API tools faq
paste
Guest User

Untitled

a guest
Jul 18th, 2018
182
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 10.92 KB | None | 0 0
raw download clone embed print report
  1.  
  2. pbxtra6644*CLI>
  3. -- Executing GotoIf("SIP/000B8220FBD9-08d41528", "0?20") in new stack
  4. -- Executing Set("SIP/000B8220FBD9-08d41528", "EXTENSION=7003") in new stack
  5. -- Executing AGI("SIP/000B8220FBD9-08d41528", "fon://localhost:4574") in new stack
  6. -- AGI Script Executing Application: (SetCIDName) Options: (Chris Gibbin)
  7. -- AGI Script Executing Application: (SetCIDNum) Options: (3103028900)
  8. -- AGI Script fon://localhost:4574 completed, returning 0
  9. -- Executing Dial("SIP/000B8220FBD9-08d41528", "SIP/speakeasysip/18773662548") in new stack
  10. We're at 66.253.120.98 port 19652
  11. Adding codec 0x4 (ulaw) to SDP
  12. Adding codec 0x8 (alaw) to SDP
  13. Adding codec 0x2 (gsm) to SDP
  14. Adding non-codec 0x1 (telephone-event) to SDP
  15. 13 headers, 12 lines
  16. Reliably Transmitting (NAT) to 64.81.79.177:5060:
  17. INVITE sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net SIP/2.0
  18. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK2f6fe64a;rport
  19. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  20. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>
  21. Contact: <sip:3103028900@66.253.120.98>
  22. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  23. CSeq: 102 INVITE
  24. User-Agent: Asterisk PBX
  25. Max-Forwards: 70
  26. Date: Wed, 25 Nov 2009 19:04:11 GMT
  27. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  28. Content-Type: application/sdp
  29. Content-Length: 263
  30.  
  31. v=0
  32. o=root 2603 2603 IN IP4 66.253.120.98
  33. s=session
  34. c=IN IP4 66.253.120.98
  35. t=0 0
  36. m=audio 19652 RTP/AVP 0 8 3 101
  37. a=rtpmap:0 PCMU/8000
  38. a=rtpmap:8 PCMA/8000
  39. a=rtpmap:3 GSM/8000
  40. a=rtpmap:101 telephone-event/8000
  41. a=fmtp:101 0-16
  42. a=silenceSupp:off - - - -
  43.  
  44. ---
  45. -- Called speakeasysip/18773662548
  46. pbxtra6644*CLI>
  47. <-- SIP read from 64.81.79.177:5060:
  48. SIP/2.0 100 Trying
  49. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK2f6fe64a;rport=5060
  50. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  51. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>
  52. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  53. CSeq: 102 INVITE
  54.  
  55.  
  56. --- (6 headers 0 lines) ---
  57. pbxtra6644*CLI>
  58. <-- SIP read from 64.81.79.177:5060:
  59. SIP/2.0 401 Unauthorized
  60. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK2f6fe64a;rport=5060
  61. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  62. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>;tag=SDa9s0999-2088899767-1259175851038
  63. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  64. CSeq: 102 INVITE
  65. WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXg2ggsgtaT9usvapBW",algorithm=MD5,realm="BroadWorks"
  66. Content-Length: 0
  67.  
  68.  
  69. --- (8 headers 0 lines) ---
  70. Transmitting (NAT) to 64.81.79.177:5060:
  71. ACK sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net SIP/2.0
  72. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK2f6fe64a;rport
  73. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  74. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>;tag=SDa9s0999-2088899767-1259175851038
  75. Contact: <sip:3103028900@66.253.120.98>
  76. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  77. CSeq: 102 ACK
  78. User-Agent: Asterisk PBX
  79. Max-Forwards: 70
  80. Content-Length: 0
  81.  
  82.  
  83. ---
  84. We're at 66.253.120.98 port 19652
  85. Adding codec 0x4 (ulaw) to SDP
  86. Adding codec 0x8 (alaw) to SDP
  87. Adding codec 0x2 (gsm) to SDP
  88. Adding non-codec 0x1 (telephone-event) to SDP
  89. Reliably Transmitting (NAT) to 64.81.79.177:5060:
  90. INVITE sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net SIP/2.0
  91. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK10733d4c;rport
  92. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  93. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>
  94. Contact: <sip:3103028900@66.253.120.98>
  95. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  96. CSeq: 103 INVITE
  97. User-Agent: Asterisk PBX
  98. Max-Forwards: 70
  99. Authorization: Digest username="9000007041", realm="BroadWorks", algorithm=MD5, uri="sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net", nonce="BroadWorksXg2ggsgtaT9usvapBW", response="6fba149a9946c0747a38053000713cb2", opaque="", qop=auth, cnonce="7d7e123a", nc=00000001
  100. Date: Wed, 25 Nov 2009 19:04:11 GMT
  101. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  102. Content-Type: application/sdp
  103. Content-Length: 263
  104.  
  105. v=0
  106. o=root 2603 2604 IN IP4 66.253.120.98
  107. s=session
  108. c=IN IP4 66.253.120.98
  109. t=0 0
  110. m=audio 19652 RTP/AVP 0 8 3 101
  111. a=rtpmap:0 PCMU/8000
  112. a=rtpmap:8 PCMA/8000
  113. a=rtpmap:3 GSM/8000
  114. a=rtpmap:101 telephone-event/8000
  115. a=fmtp:101 0-16
  116. a=silenceSupp:off - - - -
  117.  
  118. ---
  119. pbxtra6644*CLI>
  120. <-- SIP read from 64.81.79.177:5060:
  121. SIP/2.0 100 Trying
  122. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK10733d4c;rport=5060
  123. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  124. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>
  125. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  126. CSeq: 103 INVITE
  127.  
  128.  
  129. --- (6 headers 0 lines) ---
  130. pbxtra6644*CLI>
  131. <-- SIP read from 64.81.79.177:5060:
  132. SIP/2.0 200 OK
  133. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK10733d4c;rport=5060
  134. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  135. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>;tag=SDa9s0999-1350687047-1259175855371
  136. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  137. CSeq: 103 INVITE
  138. Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
  139. Supported:
  140. Contact: <sip:18773662548@64.81.79.177:5060;transport=udp>
  141. Remote-Party-ID: <sip:18773662548@64.81.79.177;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber
  142. Accept: multipart/mixed,application/media_control+xml,application/sdp
  143. Content-Type: application/sdp
  144. Content-Length: 198
  145.  
  146. v=0
  147. o=- 822211736 1 IN IP4 64.81.79.177
  148. s=-
  149. c=IN IP4 64.81.79.177
  150. t=0 0
  151. m=audio 50580 RTP/AVP 0 101
  152. a=rtpmap:101 telephone-event/8000
  153. a=fmtp:101 0-16
  154. a=ptime:20
  155. a=bsoft: 1 image udptl t38
  156.  
  157. --- (13 headers 10 lines) ---
  158. Found RTP audio format 0
  159. Found RTP audio format 101
  160. Peer audio RTP is at port 64.81.79.177:50580
  161. Found description format telephone-event
  162. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  163. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  164. list_route: hop: <sip:18773662548@64.81.79.177:5060;transport=udp>
  165. set_destination: Parsing <sip:18773662548@64.81.79.177:5060;transport=udp> for address/port to send to
  166. set_destination: set destination to 64.81.79.177, port 5060
  167. Transmitting (NAT) to 64.81.79.177:5060:
  168. ACK sip:18773662548@64.81.79.177:5060;transport=udp SIP/2.0
  169. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK1cbb78f1;rport
  170. From: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  171. To: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>;tag=SDa9s0999-1350687047-1259175855371
  172. Contact: <sip:3103028900@66.253.120.98>
  173. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  174. CSeq: 103 ACK
  175. User-Agent: Asterisk PBX
  176. Max-Forwards: 70
  177. Content-Length: 0
  178.  
  179.  
  180. ---
  181. -- SIP/speakeasysip-08d44148 answered SIP/000B8220FBD9-08d41528
  182. -- Attempting native bridge of SIP/000B8220FBD9-08d41528 and SIP/speakeasysip-08d44148
  183. Destroying call '0ae4ecc572288a4c1dcae14a05305ba6@127.0.0.1'
  184. Nov 25 11:04:42 NOTICE[11065]: chan_sip.c:5553 sip_reregister: -- Re-registration for 9000007041@ca1-siptrunk-a.voice.speakeasy.net
  185. REGISTER 13 headers, 0 lines
  186. Reliably Transmitting (no NAT) to 64.81.79.177:5060:
  187. REGISTER sip:ca1-siptrunk-a.voice.speakeasy.net SIP/2.0
  188. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK46518096;rport
  189. From: <sip:9000007041@ca1-siptrunk-a.voice.speakeasy.net>;tag=as3543acfb
  190. To: <sip:9000007041@ca1-siptrunk-a.voice.speakeasy.net>
  191. Call-ID: 0ae4ecc572288a4c1dcae14a05305ba6@127.0.0.1
  192. CSeq: 107 REGISTER
  193. User-Agent: Asterisk PBX
  194. Max-Forwards: 70
  195. Authorization: Digest username="9000007041", realm="BroadWorks", algorithm=MD5, uri="sip:ca1-siptrunk-a.voice.speakeasy.net", nonce="BroadWorksXg2ggr7t4Tq1yjdiBW", response="60796bb2e49628ee23c80805bd878e80", opaque="", qop=auth, cnonce="63e19815", nc=00000003
  196. Expires: 120
  197. Contact: <sip:9000007041@66.253.120.98>
  198. Event: registration
  199. Content-Length: 0
  200.  
  201.  
  202. ---
  203. pbxtra6644*CLI>
  204. <-- SIP read from 64.81.79.177:5060:
  205. SIP/2.0 200 OK
  206. Via: SIP/2.0/UDP 66.253.120.98:5060;branch=z9hG4bK46518096;rport=5060
  207. From: <sip:9000007041@ca1-siptrunk-a.voice.speakeasy.net>;tag=as3543acfb
  208. To: <sip:9000007041@ca1-siptrunk-a.voice.speakeasy.net>;tag=aprqngpf110-jis8kj10000m6
  209. Call-ID: 0ae4ecc572288a4c1dcae14a05305ba6@127.0.0.1
  210. CSeq: 107 REGISTER
  211. Contact: <sip:9000007041@66.253.120.98>;expires=60
  212.  
  213.  
  214. --- (7 headers 0 lines) ---
  215. Scheduling destruction of call '0ae4ecc572288a4c1dcae14a05305ba6@127.0.0.1' in 32000 ms
  216. Nov 25 11:04:43 NOTICE[11065]: chan_sip.c:10165 handle_response_register: Outbound Registration: Expiry for ca1-siptrunk-a.voice.speakeasy.net is 60 sec (Scheduling reregistration in 45 s)
  217. pbxtra6644*CLI>
  218. <-- SIP read from 64.81.79.177:5060:
  219. BYE sip:3103028900@66.253.120.98 SIP/2.0
  220. Via: SIP/2.0/UDP 64.81.79.177:5060;branch=z9hG4bK4ts80i0088ig1ecgl2o0sdm8nk501.1
  221. From: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>;tag=SDa9s0999-1350687047-1259175855371
  222. To: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  223. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  224. CSeq: 375216704 BYE
  225. Max-Forwards: 9
  226. Content-Length: 0
  227.  
  228.  
  229. --- (8 headers 0 lines) ---
  230. Sending to 64.81.79.177 : 5060 (NAT)
  231. Transmitting (NAT) to 64.81.79.177:5060:
  232. SIP/2.0 200 OK
  233. Via: SIP/2.0/UDP 64.81.79.177:5060;branch=z9hG4bK4ts80i0088ig1ecgl2o0sdm8nk501.1;received=64.81.79.177
  234. From: <sip:18773662548@ca1-siptrunk-a.voice.speakeasy.net>;tag=SDa9s0999-1350687047-1259175855371
  235. To: "Chris Gibbin" <sip:3103028900@ca1-siptrunk-a.voice.speakeasy.net>;tag=as0800df53
  236. Call-ID: 6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net
  237. CSeq: 375216704 BYE
  238. User-Agent: Asterisk PBX
  239. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  240. Contact: <sip:3103028900@66.253.120.98>
  241. Content-Length: 0
  242. X-Asterisk-HangupCause: Normal Clearing
  243.  
  244.  
  245. ---
  246. == Spawn extension (internal, 918773662548, 4) exited non-zero on 'SIP/000B8220FBD9-08d41528'
  247. -- Executing ResetCDR("SIP/000B8220FBD9-08d41528", "w") in new stack
  248. -- Executing NoCDR("SIP/000B8220FBD9-08d41528", "") in new stack
  249. Nov 25 11:04:48 NOTICE[15157]: cdr.c:445 ast_cdr_free: CDR on channel 'SIP/000B8220FBD9-08d41528' lacks end
  250. -- Executing GotoIf("SIP/000B8220FBD9-08d41528", "1?5") in new stack
  251. -- Goto (internal,h,5)
  252. -- Executing Hangup("SIP/000B8220FBD9-08d41528", "") in new stack
  253. == Spawn extension (internal, h, 5) exited non-zero on 'SIP/000B8220FBD9-08d41528'
  254. Destroying call '6895c591006629a91d5a832e41f2dc10@ca1-siptrunk-a.voice.speakeasy.net'
Add Comment
Please, Sign In to add comment
Public Pastes
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!