  89. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -100rel-
  90. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Matched SIP option: 100rel
  91. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -replaces-
  92. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Matched SIP option: replaces
  93. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -from-change-
  94. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found no match for SIP option: from-change (Please file bug report!)
  95. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Sending to 192.168.10.252 : 5060 (NAT)
  96. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Using INVITE request as basis request - f1882b3c0d38-xtxggv3ikaiv
  97. [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
  98. [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
  99. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
  100. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  101. <--- Reliably Transmitting (NAT) to 192.168.10.252:5060 --->
  102. SIP/2.0 407 Proxy Authentication Required
  103. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;received=192.168.10.252;rport=5060
  104. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  105. To: <sip:[email protected]>;tag=as2d7f3984
  106. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  107. CSeq: 1 INVITE
  108. User-Agent: Asterisk PBX
  109. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  110. Supported: replaces
  111. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="629537b8"
  112. Content-Length: 0
  113.  
  114.  
  115. <------------>
  116. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  117. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Scheduling destruction of SIP dialog 'f1882b3c0d38-xtxggv3ikaiv' in 32000 ms (Method: INVITE)
  118. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found user '10013'
  119. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  120. <--- SIP read from 192.168.10.252:5060 --->
  121. ACK sip:[email protected] SIP/2.0
  122. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport
  123. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  124. To: <sip:[email protected]>;tag=as2d7f3984
  125. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  126. CSeq: 1 ACK
  127. Max-Forwards: 70
  128. Contact: <sip:[email protected]:5060>;reg-id=1
  129. Content-Length: 0
  130.  
  131.  
  132. <------------->
  133. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: ACK sip:[email protected] SIP/2.0 (35)
  134. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport (70)
  135. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=ac9v354jjd (54)
  136. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=as2d7f3984 (44)
  137. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
  138. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 ACK (11)
  139. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  140. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  141. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
  142. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: (0)
  143. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (9 headers 0 lines) ---
  144. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  145. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #230
  146. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Stopping retransmission on 'f1882b3c0d38-xtxggv3ikaiv' of Response 1: Match Found
  147. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  148. <--- SIP read from 192.168.10.252:5060 --->
  149. INVITE sip:[email protected] SIP/2.0
  150. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;rport
  151. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  152. To: <sip:[email protected]>
  153. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  154. CSeq: 2 INVITE
  155. Max-Forwards: 70
  156. Contact: <sip:[email protected]:5060>;reg-id=1
  157. X-Serialnumber: 0004134110AC
  158. P-Key-Flags: resolution="31x13", keys="4"
  159. User-Agent: snom870/8.3.6
  160. Accept: application/sdp
  161. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
  162. Allow-Events: talk, hold, refer, call-info
  163. Supported: timer, 100rel, replaces, from-change
  164. Session-Expires: 3600;refresher=uas
  165. Min-SE: 90
  166. Proxy-Authorization: Digest username="10013",realm="asterisk",nonce="629537b8",uri="sip:[email protected]",response="0674eb6676f2bb09c78276ad58a42c88",algorithm=MD5
  167. Content-Type: application/sdp
  168. Content-Length: 374
  169.  
  170. v=0
  171. o=root 1564119033 1564119033 IN IP4 192.168.10.252
  172. s=call
  173. c=IN IP4 192.168.10.252
  174. t=0 0
  175. m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101
  176. a=rtpmap:0 pcmu/8000
  177. a=rtpmap:8 pcma/8000
  178. a=rtpmap:9 g722/8000
  179. a=rtpmap:99 g726-32/8000
  180. a=rtpmap:3 gsm/8000
  181. a=rtpmap:18 g729/8000
  182. a=rtpmap:4 g723/8000
  183. a=rtpmap:101 telephone-event/8000
  184. a=fmtp:101 0-16
  185. a=ptime:20
  186. a=sendrecv
  187.  
  188. <------------->
  189. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:[email protected] SIP/2.0 (38)
  190. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;rport (70)
  191. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=ac9v354jjd (54)
  192. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]> (29)
  193. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
  194. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 INVITE (14)
  195. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  196. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  197. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: X-Serialnumber: 0004134110AC (28)
  198. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
  199. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom870/8.3.6 (25)
  200. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
  201. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE (96)
  202. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
  203. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
  204. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
  205. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
  206. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 17: Proxy-Authorization: Digest username="10013",realm="asterisk",nonce="629537b8",uri="sip:[email protected]",response="0674eb6676f2bb09c78276ad58a42c88",algorithm=MD5 (166)
  207. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 18: Content-Type: application/sdp (29)
  208. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 19: Content-Length: 374 (19)
  209. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 20: (0)
  210. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: v=0 (3)
  211. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: o=root 1564119033 1564119033 IN IP4 192.168.10.252 (50)
  212. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: s=call (6)
  213. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.252 (23)
  214. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
  215. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101 (41)
  216. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
  217. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
  218. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
  219. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
  220. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
  221. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
  222. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
  223. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
  224. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
  225. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
  226. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
  227. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (20 headers 17 lines) ---
  228. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  229. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Sending to 192.168.10.252 : 5060 (NAT)
  230. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Using INVITE request as basis request - f1882b3c0d38-xtxggv3ikaiv
  231. [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
  232. [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
  233. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
  234. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found user '10013'
  235. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 0
  236. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 8
  237. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 9
  238. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 99
  239. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 3
  240. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 18
  241. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 4
  242. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 101
  243. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Peer audio RTP is at port 192.168.10.252:54030
  244. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format pcmu for ID 0
  245. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format pcma for ID 8
  246. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g722 for ID 9
  247. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g726-32 for ID 99
  248. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format gsm for ID 3
  249. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g729 for ID 18
  250. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g723 for ID 4
  251. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format telephone-event for ID 101
  252. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Got unsupported a:fmtp in SDP offer
  253. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: T38 state changed to 0 on channel <none>
  254. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x8 (alaw)
  255. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  256. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Peer audio RTP is at port 192.168.10.252:54030
  257. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: We're settling with these formats: 0x8 (alaw)
  258. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Checking SIP call limits for device 10013
  259. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Updating call counter for incoming call
  260. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Call from peer '10013' is 1 out of 50
  261. [Feb 9 13:42:36] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
  262. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  263. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Looking for 10012 in from-internal (domain 192.168.10.10)
  264. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  265. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 2 (In use)
  266. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  267. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  268. [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  269. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** Our native formats are 0x8 (alaw)
  270. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** Joint capabilities are 0x8 (alaw)
  271. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** Our capabilities are 0x8 (alaw)
  272. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  273. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
  274. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  275. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: This channel will not be able to handle video.
  276. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44996af4-i4ocu4zvtwre
  277. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5060>;reg-id=1
  278. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  279. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] list_route: hop: <sip:[email protected]:5060>
  280. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: SIP/10013-0993e6e0: New call is still down.... Trying...
  281. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: set destination to 192.168.10.253, port 5060
  282. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  283. <--- Transmitting (NAT) to 192.168.10.252:5060 --->
  284. SIP/2.0 100 Trying
  285. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;received=192.168.10.252;rport=5060
  286. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  287. To: <sip:[email protected]>
  288. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  289. CSeq: 2 INVITE
  290. User-Agent: Asterisk PBX
  291. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  292. Supported: replaces
  293. Contact: <sip:[email protected]>
  294. Content-Length: 0
  295.  
  296.  
  297. <------------>
  298. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.253:5060:
  299. NOTIFY sip:[email protected]:5060 SIP/2.0
  300. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2fb5c7af;rport
  301. From: <sip:[email protected]>;tag=as6d58a6fc
  302. To: <sip:[email protected]>;tag=5e5cbuyyd1
  303. Contact: <sip:[email protected]>
  304. Call-ID: 3c2d44996af4-i4ocu4zvtwre
  305. CSeq: 113 NOTIFY
  306. User-Agent: Asterisk PBX
  307. Max-Forwards: 70
  308. Event: dialog
  309. Content-Type: application/dialog-info+xml
  310. Subscription-State: active
  311. Content-Length: 208
  312.  
  313. <?xml version="1.0"?>
  314. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:[email protected]">
  315. <dialog id="10013">
  316. <state>confirmed</state>
  317. </dialog>
  318. </dialog-info>
  319.  
  320. ---
  321. [Feb 9 13:42:36] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
  322. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  323. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Extension Changed 10013[ext-local] new state InUse for Notify User 10012
  324. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  325. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  326. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb7594-mfl0ssslkb6l
  327. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  328. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: set destination to 192.168.10.251, port 5060
  329. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.251:5060:
  330. NOTIFY sip:[email protected]:5060 SIP/2.0
  331. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2021a1b8;rport
  332. From: <sip:[email protected]>;tag=as647499f8
  333. To: <sip:[email protected]>;tag=wesbz3i941
  334. Contact: <sip:[email protected]>
  335. Call-ID: 3c27bbfb7594-mfl0ssslkb6l
  336. CSeq: 113 NOTIFY
  337. User-Agent: Asterisk PBX
  338. Max-Forwards: 70
  339. Event: dialog
  340. Content-Type: application/dialog-info+xml
  341. Subscription-State: active
  342. Content-Length: 208
  343.  
  344. <?xml version="1.0"?>
  345. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:[email protected]">
  346. <dialog id="10013">
  347. <state>confirmed</state>
  348. </dialog>
  349. </dialog-info>
  350.  
  351. ---
  352. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
  353. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  354. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [10012@from-internal:1] Macro("SIP/10013-0993e6e0", "exten-vm|novm|10012") in new stack
  355. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Extension Changed 10013[ext-local] new state InUse for Notify User 10014
  356. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  357. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
  358. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  359. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 2 (In use)
  360. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:1] Macro("SIP/10013-0993e6e0", "user-callerid") in new stack
  361. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  362. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  363. [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  364. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
  365. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
  366. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
  367. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  368. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:1] Set("SIP/10013-0993e6e0", "AMPUSER=10013") in new stack
  369. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  370. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
  371. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
  372. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:2] GotoIf("SIP/10013-0993e6e0", "0?report") in new stack
  373. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Not taking any branch
  374. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
  375. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
  376. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
  377. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'ExecIf'
  378. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:3] ExecIf("SIP/10013-0993e6e0", "1|Set|REALCALLERIDNUM=10013") in new stack
  379. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: ExecIf
  380. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
  381. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
  382. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
  383. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  384. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:4] Set("SIP/10013-0993e6e0", "AMPUSER=10013") in new stack
  385. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  386. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is 'Test3'
  387. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  388. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:5] Set("SIP/10013-0993e6e0", "AMPUSERCIDNAME=Test3") in new stack
  389. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  390. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
  391. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
  392. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:6] GotoIf("SIP/10013-0993e6e0", "0?report") in new stack
  393. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Not taking any branch
  394. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
  395. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '1'
  396. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
  397. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
  398. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  399. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:7] Set("SIP/10013-0993e6e0", "AMPUSERCID=10013") in new stack
  400. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  401. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  402. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:8] Set("SIP/10013-0993e6e0", "CALLERID(all)="Test3" <10013>") in new stack
  403. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  404. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
  405. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
  406. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:9] GotoIf("SIP/10013-0993e6e0", "0?continue") in new stack
  407. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Not taking any branch
  408. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
  409. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
  410. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '-1'
  411. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '64'
  412. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  413. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:10] Set("SIP/10013-0993e6e0", "__TTL=64") in new stack
  414. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  415. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
  416. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
  417. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:11] GotoIf("SIP/10013-0993e6e0", "1?continue") in new stack
  418. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Goto (macro-user-callerid,s,18)
  419. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
  420. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '"Test3" <10013>'
  421. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'NoOp'
  422. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:18] NoOp("SIP/10013-0993e6e0", "Using CallerID "Test3" <10013>") in new stack
  423. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Noop
  424. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Macro
  425. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  426. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:2] Set("SIP/10013-0993e6e0", "RingGroupMethod=none") in new stack
  427. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  428. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  429. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:3] Set("SIP/10013-0993e6e0", "VMBOX=novm") in new stack
  430. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  431. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  432. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:4] Set("SIP/10013-0993e6e0", "EXTTOCALL=10012") in new stack
  433. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  434. [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFU'
  435. [Feb 9 13:42:36] DEBUG[1643] func_db.c: DB: CFU/10012 not found in database.
  436. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is ''
  437. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  438. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:5] Set("SIP/10013-0993e6e0", "CFUEXT=") in new stack
  439. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  440. [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFB'
  441. [Feb 9 13:42:36] DEBUG[1643] func_db.c: DB: CFB/10012 not found in database.
  442. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is ''
  443. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  444. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:6] Set("SIP/10013-0993e6e0", "CFBEXT=") in new stack
  445. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  446. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
  447. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
  448. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
  449. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '""'
  450. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
  451. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:7] Set("SIP/10013-0993e6e0", "RT=""") in new stack
  452. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
  453. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
  454. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:8] Macro("SIP/10013-0993e6e0", "record-enable|10012|IN") in new stack
  455. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
  456. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
  457. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-record-enable:1] GotoIf("SIP/10013-0993e6e0", "1?check") in new stack
  458. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Goto (macro-record-enable,s,4)
  459. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
  460. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '20100209-134236'
  461. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'AGI'
  462. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-record-enable:4] AGI("SIP/10013-0993e6e0", "recordingcheck|20100209-134236|1265719356.18") in new stack
  463. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  464. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  465. <--- SIP read from 192.168.10.251:5060 --->
  466. SIP/2.0 200 Ok
  467. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2021a1b8;rport=5060
  468. From: <sip:[email protected]>;tag=as647499f8
  469. To: <sip:[email protected]>;tag=wesbz3i941
  470. Call-ID: 3c27bbfb7594-mfl0ssslkb6l
  471. CSeq: 113 NOTIFY
  472. Content-Length: 0
  473.  
  474.  
  475. <------------->
  476. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
  477. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2021a1b8;rport=5060 (69)
  478. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as647499f8 (46)
  479. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=wesbz3i941 (44)
  480. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb7594-mfl0ssslkb6l (34)
  481. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
  482. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
  483. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: (0)
  484. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (7 headers 0 lines) ---
  485. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Acked pending invite 113
  486. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #233
  487. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb7594-mfl0ssslkb6l' of Request 113: Match Found
  488. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] SIP Response message for INCOMING dialog NOTIFY arrived
  489. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  490. <--- SIP read from 192.168.10.253:5060 --->
  491. SIP/2.0 200 Ok
  492. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2fb5c7af;rport=5060
  493. From: <sip:[email protected]>;tag=as6d58a6fc
  494. To: <sip:[email protected]>;tag=5e5cbuyyd1
  495. Call-ID: 3c2d44996af4-i4ocu4zvtwre
  496. CSeq: 113 NOTIFY
  497. Content-Length: 0
  498.  
  499.  
  500. <------------->
  501. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
  502. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2fb5c7af;rport=5060 (69)
  503. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as6d58a6fc (46)
  504. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=5e5cbuyyd1 (44)
  505. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c2d44996af4-i4ocu4zvtwre (34)
  506. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
  507. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
  508. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: (0)
  509. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (7 headers 0 lines) ---
  510. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Acked pending invite 113
  511. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #232
  512. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c2d44996af4-i4ocu4zvtwre' of Request 113: Match Found
  513. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] SIP Response message for INCOMING dialog NOTIFY arrived
  514. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] recordingcheck|20100209-134236|1265719356.18: Inbound recording not enabled
  515. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- AGI Script recordingcheck completed, returning 0
  516. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: AGI
  517. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'MacroExit'
  518. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-record-enable:5] MacroExit("SIP/10013-0993e6e0", "") in new stack
  519. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Macro
  520. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
  521. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:9] Macro("SIP/10013-0993e6e0", "dial||tr|10012") in new stack
  522. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
  523. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
  524. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-dial:1] GotoIf("SIP/10013-0993e6e0", "1?dial") in new stack
  525. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Goto (macro-dial,s,3)
  526. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
  527. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'AGI'
  528. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-dial:3] AGI("SIP/10013-0993e6e0", "dialparties.agi") in new stack
  529. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  530. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: Starting New Dialparties.agi
  531. [Feb 9 13:42:36] DEBUG[1646] manager.c: Manager received command 'login'
  532. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Parsing '/etc/asterisk/manager.conf': [Feb 9 13:42:36] DEBUG[1646] config.c: Parsing /etc/asterisk/manager.conf
  533. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] Found
  534. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Parsing '/etc/asterisk/manager_additional.conf': [Feb 9 13:42:36] DEBUG[1646] config.c: Parsing /etc/asterisk/manager_additional.conf
  535. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] Found
  536. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Parsing '/etc/asterisk/manager_custom.conf': [Feb 9 13:42:36] DEBUG[1646] config.c: Parsing /etc/asterisk/manager_custom.conf
  537. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] Found
  538. [Feb 9 13:42:36] DEBUG[1646] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer
  539. [Feb 9 13:42:36] DEBUG[1646] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer
  540. [Feb 9 13:42:36] DEBUG[1646] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0
  541. [Feb 9 13:42:36] DEBUG[1646] acl.c: ##### Testing 127.0.0.1 with 127.0.0.0
  542. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Manager 'admin' logged on from 127.0.0.1
  543. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: Caller ID name is 'Test3' number is '10013'
  544. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: USE_CONFIRMATION: 'FALSE'
  545. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: RINGGROUP_INDEX: ''
  546. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: Methodology of ring is 'none'
  547. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Added extension 10012 to extension map
  548. [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CF'
  549. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Extension 10012 cf is disabled
  550. [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'DND'
  551. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Extension 10012 do not disturb is disabled
  552. [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFB'
  553. [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFU'
  554. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] > dialparties.agi: extnum 10012 has: cw: 1; hascfb: 0 [] hascfu: 0 []
  555. [Feb 9 13:42:36] DEBUG[1646] manager.c: Manager received command 'ExtensionState'
  556. [Feb 9 13:42:36] DEBUG[1646] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  557. [Feb 9 13:42:36] DEBUG[1646] chan_sip.c: Checking device state for peer 10012
  558. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: ExtensionState: 0
  559. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: dbset CALLTRACE/10012 to 10013
  560. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Filtered ARG3: 10012
  561. [Feb 9 13:42:36] DEBUG[1646] manager.c: Manager received command 'Logoff'
  562. [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Manager 'admin' logged off from 127.0.0.1
  563. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- AGI Script dialparties.agi completed, returning 0
  564. [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: AGI
  565. [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Dial'
  566. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-dial:7] Dial("SIP/10013-0993e6e0", "SIP/10012||tr") in new stack
  567. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw)
  568. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
  569. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Setting NAT on RTP to On
  570. [Feb 9 13:42:36] DEBUG[1643] acl.c: ##### Testing 192.168.10.253 with 192.168.10.0
  571. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Our native formats are 0x8 (alaw)
  572. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Joint capabilities are 0x0 (nothing)
  573. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Our capabilities are 0x8 (alaw)
  574. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
  575. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw)
  576. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: This channel will not be able to handle video.
  577. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALEDTIME.
  578. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ANSWEREDTIME.
  579. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALEDPEERNAME.
  580. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALEDPEERNUMBER.
  581. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALSTATUS.
  582. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_DEPTH.
  583. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AGISTATUS.
  584. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ds.
  585. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable FILTERED_DIAL.
  586. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALSTATUS_CW.
  587. [Feb 9 13:42:36] DEBUG[1643] channel.c: Copying hard-transferable variable KEEPCID.
  588. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ARG3.
  589. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ARG2.
  590. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ARG1.
  591. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_PRIORITY.
  592. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_CONTEXT.
  593. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_EXTEN.
  594. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable RT.
  595. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable CFBEXT.
  596. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable CFUEXT.
  597. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable EXTTOCALL.
  598. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable VMBOX.
  599. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable RingGroupMethod.
  600. [Feb 9 13:42:36] DEBUG[1643] channel.c: Copying hard-transferable variable TTL.
  601. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AMPUSERCID.
  602. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DB_RESULT.
  603. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AMPUSERCIDNAME.
  604. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AMPUSER.
  605. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable REALCALLERIDNUM.
  606. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPCALLID.
  607. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPUSERAGENT.
  608. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPDOMAIN.
  609. [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPURI.
  610. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Outgoing Call for 10012
  611. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Updating call counter for outgoing call
  612. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Call to peer '10012' is 1 out of 50
  613. [Feb 9 13:42:36] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
  614. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Our T38 capability (0), joint T38 capability (0)
  615. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  616. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  617. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: False
  618. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 6 (Ringing)
  619. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: ** Our prefcodec: 0x8 (alaw)
  620. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  621. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Audio is at 192.168.10.10 port 18198
  622. [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
  623. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Adding codec 0x8 (alaw) to SDP
  624. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  625. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Adding non-codec 0x1 (telephone-event) to SDP
  626. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: -- Done with adding codecs to SDP
  627. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
  628. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 0: INVITE sip:[email protected]:5060 SIP/2.0 (44)
  629. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport (64)
  630. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  631. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as7741abae (54)
  632. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  633. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 3: To: <sip:[email protected]:5060> (35)
  634. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb6fc4-t8z10gt0v8fu
  635. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 4: Contact: <sip:[email protected]> (34)
  636. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 5: Call-ID: [email protected] (55)
  637. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  638. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 6: CSeq: 102 INVITE (16)
  639. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24)
  640. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: set destination to 192.168.10.251, port 5060
  641. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 8: Max-Forwards: 70 (16)
  642. [Feb 9 13:42:36] DEBUG[1565] channel.c: Avoiding initial deadlock for channel '0x9942958'
  643. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 9: Date: Tue, 09 Feb 2010 12:42:36 GMT (35)
  644. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
  645. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 11: Supported: replaces (19)
  646. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 12: Content-Type: application/sdp (29)
  647. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 13: Content-Length: 240 (19)
  648. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 14: (0)
  649. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: v=0 (3)
  650. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: o=root 1562 1562 IN IP4 192.168.10.10 (37)
  651. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: s=session (9)
  652. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: c=IN IP4 192.168.10.10 (22)
  653. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: t=0 0 (5)
  654. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: m=audio 18198 RTP/AVP 8 101 (27)
  655. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
  656. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
  657. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=fmtp:101 0-16 (15)
  658. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=silenceSupp:off - - - - (25)
  659. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=ptime:20 (10)
  660. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=sendrecv (10)
  661. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.253:5060:
  662. INVITE sip:[email protected]:5060 SIP/2.0
  663. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  664. From: "Test3" <sip:[email protected]>;tag=as7741abae
  665. To: <sip:[email protected]:5060>
  666. Contact: <sip:[email protected]>
  667. Call-ID: [email protected]
  668. CSeq: 102 INVITE
  669. User-Agent: Asterisk PBX
  670. Max-Forwards: 70
  671. Date: Tue, 09 Feb 2010 12:42:36 GMT
  672. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  673. Supported: replaces
  674. Content-Type: application/sdp
  675. Content-Length: 240
  676.  
  677. v=0
  678. o=root 1562 1562 IN IP4 192.168.10.10
  679. s=session
  680. c=IN IP4 192.168.10.10
  681. t=0 0
  682. m=audio 18198 RTP/AVP 8 101
  683. a=rtpmap:8 PCMA/8000
  684. a=rtpmap:101 telephone-event/8000
  685. a=fmtp:101 0-16
  686. a=silenceSupp:off - - - -
  687. a=ptime:20
  688. a=sendrecv
  689.  
  690. ---
  691. [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  692. [Feb 9 13:42:36] DEBUG[1565] channel.c: Avoiding initial deadlock for channel '0x9942958'
  693. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Called 10012
  694. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36]
  695. <--- Transmitting (NAT) to 192.168.10.252:5060 --->
  696. SIP/2.0 180 Ringing
  697. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;received=192.168.10.252;rport=5060
  698. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  699. To: <sip:[email protected]>;tag=as17f86ff1
  700. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  701. CSeq: 2 INVITE
  702. User-Agent: Asterisk PBX
  703. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  704. Supported: replaces
  705. Contact: <sip:[email protected]>
  706. Content-Length: 0
  707.  
  708.  
  709. <------------>
  710. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Sent call-pickup info to peer 10014
  711. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.251:5060:
  712. NOTIFY sip:[email protected]:5060 SIP/2.0
  713. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK31065cb0;rport
  714. From: <sip:[email protected]>;tag=as31f661fa
  715. To: <sip:[email protected]>;tag=ttbbodh6oe
  716. Contact: <sip:[email protected]>
  717. Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
  718. CSeq: 113 NOTIFY
  719. User-Agent: Asterisk PBX
  720. Max-Forwards: 70
  721. Event: dialog
  722. Content-Type: application/dialog-info+xml
  723. Subscription-State: active
  724. Content-Length: 497
  725.  
  726. <?xml version="1.0"?>
  727. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:[email protected]">
  728. <dialog id="10012" call-id="[email protected]" direction="recipient">
  729. <local><identity display="10012">10012</identity><target uri="sip:[email protected]"/></local>
  730. <remote><identity display="Test3">sip:[email protected]</identity><target uri="sip:[email protected]"/></remote>
  731. <state>early</state>
  732. </dialog>
  733. </dialog-info>
  734.  
  735. ---
  736. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  737. [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Extension Changed 10012[ext-local] new state Ringing for Notify User 10014
  738. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
  739. <--- SIP read from 192.168.10.253:5060 --->
  740. SIP/2.0 180 Ringing
  741. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
  742. From: "Test3" <sip:[email protected]>;tag=as7741abae
  743. To: <sip:[email protected]:5060>;tag=p88g5axhr5
  744. Call-ID: [email protected]
  745. CSeq: 102 INVITE
  746. Contact: <sip:[email protected]:5060>;flow-id=1
  747. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  748. Allow-Events: talk, hold, refer, call-info
  749. Content-Length: 0
  750.  
  751.  
  752. <------------->
  753. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19)
  754. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
  755. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as7741abae (54)
  756. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]:5060>;tag=p88g5axhr5 (50)
  757. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: [email protected] (55)
  758. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
  759. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;flow-id=1 (50)
  760. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
  761. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: Allow-Events: talk, hold, refer, call-info (42)
  762. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: Content-Length: 0 (17)
  763. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 10: (0)
  764. [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (10 headers 0 lines) ---
  765. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** SIP TIMER: Cancelling retransmission #234 - INVITE (got response)
  766. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 102: Found
  767. [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: SIP response 180 to standard invite
  768. [Feb 9 13:42:36] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
  769. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  770. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  771. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 6 (Ringing)
  772. [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  773. [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  774. [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
  775. [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- SIP/10012-09943518 is ringing
  776. [Feb 9 13:42:37] VERBOSE[1571] logger.c: [Feb 9 13:42:37]
  777. <--- SIP read from 192.168.10.251:5060 --->
  778. SIP/2.0 200 Ok
  779. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK31065cb0;rport=5060
  780. From: <sip:[email protected]>;tag=as31f661fa
  781. To: <sip:[email protected]>;tag=ttbbodh6oe
  782. Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
  783. CSeq: 113 NOTIFY
  784. Content-Length: 0
  785.  
  786.  
  787. <------------->
  788. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
  789. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK31065cb0;rport=5060 (69)
  790. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as31f661fa (46)
  791. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=ttbbodh6oe (44)
  792. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu (34)
  793. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
  794. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
  795. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 7: (0)
  796. [Feb 9 13:42:37] VERBOSE[1571] logger.c: [Feb 9 13:42:37] --- (7 headers 0 lines) ---
  797. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Acked pending invite 113
  798. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #236
  799. [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb6fc4-t8z10gt0v8fu' of Request 113: Match Found
  800. [Feb 9 13:42:37] VERBOSE[1571] logger.c: [Feb 9 13:42:37] SIP Response message for INCOMING dialog NOTIFY arrived
  801. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  802. <--- SIP read from 192.168.10.251:5060 --->
  803. INVITE sip:[email protected] SIP/2.0
  804. Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport
  805. From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  806. To: "Test3" <sip:[email protected]>
  807. Call-ID: 3c27bc5ad168-wohs43kd06vu
  808. CSeq: 1 INVITE
  809. Max-Forwards: 70
  810. Contact: <sip:[email protected]:5060>;reg-id=1
  811. Replaces: [email protected]
  812. P-Key-Flags: resolution="31x13", keys="4"
  813. User-Agent: snom360/7.3.30
  814. Accept: application/sdp
  815. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  816. Allow-Events: talk, hold, refer, call-info
  817. Supported: timer, 100rel, replaces, from-change
  818. Session-Expires: 3600;refresher=uas
  819. Min-SE: 90
  820. Content-Type: application/sdp
  821. Content-Length: 374
  822.  
  823. v=0
  824. o=root 1866843021 1866843021 IN IP4 192.168.10.251
  825. s=call
  826. c=IN IP4 192.168.10.251
  827. t=0 0
  828. m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101
  829. a=rtpmap:0 pcmu/8000
  830. a=rtpmap:8 pcma/8000
  831. a=rtpmap:9 g722/8000
  832. a=rtpmap:99 g726-32/8000
  833. a=rtpmap:3 gsm/8000
  834. a=rtpmap:18 g729/8000
  835. a=rtpmap:4 g723/8000
  836. a=rtpmap:101 telephone-event/8000
  837. a=fmtp:101 0-16
  838. a=ptime:20
  839. a=sendrecv
  840.  
  841. <------------->
  842. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:[email protected] SIP/2.0 (38)
  843. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport (70)
  844. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6 (54)
  845. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:[email protected]> (37)
  846. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
  847. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 INVITE (14)
  848. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  849. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  850. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Replaces: [email protected] (56)
  851. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
  852. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom360/7.3.30 (26)
  853. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
  854. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
  855. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
  856. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
  857. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
  858. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
  859. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 17: Content-Type: application/sdp (29)
  860. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 18: Content-Length: 374 (19)
  861. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 19: (0)
  862. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: v=0 (3)
  863. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: o=root 1866843021 1866843021 IN IP4 192.168.10.251 (50)
  864. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: s=call (6)
  865. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.251 (23)
  866. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
  867. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101 (41)
  868. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
  869. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
  870. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
  871. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
  872. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
  873. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
  874. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
  875. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
  876. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
  877. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
  878. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
  879. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (19 headers 17 lines) ---
  880. [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 192.168.10.0
  881. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
  882. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Allocating new SIP dialog for 3c27bc5ad168-wohs43kd06vu - INVITE (With RTP)
  883. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  884. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change"
  885. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -timer-
  886. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Matched SIP option: timer
  887. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -100rel-
  888. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Matched SIP option: 100rel
  889. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -replaces-
  890. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Matched SIP option: replaces
  891. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -from-change-
  892. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found no match for SIP option: from-change (Please file bug report!)
  893. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: INVITE part of call transfer. Replaces [[email protected]]
  894. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : [email protected] Fromtag: <no from tag> Totag: <no to tag>
  895. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found call with callid [email protected] (ourtag=as7741abae, theirtag=p88g5axhr5)
  896. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Sending to 192.168.10.251 : 5060 (NAT)
  897. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Using INVITE request as basis request - 3c27bc5ad168-wohs43kd06vu
  898. [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
  899. [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
  900. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
  901. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  902. <--- Reliably Transmitting (NAT) to 192.168.10.251:5060 --->
  903. SIP/2.0 407 Proxy Authentication Required
  904. Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;received=192.168.10.251;rport=5060
  905. From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  906. To: "Test3" <sip:[email protected]>;tag=as5a4bc795
  907. Call-ID: 3c27bc5ad168-wohs43kd06vu
  908. CSeq: 1 INVITE
  909. User-Agent: Asterisk PBX
  910. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  911. Supported: replaces
  912. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="517a6259"
  913. Content-Length: 0
  914.  
  915.  
  916. <------------>
  917. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  918. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Scheduling destruction of SIP dialog '3c27bc5ad168-wohs43kd06vu' in 32000 ms (Method: INVITE)
  919. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found user '10014'
  920. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  921. <--- SIP read from 192.168.10.251:5060 --->
  922. ACK sip:[email protected] SIP/2.0
  923. Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport
  924. From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  925. To: "Test3" <sip:[email protected]>;tag=as5a4bc795
  926. Call-ID: 3c27bc5ad168-wohs43kd06vu
  927. CSeq: 1 ACK
  928. Max-Forwards: 70
  929. Contact: <sip:[email protected]:5060>;reg-id=1
  930. Content-Length: 0
  931.  
  932.  
  933. <------------->
  934. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: ACK sip:[email protected] SIP/2.0 (35)
  935. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport (70)
  936. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6 (54)
  937. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:[email protected]>;tag=as5a4bc795 (52)
  938. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
  939. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 ACK (11)
  940. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  941. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  942. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
  943. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: (0)
  944. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (9 headers 0 lines) ---
  945. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  946. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #237
  947. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bc5ad168-wohs43kd06vu' of Response 1: Match Found
  948. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  949. <--- SIP read from 192.168.10.251:5060 --->
  950. INVITE sip:[email protected] SIP/2.0
  951. Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;rport
  952. From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  953. To: "Test3" <sip:[email protected]>
  954. Call-ID: 3c27bc5ad168-wohs43kd06vu
  955. CSeq: 2 INVITE
  956. Max-Forwards: 70
  957. Contact: <sip:[email protected]:5060>;reg-id=1
  958. Replaces: [email protected]
  959. P-Key-Flags: resolution="31x13", keys="4"
  960. User-Agent: snom360/7.3.30
  961. Accept: application/sdp
  962. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  963. Allow-Events: talk, hold, refer, call-info
  964. Supported: timer, 100rel, replaces, from-change
  965. Session-Expires: 3600;refresher=uas
  966. Min-SE: 90
  967. Proxy-Authorization: Digest username="10014",realm="asterisk",nonce="517a6259",uri="sip:[email protected]",response="7c7b135179e96dde5516c178fc21f003",algorithm=MD5
  968. Content-Type: application/sdp
  969. Content-Length: 374
  970.  
  971. v=0
  972. o=root 1866843021 1866843021 IN IP4 192.168.10.251
  973. s=call
  974. c=IN IP4 192.168.10.251
  975. t=0 0
  976. m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101
  977. a=rtpmap:0 pcmu/8000
  978. a=rtpmap:8 pcma/8000
  979. a=rtpmap:9 g722/8000
  980. a=rtpmap:99 g726-32/8000
  981. a=rtpmap:3 gsm/8000
  982. a=rtpmap:18 g729/8000
  983. a=rtpmap:4 g723/8000
  984. a=rtpmap:101 telephone-event/8000
  985. a=fmtp:101 0-16
  986. a=ptime:20
  987. a=sendrecv
  988.  
  989. <------------->
  990. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:[email protected] SIP/2.0 (38)
  991. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;rport (70)
  992. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6 (54)
  993. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:[email protected]> (37)
  994. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
  995. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 INVITE (14)
  996. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  997. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  998. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Replaces: [email protected] (56)
  999. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
  1000. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom360/7.3.30 (26)
  1001. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
  1002. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
  1003. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
  1004. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
  1005. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
  1006. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
  1007. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 17: Proxy-Authorization: Digest username="10014",realm="asterisk",nonce="517a6259",uri="sip:[email protected]",response="7c7b135179e96dde5516c178fc21f003",algorithm=MD5 (166)
  1008. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 18: Content-Type: application/sdp (29)
  1009. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 19: Content-Length: 374 (19)
  1010. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 20: (0)
  1011. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: v=0 (3)
  1012. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: o=root 1866843021 1866843021 IN IP4 192.168.10.251 (50)
  1013. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: s=call (6)
  1014. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.251 (23)
  1015. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
  1016. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101 (41)
  1017. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
  1018. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
  1019. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
  1020. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
  1021. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
  1022. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
  1023. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
  1024. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
  1025. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
  1026. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
  1027. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
  1028. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (20 headers 17 lines) ---
  1029. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
  1030. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: INVITE part of call transfer. Replaces [[email protected]]
  1031. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : [email protected] Fromtag: <no from tag> Totag: <no to tag>
  1032. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found call with callid [email protected] (ourtag=as7741abae, theirtag=p88g5axhr5)
  1033. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Sending to 192.168.10.251 : 5060 (NAT)
  1034. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Using INVITE request as basis request - 3c27bc5ad168-wohs43kd06vu
  1035. [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
  1036. [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
  1037. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
  1038. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found user '10014'
  1039. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 0
  1040. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 8
  1041. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 9
  1042. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 99
  1043. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 3
  1044. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 18
  1045. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 4
  1046. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 101
  1047. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Peer audio RTP is at port 192.168.10.251:56984
  1048. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format pcmu for ID 0
  1049. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format pcma for ID 8
  1050. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g722 for ID 9
  1051. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g726-32 for ID 99
  1052. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format gsm for ID 3
  1053. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g729 for ID 18
  1054. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g723 for ID 4
  1055. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format telephone-event for ID 101
  1056. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Got unsupported a:fmtp in SDP offer
  1057. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: T38 state changed to 0 on channel <none>
  1058. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x8 (alaw)
  1059. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  1060. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Peer audio RTP is at port 192.168.10.251:56984
  1061. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: We're settling with these formats: 0x8 (alaw)
  1062. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Checking SIP call limits for device 10014
  1063. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Updating call counter for incoming call
  1064. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Call from peer '10014' is 1 out of 50
  1065. [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
  1066. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1067. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Looking for 10012 in from-internal (domain 192.168.10.10)
  1068. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1069. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 2 (In use)
  1070. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1071. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  1072. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1073. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** Our native formats are 0x8 (alaw)
  1074. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** Joint capabilities are 0x8 (alaw)
  1075. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1076. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** Our capabilities are 0x8 (alaw)
  1077. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1078. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
  1079. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44997030-61xfuv9xjhtd
  1080. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: This channel will not be able to handle video.
  1081. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: build_route: Contact hop: <sip:[email protected]:5060>;reg-id=1
  1082. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  1083. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] list_route: hop: <sip:[email protected]:5060>
  1084. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: set destination to 192.168.10.253, port 5060
  1085. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Sending this call to the invite/replcaes handler 3c27bc5ad168-wohs43kd06vu
  1086. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Attended transfer attempted to replace call with no bridge (maybe ringing). Channel SIP/10012-09943518!
  1087. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP transfer: Invite Replace incoming channel should replace and hang up channel SIP/10012-09943518 (one call leg)
  1088. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Reliably Transmitting (NAT) to 192.168.10.253:5060:
  1089. NOTIFY sip:[email protected]:5060 SIP/2.0
  1090. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport
  1091. From: <sip:[email protected]>;tag=as68be196f
  1092. To: <sip:[email protected]>;tag=nw4c6bdjzk
  1093. Contact: <sip:[email protected]>
  1094. Call-ID: 3c2d44997030-61xfuv9xjhtd
  1095. CSeq: 113 NOTIFY
  1096. User-Agent: Asterisk PBX
  1097. Max-Forwards: 70
  1098. Event: dialog
  1099. Content-Type: application/dialog-info+xml
  1100. Subscription-State: active
  1101. Content-Length: 208
  1102.  
  1103. <?xml version="1.0"?>
  1104. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:[email protected]">
  1105. <dialog id="10014">
  1106. <state>confirmed</state>
  1107. </dialog>
  1108. </dialog-info>
  1109.  
  1110. ---
  1111. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1112. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1113. <--- Transmitting (NAT) to 192.168.10.251:5060 --->
  1114. SIP/2.0 100 Trying
  1115. Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;received=192.168.10.251;rport=5060
  1116. From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  1117. To: "Test3" <sip:[email protected]>
  1118. Call-ID: 3c27bc5ad168-wohs43kd06vu
  1119. CSeq: 2 INVITE
  1120. User-Agent: Asterisk PBX
  1121. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1122. Supported: replaces
  1123. Contact: <sip:[email protected]>
  1124. Content-Length: 0
  1125.  
  1126.  
  1127. <------------>
  1128. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Extension Changed 10014[ext-local] new state InUse for Notify User 10012
  1129. [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
  1130. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1131. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1132. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting framing from config on incoming call
  1133. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 2 (In use)
  1134. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True
  1135. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1136. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
  1137. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1138. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Audio is at 192.168.10.10 port 19568
  1139. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  1140. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Adding codec 0x8 (alaw) to SDP
  1141. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Adding non-codec 0x1 (telephone-event) to SDP
  1142. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- Done with adding codecs to SDP
  1143. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
  1144. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1145. <--- Reliably Transmitting (NAT) to 192.168.10.251:5060 --->
  1146. SIP/2.0 200 OK
  1147. Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;received=192.168.10.251;rport=5060
  1148. From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  1149. To: "Test3" <sip:[email protected]>;tag=as673f2fe0
  1150. Call-ID: 3c27bc5ad168-wohs43kd06vu
  1151. CSeq: 2 INVITE
  1152. User-Agent: Asterisk PBX
  1153. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1154. Supported: replaces
  1155. Contact: <sip:[email protected]>
  1156. Content-Type: application/sdp
  1157. Content-Length: 240
  1158.  
  1159. v=0
  1160. o=root 1562 1562 IN IP4 192.168.10.10
  1161. s=session
  1162. c=IN IP4 192.168.10.10
  1163. t=0 0
  1164. m=audio 19568 RTP/AVP 8 101
  1165. a=rtpmap:8 PCMA/8000
  1166. a=rtpmap:101 telephone-event/8000
  1167. a=fmtp:101 0-16
  1168. a=silenceSupp:off - - - -
  1169. a=ptime:20
  1170. a=sendrecv
  1171.  
  1172. <------------>
  1173. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1174. [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
  1175. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/Replaces: preparing to masquerade SIP/10014-0994c8c8 into SIP/10012-09943518
  1176. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1177. [Feb 9 13:42:38] DEBUG[1571] channel.c: Planning to masquerade channel SIP/10014-0994c8c8 into the structure of SIP/10012-09943518
  1178. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1179. [Feb 9 13:42:38] DEBUG[1571] channel.c: Done planning to masquerade channel SIP/10014-0994c8c8 into the structure of SIP/10012-09943518
  1180. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 2 (In use)
  1181. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/Replaces: Going to masquerade SIP/10014-0994c8c8 into SIP/10012-09943518
  1182. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1183. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  1184. [Feb 9 13:42:38] DEBUG[1571] channel.c: Actually Masquerading SIP/10014-0994c8c8(6) into the structure of SIP/10012-09943518(5)
  1185. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1186. [Feb 9 13:42:38] DEBUG[1571] channel.c: Got clone lock for masquerade on 'SIP/10014-0994c8c8' at 0x97e1868
  1187. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP Fixup: New owner for dialogue [email protected]: SIP/10014-0994c8c8<MASQ> (Old parent: SIP/10014-0994c8c8)
  1188. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Hangup call SIP/10014-0994c8c8<MASQ>, SIP callid [email protected])
  1189. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: update_call_counter(10012) - decrement call limit counter on hangup
  1190. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Updating call counter for outgoing call
  1191. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Call to peer '10012' removed from call limit 50
  1192. [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
  1193. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1194. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Hanging up channel in state Ringing (not UP)
  1195. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1196. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
  1197. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  1198. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1199. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1200. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1201. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 102: Found
  1202. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Reliably Transmitting (NAT) to 192.168.10.253:5060:
  1203. CANCEL sip:[email protected]:5060 SIP/2.0
  1204. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1205. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1206. To: <sip:[email protected]:5060>
  1207. Call-ID: [email protected]
  1208. CSeq: 102 CANCEL
  1209. User-Agent: Asterisk PBX
  1210. Max-Forwards: 70
  1211. Content-Length: 0
  1212.  
  1213.  
  1214. ---
  1215. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1216. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1217. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1218. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  1219. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb6fc4-t8z10gt0v8fu
  1220. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  1221. [Feb 9 13:42:38] DEBUG[1571] channel.c: Putting channel SIP/10014-0994c8c8 in 8/8 formats
  1222. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: set destination to 192.168.10.251, port 5060
  1223. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP Fixup: New owner for dialogue 3c27bc5ad168-wohs43kd06vu: SIP/10014-0994c8c8 (Old parent: SIP/10012-09943518<ZOMBIE>)
  1224. [Feb 9 13:42:38] DEBUG[1571] channel.c: Released clone lock on 'SIP/10012-09943518<ZOMBIE>'
  1225. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Reliably Transmitting (NAT) to 192.168.10.251:5060:
  1226. NOTIFY sip:[email protected]:5060 SIP/2.0
  1227. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK05ef9d45;rport
  1228. From: <sip:[email protected]>;tag=as31f661fa
  1229. To: <sip:[email protected]>;tag=ttbbodh6oe
  1230. Contact: <sip:[email protected]>
  1231. Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
  1232. CSeq: 114 NOTIFY
  1233. User-Agent: Asterisk PBX
  1234. Max-Forwards: 70
  1235. Event: dialog
  1236. Content-Type: application/dialog-info+xml
  1237. Subscription-State: active
  1238. Content-Length: 209
  1239.  
  1240. <?xml version="1.0"?>
  1241. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:[email protected]">
  1242. <dialog id="10012">
  1243. <state>terminated</state>
  1244. </dialog>
  1245. </dialog-info>
  1246.  
  1247. ---
  1248. [Feb 9 13:42:38] DEBUG[1571] channel.c: Done Masquerading SIP/10014-0994c8c8 (6)
  1249. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1250. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/Replace: Could successfully read frame from RING channel!
  1251. [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
  1252. [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Extension Changed 10012[ext-local] new state Idle for Notify User 10014
  1253. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: After transfer:----------------------------
  1254. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1255. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- C: SIP/10012-09943518<ZOMBIE> State Down
  1256. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1257. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- replacecall: SIP/10014-0994c8c8 State Up
  1258. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
  1259. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- P->owner: SIP/10014-0994c8c8 State Up
  1260. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1261. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- No call bridged to C->owner
  1262. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1263. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1264. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: End After transfer:----------------------------
  1265. [Feb 9 13:42:38] DEBUG[1571] channel.c: Hanging up zombie 'SIP/10012-09943518<ZOMBIE>'
  1266. [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
  1267. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1268. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1269. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
  1270. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1271. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1272. [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] -- SIP/10014-0994c8c8 answered SIP/10013-0993e6e0
  1273. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1274. [Feb 9 13:42:38] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
  1275. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1276. [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: SIP answering channel: SIP/10013-0993e6e0
  1277. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1278. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 2 (In use)
  1279. [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: Setting framing from config on incoming call
  1280. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1281. [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True
  1282. [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
  1283. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1284. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  1285. [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] Audio is at 192.168.10.10 port 10000
  1286. [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] Adding codec 0x8 (alaw) to SDP
  1287. [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] Adding non-codec 0x1 (telephone-event) to SDP
  1288. [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: -- Done with adding codecs to SDP
  1289. [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
  1290. [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38]
  1291. <--- Reliably Transmitting (NAT) to 192.168.10.252:5060 --->
  1292. SIP/2.0 200 OK
  1293. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;received=192.168.10.252;rport=5060
  1294. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  1295. To: <sip:[email protected]>;tag=as17f86ff1
  1296. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  1297. CSeq: 2 INVITE
  1298. User-Agent: Asterisk PBX
  1299. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  1300. Supported: replaces
  1301. Contact: <sip:[email protected]>
  1302. Content-Type: application/sdp
  1303. Content-Length: 240
  1304.  
  1305. v=0
  1306. o=root 1562 1562 IN IP4 192.168.10.10
  1307. s=session
  1308. c=IN IP4 192.168.10.10
  1309. t=0 0
  1310. m=audio 10000 RTP/AVP 8 101
  1311. a=rtpmap:8 PCMA/8000
  1312. a=rtpmap:101 telephone-event/8000
  1313. a=fmtp:101 0-16
  1314. a=silenceSupp:off - - - -
  1315. a=ptime:20
  1316. a=sendrecv
  1317.  
  1318. <------------>
  1319. [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1320. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1321. <--- SIP read from 192.168.10.252:5060 --->
  1322. ACK sip:[email protected] SIP/2.0
  1323. Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-lrtc2e94lbnh;rport
  1324. From: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  1325. To: <sip:[email protected]>;tag=as17f86ff1
  1326. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  1327. CSeq: 2 ACK
  1328. Max-Forwards: 70
  1329. Contact: <sip:[email protected]:5060>;reg-id=1
  1330. Content-Length: 0
  1331.  
  1332.  
  1333. <------------->
  1334. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: ACK sip:[email protected] SIP/2.0 (35)
  1335. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-lrtc2e94lbnh;rport (70)
  1336. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=ac9v354jjd (54)
  1337. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=as17f86ff1 (44)
  1338. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
  1339. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 ACK (11)
  1340. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  1341. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  1342. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
  1343. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: (0)
  1344. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (9 headers 0 lines) ---
  1345. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  1346. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #245
  1347. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on 'f1882b3c0d38-xtxggv3ikaiv' of Response 2: Match Found
  1348. [Feb 9 13:42:38] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1349. [Feb 9 13:42:38] DEBUG[1643] rtp.c: Ooh, format changed from unknown to alaw
  1350. [Feb 9 13:42:38] DEBUG[1643] rtp.c: Created smoother: format: 8 ms: 20 len: 160
  1351. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1352. <--- SIP read from 192.168.10.253:5060 --->
  1353. SIP/2.0 487 Request Terminated
  1354. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
  1355. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1356. To: <sip:[email protected]:5060>;tag=p88g5axhr5
  1357. Call-ID: [email protected]
  1358. CSeq: 102 INVITE
  1359. Contact: <sip:[email protected]:5060>;flow-id=1
  1360. Content-Length: 0
  1361.  
  1362.  
  1363. <------------->
  1364. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
  1365. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
  1366. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as7741abae (54)
  1367. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]:5060>;tag=p88g5axhr5 (50)
  1368. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: [email protected] (55)
  1369. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
  1370. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;flow-id=1 (50)
  1371. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
  1372. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: (0)
  1373. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (8 headers 0 lines) ---
  1374. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Acked pending invite 102
  1375. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
  1376. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP response 487 to standard invite
  1377. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Transmitting (NAT) to 192.168.10.253:5060:
  1378. ACK sip:[email protected]:5060 SIP/2.0
  1379. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1380. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1381. To: <sip:[email protected]:5060>;tag=p88g5axhr5
  1382. Contact: <sip:[email protected]>
  1383. Call-ID: [email protected]
  1384. CSeq: 102 ACK
  1385. User-Agent: Asterisk PBX
  1386. Max-Forwards: 70
  1387. Content-Length: 0
  1388.  
  1389.  
  1390. ---
  1391. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Updating call counter for outgoing call
  1392. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Call to peer '10012' removed from call limit 50
  1393. [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
  1394. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting SIP_ALREADYGONE on dialog [email protected]
  1395. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1396. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1397. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
  1398. [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
  1399. [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1400. [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
  1401. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1402. <--- SIP read from 192.168.10.251:5060 --->
  1403. ACK sip:[email protected] SIP/2.0
  1404. Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-5cikq0utxig2;rport
  1405. From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  1406. To: "Test3" <sip:[email protected]>;tag=as673f2fe0
  1407. Call-ID: 3c27bc5ad168-wohs43kd06vu
  1408. CSeq: 2 ACK
  1409. Max-Forwards: 70
  1410. Contact: <sip:[email protected]:5060>;reg-id=1
  1411. Content-Length: 0
  1412.  
  1413.  
  1414. <------------->
  1415. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: ACK sip:[email protected] SIP/2.0 (35)
  1416. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-5cikq0utxig2;rport (70)
  1417. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:[email protected]>;tag=j0kehrl2e6 (54)
  1418. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:[email protected]>;tag=as673f2fe0 (52)
  1419. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
  1420. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 ACK (11)
  1421. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
  1422. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  1423. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
  1424. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: (0)
  1425. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (9 headers 0 lines) ---
  1426. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
  1427. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #240
  1428. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bc5ad168-wohs43kd06vu' of Response 2: Match Found
  1429. [Feb 9 13:42:38] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1430. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #239 (1) NOTIFY - 4
  1431. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #239))
  1432. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Retransmitting #1 (NAT) to 192.168.10.253:5060:
  1433. NOTIFY sip:[email protected]:5060 SIP/2.0
  1434. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport
  1435. From: <sip:[email protected]>;tag=as68be196f
  1436. To: <sip:[email protected]>;tag=nw4c6bdjzk
  1437. Contact: <sip:[email protected]>
  1438. Call-ID: 3c2d44997030-61xfuv9xjhtd
  1439. CSeq: 113 NOTIFY
  1440. User-Agent: Asterisk PBX
  1441. Max-Forwards: 70
  1442. Event: dialog
  1443. Content-Type: application/dialog-info+xml
  1444. Subscription-State: active
  1445. Content-Length: 208
  1446.  
  1447. <?xml version="1.0"?>
  1448. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:[email protected]">
  1449. <dialog id="10014">
  1450. <state>confirmed</state>
  1451. </dialog>
  1452. </dialog-info>
  1453.  
  1454. ---
  1455. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (1) CANCEL - 14
  1456. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #242))
  1457. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Retransmitting #1 (NAT) to 192.168.10.253:5060:
  1458. CANCEL sip:[email protected]:5060 SIP/2.0
  1459. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1460. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1461. To: <sip:[email protected]:5060>
  1462. Call-ID: [email protected]
  1463. CSeq: 102 CANCEL
  1464. User-Agent: Asterisk PBX
  1465. Max-Forwards: 70
  1466. Content-Length: 0
  1467.  
  1468.  
  1469. ---
  1470. [Feb 9 13:42:38] DEBUG[1643] rtp.c: Ooh, format changed from unknown to alaw
  1471. [Feb 9 13:42:38] DEBUG[1643] rtp.c: Created smoother: format: 8 ms: 20 len: 160
  1472. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1473. <--- SIP read from 192.168.10.251:5060 --->
  1474. SIP/2.0 200 Ok
  1475. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK05ef9d45;rport=5060
  1476. From: <sip:[email protected]>;tag=as31f661fa
  1477. To: <sip:[email protected]>;tag=ttbbodh6oe
  1478. Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
  1479. CSeq: 114 NOTIFY
  1480. Content-Length: 0
  1481.  
  1482.  
  1483. <------------->
  1484. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
  1485. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK05ef9d45;rport=5060 (69)
  1486. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as31f661fa (46)
  1487. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=ttbbodh6oe (44)
  1488. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu (34)
  1489. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 114 NOTIFY (16)
  1490. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
  1491. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: (0)
  1492. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (7 headers 0 lines) ---
  1493. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Acked pending invite 114
  1494. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #244
  1495. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb6fc4-t8z10gt0v8fu' of Request 114: Match Found
  1496. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] SIP Response message for INCOMING dialog NOTIFY arrived
  1497. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1498. <--- SIP read from 192.168.10.253:5060 --->
  1499. SIP/2.0 200 Ok
  1500. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport=5060
  1501. From: <sip:[email protected]>;tag=as68be196f
  1502. To: <sip:[email protected]>;tag=nw4c6bdjzk
  1503. Call-ID: 3c2d44997030-61xfuv9xjhtd
  1504. CSeq: 113 NOTIFY
  1505. Content-Length: 0
  1506.  
  1507.  
  1508. <------------->
  1509. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
  1510. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport=5060 (69)
  1511. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as68be196f (46)
  1512. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=nw4c6bdjzk (44)
  1513. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c2d44997030-61xfuv9xjhtd (34)
  1514. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
  1515. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
  1516. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: (0)
  1517. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (7 headers 0 lines) ---
  1518. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Acked pending invite 113
  1519. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #239
  1520. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c2d44997030-61xfuv9xjhtd' of Request 113: Match Found
  1521. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] SIP Response message for INCOMING dialog NOTIFY arrived
  1522. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
  1523. <--- SIP read from 192.168.10.253:5060 --->
  1524. SIP/2.0 487 Request Terminated
  1525. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
  1526. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1527. To: <sip:[email protected]:5060>;tag=p88g5axhr5
  1528. Call-ID: [email protected]
  1529. CSeq: 102 INVITE
  1530. Contact: <sip:[email protected]:5060>;flow-id=1
  1531. Content-Length: 0
  1532.  
  1533.  
  1534. <------------->
  1535. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
  1536. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
  1537. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as7741abae (54)
  1538. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]:5060>;tag=p88g5axhr5 (50)
  1539. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: [email protected] (55)
  1540. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
  1541. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;flow-id=1 (50)
  1542. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
  1543. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: (0)
  1544. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (8 headers 0 lines) ---
  1545. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Not Found
  1546. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (2) CANCEL - 14
  1547. [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #242))
  1548. [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Retransmitting #2 (NAT) to 192.168.10.253:5060:
  1549. CANCEL sip:[email protected]:5060 SIP/2.0
  1550. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1551. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1552. To: <sip:[email protected]:5060>
  1553. Call-ID: [email protected]
  1554. CSeq: 102 CANCEL
  1555. User-Agent: Asterisk PBX
  1556. Max-Forwards: 70
  1557. Content-Length: 0
  1558.  
  1559.  
  1560. ---
  1561. [Feb 9 13:42:39] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (3) CANCEL - 14
  1562. [Feb 9 13:42:39] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #242))
  1563. [Feb 9 13:42:39] VERBOSE[1571] logger.c: [Feb 9 13:42:39] Retransmitting #3 (NAT) to 192.168.10.253:5060:
  1564. CANCEL sip:[email protected]:5060 SIP/2.0
  1565. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1566. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1567. To: <sip:[email protected]:5060>
  1568. Call-ID: [email protected]
  1569. CSeq: 102 CANCEL
  1570. User-Agent: Asterisk PBX
  1571. Max-Forwards: 70
  1572. Content-Length: 0
  1573.  
  1574.  
  1575. ---
  1576. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (4) CANCEL - 14
  1577. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #242))
  1578. [Feb 9 13:42:40] VERBOSE[1571] logger.c: [Feb 9 13:42:40] Retransmitting #4 (NAT) to 192.168.10.253:5060:
  1579. CANCEL sip:[email protected]:5060 SIP/2.0
  1580. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1581. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1582. To: <sip:[email protected]:5060>
  1583. Call-ID: [email protected]
  1584. CSeq: 102 CANCEL
  1585. User-Agent: Asterisk PBX
  1586. Max-Forwards: 70
  1587. Content-Length: 0
  1588.  
  1589.  
  1590. ---
  1591. [Feb 9 13:42:40] VERBOSE[1571] logger.c: [Feb 9 13:42:40]
  1592. <--- SIP read from 192.168.10.253:5060 --->
  1593. SIP/2.0 487 Request Terminated
  1594. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
  1595. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1596. To: <sip:[email protected]:5060>;tag=p88g5axhr5
  1597. Call-ID: [email protected]
  1598. CSeq: 102 INVITE
  1599. Contact: <sip:[email protected]:5060>;flow-id=1
  1600. Content-Length: 0
  1601.  
  1602.  
  1603. <------------->
  1604. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
  1605. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
  1606. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as7741abae (54)
  1607. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]:5060>;tag=p88g5axhr5 (50)
  1608. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 4: Call-ID: [email protected] (55)
  1609. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
  1610. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;flow-id=1 (50)
  1611. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
  1612. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 8: (0)
  1613. [Feb 9 13:42:40] VERBOSE[1571] logger.c: [Feb 9 13:42:40] --- (8 headers 0 lines) ---
  1614. [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Not Found
  1615. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (5) CANCEL - 14
  1616. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #242))
  1617. [Feb 9 13:42:41] VERBOSE[1571] logger.c: [Feb 9 13:42:41] Retransmitting #5 (NAT) to 192.168.10.253:5060:
  1618. CANCEL sip:[email protected]:5060 SIP/2.0
  1619. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1620. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1621. To: <sip:[email protected]:5060>
  1622. Call-ID: [email protected]
  1623. CSeq: 102 CANCEL
  1624. User-Agent: Asterisk PBX
  1625. Max-Forwards: 70
  1626. Content-Length: 0
  1627.  
  1628.  
  1629. ---
  1630. [Feb 9 13:42:41] VERBOSE[1571] logger.c: [Feb 9 13:42:41]
  1631. <--- SIP read from 192.168.10.253:5060 --->
  1632. SIP/2.0 487 Request Terminated
  1633. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
  1634. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1635. To: <sip:[email protected]:5060>;tag=p88g5axhr5
  1636. Call-ID: [email protected]
  1637. CSeq: 102 INVITE
  1638. Contact: <sip:[email protected]:5060>;flow-id=1
  1639. Content-Length: 0
  1640.  
  1641.  
  1642. <------------->
  1643. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
  1644. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
  1645. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as7741abae (54)
  1646. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]:5060>;tag=p88g5axhr5 (50)
  1647. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 4: Call-ID: [email protected] (55)
  1648. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
  1649. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;flow-id=1 (50)
  1650. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
  1651. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 8: (0)
  1652. [Feb 9 13:42:41] VERBOSE[1571] logger.c: [Feb 9 13:42:41] --- (8 headers 0 lines) ---
  1653. [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Not Found
  1654. [Feb 9 13:42:43] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1655. [Feb 9 13:42:43] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1656. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (6) CANCEL - 14
  1657. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #242))
  1658. [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44] Retransmitting #6 (NAT) to 192.168.10.253:5060:
  1659. CANCEL sip:[email protected]:5060 SIP/2.0
  1660. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
  1661. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1662. To: <sip:[email protected]:5060>
  1663. Call-ID: [email protected]
  1664. CSeq: 102 CANCEL
  1665. User-Agent: Asterisk PBX
  1666. Max-Forwards: 70
  1667. Content-Length: 0
  1668.  
  1669.  
  1670. ---
  1671. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Auto destroying SIP dialog '[email protected]'
  1672. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Destroying SIP dialog [email protected]
  1673. [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44] Really destroying SIP dialog '[email protected]' Method: INVITE
  1674. [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44]
  1675. <--- SIP read from 192.168.10.253:5060 --->
  1676. SIP/2.0 487 Request Terminated
  1677. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
  1678. From: "Test3" <sip:[email protected]>;tag=as7741abae
  1679. To: <sip:[email protected]:5060>;tag=p88g5axhr5
  1680. Call-ID: [email protected]
  1681. CSeq: 102 INVITE
  1682. Contact: <sip:[email protected]:5060>;flow-id=1
  1683. Content-Length: 0
  1684.  
  1685.  
  1686. <------------->
  1687. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
  1688. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
  1689. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as7741abae (54)
  1690. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]:5060>;tag=p88g5axhr5 (50)
  1691. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 4: Call-ID: [email protected] (55)
  1692. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
  1693. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;flow-id=1 (50)
  1694. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
  1695. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 8: (0)
  1696. [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44] --- (8 headers 0 lines) ---
  1697. [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Invalid SIP message - rejected , no callid, len 359
  1698. [Feb 9 13:42:47] VERBOSE[1642] logger.c: [Feb 9 13:42:47] -- Remote UNIX connection disconnected
  1699. [Feb 9 13:42:48] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1700. [Feb 9 13:42:48] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1701. [Feb 9 13:42:53] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1702. [Feb 9 13:42:53] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
  1703. [Feb 9 13:42:56] DEBUG[1562] channel.c: Soft-Hanging up channel 'SIP/10014-0994c8c8'
  1704. [Feb 9 13:42:56] DEBUG[1562] channel.c: Soft-Hanging up channel 'SIP/10013-0993e6e0'
  1705. [Feb 9 13:42:56] DEBUG[1643] channel.c: Didn't get a frame from channel: SIP/10014-0994c8c8
  1706. [Feb 9 13:42:56] DEBUG[1643] channel.c: Bridge stops bridging channels SIP/10013-0993e6e0 and SIP/10014-0994c8c8
  1707. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'Macro'
  1708. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [h@macro-dial:1] Macro("SIP/10013-0993e6e0", "hangupcall") in new stack
  1709. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Expression result is '1'
  1710. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'GotoIf'
  1711. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:1] GotoIf("SIP/10013-0993e6e0", "1?skiprg") in new stack
  1712. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Goto (macro-hangupcall,s,4)
  1713. [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Executed application: GotoIf
  1714. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Expression result is '1'
  1715. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'GotoIf'
  1716. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:4] GotoIf("SIP/10013-0993e6e0", "1?skipblkvm") in new stack
  1717. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Goto (macro-hangupcall,s,7)
  1718. [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Executed application: GotoIf
  1719. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Expression result is '1'
  1720. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'GotoIf'
  1721. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:7] GotoIf("SIP/10013-0993e6e0", "1?theend") in new stack
  1722. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Goto (macro-hangupcall,s,9)
  1723. [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Executed application: GotoIf
  1724. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'Hangup'
  1725. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:9] Hangup("SIP/10013-0993e6e0", "") in new stack
  1726. [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Spawn extension (macro-hangupcall,s,9) exited non-zero on 'SIP/10013-0993e6e0' in macro 'hangupcall'
  1727. [Feb 9 13:42:56] DEBUG[1643] res_features.c: Spawn h extension (macro-dial,h,1) exited non-zero on 'SIP/10013-0993e6e0'
  1728. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] == Spawn h extension (macro-dial, h, 1) exited non-zero on 'SIP/10013-0993e6e0'
  1729. [Feb 9 13:42:56] DEBUG[1643] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
  1730. [Feb 9 13:42:56] DEBUG[1643] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield) VALUES ('2010-02-09 13:42:36','\"Test3\" <10013>','10013','10012','from-internal', 'SIP/10013-0993e6e0','SIP/10014-0994c8c8','Dial','SIP/10012||tr',20,18,'ANSWERED',3,'','')
  1731. [Feb 9 13:42:56] DEBUG[1643] channel.c: Hanging up channel 'SIP/10014-0994c8c8'
  1732. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Hangup call SIP/10014-0994c8c8, SIP callid 3c27bc5ad168-wohs43kd06vu)
  1733. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: update_call_counter(10014) - decrement call limit counter on hangup
  1734. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Updating call counter for incoming call
  1735. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Call from peer '10014' removed from call limit 50
  1736. [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
  1737. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1738. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Scheduling destruction of SIP dialog '3c27bc5ad168-wohs43kd06vu' in 32000 ms (Method: ACK)
  1739. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1740. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 1 (Not in use)
  1741. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Strict routing enforced for session 3c27bc5ad168-wohs43kd06vu
  1742. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1743. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  1744. [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1745. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1746. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.251, port 5060
  1747. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.251:5060:
  1748. BYE sip:[email protected]:5060 SIP/2.0
  1749. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK6793e7c9;rport
  1750. From: "Test3" <sip:[email protected]>;tag=as673f2fe0
  1751. To: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  1752. Call-ID: 3c27bc5ad168-wohs43kd06vu
  1753. CSeq: 102 BYE
  1754. User-Agent: Asterisk PBX
  1755. Max-Forwards: 70
  1756. X-Asterisk-HangupCause: Normal Clearing
  1757. X-Asterisk-HangupCauseCode: 16
  1758. Content-Length: 0
  1759.  
  1760.  
  1761. ---
  1762. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1763. [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
  1764. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1765. [Feb 9 13:42:56] DEBUG[1643] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
  1766. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1767. [Feb 9 13:42:56] DEBUG[1643] app_dial.c: Exiting with DIALSTATUS=ANSWER.
  1768. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44997030-61xfuv9xjhtd
  1769. [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Spawn extension (macro-dial,s,7) exited non-zero on 'SIP/10013-0993e6e0' in macro 'dial'
  1770. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  1771. [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Spawn extension (macro-exten-vm,s,9) exited non-zero on 'SIP/10013-0993e6e0' in macro 'exten-vm'
  1772. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.253, port 5060
  1773. [Feb 9 13:42:56] DEBUG[1643] pbx.c: Spawn extension (from-internal,10012,1) exited non-zero on 'SIP/10013-0993e6e0'
  1774. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.253:5060:
  1775. NOTIFY sip:[email protected]:5060 SIP/2.0
  1776. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK52b93ef3;rport
  1777. From: <sip:[email protected]>;tag=as68be196f
  1778. To: <sip:[email protected]>;tag=nw4c6bdjzk
  1779. Contact: <sip:[email protected]>
  1780. Call-ID: 3c2d44997030-61xfuv9xjhtd
  1781. CSeq: 114 NOTIFY
  1782. User-Agent: Asterisk PBX
  1783. Max-Forwards: 70
  1784. Event: dialog
  1785. Content-Type: application/dialog-info+xml
  1786. Subscription-State: active
  1787. Content-Length: 209
  1788.  
  1789. <?xml version="1.0"?>
  1790. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:[email protected]">
  1791. <dialog id="10014">
  1792. <state>terminated</state>
  1793. </dialog>
  1794. </dialog-info>
  1795.  
  1796. ---
  1797. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] == Spawn extension (from-internal, 10012, 1) exited non-zero on 'SIP/10013-0993e6e0'
  1798. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1799. [Feb 9 13:42:56] DEBUG[1643] channel.c: Soft-Hanging up channel 'SIP/10013-0993e6e0'
  1800. [Feb 9 13:42:56] DEBUG[1643] channel.c: Hanging up channel 'SIP/10013-0993e6e0'
  1801. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Extension Changed 10014[ext-local] new state Idle for Notify User 10012
  1802. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Hangup call SIP/10013-0993e6e0, SIP callid f1882b3c0d38-xtxggv3ikaiv)
  1803. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: update_call_counter(10013) - decrement call limit counter on hangup
  1804. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1805. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Updating call counter for incoming call
  1806. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1807. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Call from peer '10013' removed from call limit 50
  1808. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 1 (Not in use)
  1809. [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
  1810. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
  1811. [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1812. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Scheduling destruction of SIP dialog 'f1882b3c0d38-xtxggv3ikaiv' in 32000 ms (Method: ACK)
  1813. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Strict routing enforced for session f1882b3c0d38-xtxggv3ikaiv
  1814. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  1815. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
  1816. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.252, port 5060
  1817. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1818. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1819. [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.252:5060:
  1820. BYE sip:[email protected]:5060 SIP/2.0
  1821. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK68b5a119;rport
  1822. From: <sip:[email protected]>;tag=as17f86ff1
  1823. To: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  1824. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  1825. CSeq: 102 BYE
  1826. User-Agent: Asterisk PBX
  1827. Max-Forwards: 70
  1828. X-Asterisk-HangupCause: Normal Clearing
  1829. X-Asterisk-HangupCauseCode: 16
  1830. Content-Length: 0
  1831.  
  1832.  
  1833. ---
  1834. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 1 (Not in use)
  1835. [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1836. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1837. [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1838. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1839. [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
  1840. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1841. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1842. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44996af4-i4ocu4zvtwre
  1843. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  1844. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.253, port 5060
  1845. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.253:5060:
  1846. NOTIFY sip:[email protected]:5060 SIP/2.0
  1847. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0ed066d1;rport
  1848. From: <sip:[email protected]>;tag=as6d58a6fc
  1849. To: <sip:[email protected]>;tag=5e5cbuyyd1
  1850. Contact: <sip:[email protected]>
  1851. Call-ID: 3c2d44996af4-i4ocu4zvtwre
  1852. CSeq: 114 NOTIFY
  1853. User-Agent: Asterisk PBX
  1854. Max-Forwards: 70
  1855. Event: dialog
  1856. Content-Type: application/dialog-info+xml
  1857. Subscription-State: active
  1858. Content-Length: 209
  1859.  
  1860. <?xml version="1.0"?>
  1861. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:[email protected]">
  1862. <dialog id="10013">
  1863. <state>terminated</state>
  1864. </dialog>
  1865. </dialog-info>
  1866.  
  1867. ---
  1868. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1869. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Extension Changed 10013[ext-local] new state Idle for Notify User 10012
  1870. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1871. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1872. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb7594-mfl0ssslkb6l
  1873. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  1874. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.251, port 5060
  1875. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.251:5060:
  1876. NOTIFY sip:[email protected]:5060 SIP/2.0
  1877. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5e26394a;rport
  1878. From: <sip:[email protected]>;tag=as647499f8
  1879. To: <sip:[email protected]>;tag=wesbz3i941
  1880. Contact: <sip:[email protected]>
  1881. Call-ID: 3c27bbfb7594-mfl0ssslkb6l
  1882. CSeq: 114 NOTIFY
  1883. User-Agent: Asterisk PBX
  1884. Max-Forwards: 70
  1885. Event: dialog
  1886. Content-Type: application/dialog-info+xml
  1887. Subscription-State: active
  1888. Content-Length: 209
  1889.  
  1890. <?xml version="1.0"?>
  1891. <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:[email protected]">
  1892. <dialog id="10013">
  1893. <state>terminated</state>
  1894. </dialog>
  1895. </dialog-info>
  1896.  
  1897. ---
  1898. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
  1899. [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Extension Changed 10013[ext-local] new state Idle for Notify User 10014
  1900. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1901. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1902. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 1 (Not in use)
  1903. [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
  1904. [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
  1905. [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  1906. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
  1907. <--- SIP read from 192.168.10.253:5060 --->
  1908. SIP/2.0 200 Ok
  1909. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK52b93ef3;rport=5060
  1910. From: <sip:[email protected]>;tag=as68be196f
  1911. To: <sip:[email protected]>;tag=nw4c6bdjzk
  1912. Call-ID: 3c2d44997030-61xfuv9xjhtd
  1913. CSeq: 114 NOTIFY
  1914. Content-Length: 0
  1915.  
  1916.  
  1917. <------------->
  1918. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
  1919. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK52b93ef3;rport=5060 (69)
  1920. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as68be196f (46)
  1921. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=nw4c6bdjzk (44)
  1922. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c2d44997030-61xfuv9xjhtd (34)
  1923. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 114 NOTIFY (16)
  1924. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
  1925. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: (0)
  1926. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (7 headers 0 lines) ---
  1927. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Acked pending invite 114
  1928. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #250
  1929. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c2d44997030-61xfuv9xjhtd' of Request 114: Match Found
  1930. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog NOTIFY arrived
  1931. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
  1932. <--- SIP read from 192.168.10.252:5060 --->
  1933. SIP/2.0 200 OK
  1934. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK68b5a119;rport=5060
  1935. From: <sip:[email protected]>;tag=as17f86ff1
  1936. To: "Test3" <sip:[email protected]>;tag=ac9v354jjd
  1937. Call-ID: f1882b3c0d38-xtxggv3ikaiv
  1938. CSeq: 102 BYE
  1939. Contact: <sip:[email protected]:5060>;reg-id=1
  1940. User-Agent: snom870/8.3.6
  1941. RTP-RxStat: Total_Rx_Pkts=899,Rx_Pkts=899,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
  1942. RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=939,Remote_Tx_Pkts=740
  1943. Content-Length: 0
  1944.  
  1945.  
  1946. <------------->
  1947. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
  1948. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK68b5a119;rport=5060 (69)
  1949. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as17f86ff1 (46)
  1950. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:[email protected]>;tag=ac9v354jjd (52)
  1951. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
  1952. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 BYE (13)
  1953. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  1954. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: User-Agent: snom870/8.3.6 (25)
  1955. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 8: RTP-RxStat: Total_Rx_Pkts=899,Rx_Pkts=899,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (78)
  1956. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 9: RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=939,Remote_Tx_Pkts=740 (58)
  1957. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 10: Content-Length: 0 (17)
  1958. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 11: (0)
  1959. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (11 headers 0 lines) ---
  1960. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #252
  1961. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on 'f1882b3c0d38-xtxggv3ikaiv' of Request 102: Match Found
  1962. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog BYE arrived
  1963. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] Really destroying SIP dialog 'f1882b3c0d38-xtxggv3ikaiv' Method: ACK
  1964. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
  1965. <--- SIP read from 192.168.10.251:5060 --->
  1966. SIP/2.0 200 Ok
  1967. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5e26394a;rport=5060
  1968. From: <sip:[email protected]>;tag=as647499f8
  1969. To: <sip:[email protected]>;tag=wesbz3i941
  1970. Call-ID: 3c27bbfb7594-mfl0ssslkb6l
  1971. CSeq: 114 NOTIFY
  1972. Content-Length: 0
  1973.  
  1974.  
  1975. <------------->
  1976. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
  1977. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5e26394a;rport=5060 (69)
  1978. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: <sip:[email protected]>;tag=as647499f8 (46)
  1979. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: <sip:[email protected]>;tag=wesbz3i941 (44)
  1980. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb7594-mfl0ssslkb6l (34)
  1981. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 114 NOTIFY (16)
  1982. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
  1983. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: (0)
  1984. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (7 headers 0 lines) ---
  1985. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Acked pending invite 114
  1986. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #254
  1987. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb7594-mfl0ssslkb6l' of Request 114: Match Found
  1988. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog NOTIFY arrived
  1989. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
  1990. <--- SIP read from 192.168.10.251:5060 --->
  1991. SIP/2.0 200 OK
  1992. Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK6793e7c9;rport=5060
  1993. From: "Test3" <sip:[email protected]>;tag=as673f2fe0
  1994. To: "Test4" <sip:[email protected]>;tag=j0kehrl2e6
  1995. Call-ID: 3c27bc5ad168-wohs43kd06vu
  1996. CSeq: 102 BYE
  1997. Contact: <sip:[email protected]:5060>;reg-id=1
  1998. User-Agent: snom360/7.3.30
  1999. RTP-RxStat: Total_Rx_Pkts=910,Rx_Pkts=910,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
  2000. RTP-TxStat: Total_Tx_Pkts=919,Tx_Pkts=919,Remote_Tx_Pkts=751
  2001. Content-Length: 0
  2002.  
  2003.  
  2004. <------------->
  2005. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
  2006. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK6793e7c9;rport=5060 (69)
  2007. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:[email protected]>;tag=as673f2fe0 (54)
  2008. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: "Test4" <sip:[email protected]>;tag=j0kehrl2e6 (52)
  2009. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
  2010. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 BYE (13)
  2011. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:[email protected]:5060>;reg-id=1 (49)
  2012. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: User-Agent: snom360/7.3.30 (26)
  2013. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 8: RTP-RxStat: Total_Rx_Pkts=910,Rx_Pkts=910,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (78)
  2014. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 9: RTP-TxStat: Total_Tx_Pkts=919,Tx_Pkts=919,Remote_Tx_Pkts=751 (60)
  2015. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 10: Content-Length: 0 (17)
  2016. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 11: (0)
  2017. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (11 headers 0 lines) ---
  2018. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #249
  2019. [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bc5ad168-wohs43kd06vu' of Request 102: Match Found
  2020. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog BYE arrived
  2021. [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] Really destroying SIP dialog '3c27bc5ad168-wohs43kd06vu' Method: ACK
  2022. [Feb 9 13:42:56] VERBOSE[1562] logger.c: [Feb 9 13:42:56] Executing last minute cleanups
  2023. [Feb 9 13:42:56] VERBOSE[1562] logger.c: [Feb 9 13:42:56] == Destroying musiconhold processes
  2024. [Feb 9 13:42:56] DEBUG[1562] res_musiconhold.c: Destroying MOH class 'default'
  2025. [Feb 9 13:42:56] DEBUG[1562] res_musiconhold.c: Destroying MOH class 'none'
  2026. [Feb 9 13:42:56] DEBUG[1562] asterisk.c: Asterisk ending (0).
  2027.  
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