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  1.  
  2. == Using SIP RTP CoS mark 5
  3. -- Executing [18885551212@default:1] Set("SIP/2928-00000072", "CALLERID(num)=12124504543") in new stack
  4. -- Executing [18885551212@default:2] Dial("SIP/2928-00000072", "SIP/voipcheap/18885551212") in new stack
  5. == Using SIP RTP CoS mark 5
  6. Audio is at 5060
  7. Adding codec 0x2 (gsm) to SDP
  8. Adding codec 0x8 (alaw) to SDP
  9. Adding codec 0x4 (ulaw) to SDP
  10. Adding codec 0x800 (g726) to SDP
  11. Adding non-codec 0x1 (telephone-event) to SDP
  12. Reliably Transmitting (NAT) to 77.72.169.134:5060:
  13. INVITE sip:18885551212@sip.voipcheap.com SIP/2.0
  14. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK2c211ff7;rport
  15. Max-Forwards: 70
  16. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  17. To: <sip:18885551212@sip.voipcheap.com>
  18. Contact: <sip:12124504543@194.90.1.5:5060>
  19. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  20. CSeq: 102 INVITE
  21. User-Agent: Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  22. Date: Wed, 04 Apr 2012 00:30:27 GMT
  23. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  24. Supported: replaces, timer
  25. Remote-Party-ID: "2928" <sip:12124504543@194.90.1.5>;party=calling;privacy=off;screen=no
  26. Content-Type: application/sdp
  27. Content-Length: 339
  28.  
  29. v=0
  30. o=root 1164249503 1164249503 IN IP4 194.90.1.5
  31. s=Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  32. c=IN IP4 194.90.1.5
  33. t=0 0
  34. m=audio 17396 RTP/AVP 3 8 0 111 101
  35. a=rtpmap:3 GSM/8000
  36. a=rtpmap:8 PCMA/8000
  37. a=rtpmap:0 PCMU/8000
  38. a=rtpmap:111 G726-32/8000
  39. a=rtpmap:101 telephone-event/8000
  40. a=fmtp:101 0-16
  41. a=ptime:20
  42. a=sendrecv
  43.  
  44. ---
  45. -- Called SIP/voipcheap/18885551212
  46.  
  47. <--- SIP read from UDP:77.72.169.134:5060 --->
  48. SIP/2.0 401 Unauthorized
  49. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK2c211ff7;rport
  50. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  51. To: <sip:18885551212@sip.voipcheap.com>
  52. Contact: sip:18885551212@77.72.169.134:5060
  53. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  54. CSeq: 102 INVITE
  55. Server: (Very nice Sip Registrar/Proxy Server)
  56. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  57. WWW-Authenticate: Digest realm="sipdiscount.com",nonce="3595867062",algorithm=MD5
  58. Content-Length: 0
  59.  
  60. <------------->
  61. --- (11 headers 0 lines) ---
  62. Transmitting (NAT) to 77.72.169.134:5060:
  63. ACK sip:18885551212@sip.voipcheap.com SIP/2.0
  64. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK2c211ff7;rport
  65. Max-Forwards: 70
  66. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  67. To: <sip:18885551212@sip.voipcheap.com>
  68. Contact: <sip:12124504543@194.90.1.5:5060>
  69. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  70. CSeq: 102 ACK
  71. User-Agent: Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  72. Content-Length: 0
  73.  
  74.  
  75. ---
  76. Audio is at 5060
  77. Adding codec 0x2 (gsm) to SDP
  78. Adding codec 0x8 (alaw) to SDP
  79. Adding codec 0x4 (ulaw) to SDP
  80. Adding codec 0x800 (g726) to SDP
  81. Adding non-codec 0x1 (telephone-event) to SDP
  82. Reliably Transmitting (NAT) to 77.72.169.134:5060:
  83. INVITE sip:18885551212@sip.voipcheap.com SIP/2.0
  84. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK06c93736;rport
  85. Max-Forwards: 70
  86. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  87. To: <sip:18885551212@sip.voipcheap.com>
  88. Contact: <sip:12124504543@194.90.1.5:5060>
  89. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  90. CSeq: 103 INVITE
  91. User-Agent: Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  92. Authorization: Digest username="myusername", realm="sipdiscount.com", algorithm=MD5, uri="sip:18885551212@sip.voipcheap.com", nonce="3595867062", response="a42dbd648188e6fde1482c32eeca24d6"
  93. Date: Wed, 04 Apr 2012 00:30:27 GMT
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  95. Supported: replaces, timer
  96. Remote-Party-ID: "2928" <sip:12124504543@194.90.1.5>;party=calling;privacy=off;screen=no
  97. Content-Type: application/sdp
  98. Content-Length: 339
  99.  
  100. v=0
  101. o=root 1164249503 1164249504 IN IP4 194.90.1.5
  102. s=Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  103. c=IN IP4 194.90.1.5
  104. t=0 0
  105. m=audio 17396 RTP/AVP 3 8 0 111 101
  106. a=rtpmap:3 GSM/8000
  107. a=rtpmap:8 PCMA/8000
  108. a=rtpmap:0 PCMU/8000
  109. a=rtpmap:111 G726-32/8000
  110. a=rtpmap:101 telephone-event/8000
  111. a=fmtp:101 0-16
  112. a=ptime:20
  113. a=sendrecv
  114.  
  115. ---
  116.  
  117. <--- SIP read from UDP:77.72.169.134:5060 --->
  118. SIP/2.0 100 Trying
  119. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK06c93736;rport
  120. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  121. To: <sip:18885551212@sip.voipcheap.com>
  122. Contact: sip:18885551212@77.72.169.134:5060
  123. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  124. CSeq: 103 INVITE
  125. Server: (Very nice Sip Registrar/Proxy Server)
  126. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  127. Content-Length: 0
  128.  
  129. <------------->
  130. --- (10 headers 0 lines) ---
  131.  
  132. <--- SIP read from UDP:77.72.169.134:5060 --->
  133. SIP/2.0 183 Session progress
  134. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK06c93736;rport
  135. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  136. To: <sip:18885551212@sip.voipcheap.com>;tag=780313ac4f44b87cc6a2c0
  137. Contact: sip:18885551212@77.72.169.134:5060
  138. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  139. CSeq: 103 INVITE
  140. Server: (Very nice Sip Registrar/Proxy Server)
  141. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  142. Content-Type: application/sdp
  143. Content-Length: 207
  144.  
  145. v=0
  146. o=myusername 1333499427 1333499427 IN IP4 77.72.168.31
  147. s=SIP Call
  148. c=IN IP4 77.72.168.31
  149. t=0 0
  150. m=audio 41560 RTP/AVP 3 101
  151. a=rtpmap:3 GSM/8000
  152. a=rtpmap:101 telephone-event/8000
  153. a=ptime:20
  154. <------------->
  155. --- (11 headers 9 lines) ---
  156. Found RTP audio format 3
  157. Found RTP audio format 101
  158. Found audio description format GSM for ID 3
  159. Found audio description format telephone-event for ID 101
  160. Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
  161. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  162. Peer audio RTP is at port 77.72.168.31:41560
  163. -- SIP/voipcheap-00000073 is making progress passing it to SIP/2928-00000072
  164. Scheduling destruction of SIP dialog '2a74834251eb313523034d2a7b931c04@194.90.1.5:5060' in 6400 ms (Method: INVITE)
  165. Reliably Transmitting (NAT) to 77.72.169.134:5060:
  166. CANCEL sip:18885551212@sip.voipcheap.com SIP/2.0
  167. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK06c93736;rport
  168. Max-Forwards: 70
  169. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  170. To: <sip:18885551212@sip.voipcheap.com>
  171. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  172. CSeq: 103 CANCEL
  173. User-Agent: Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  174. Content-Length: 0
  175.  
  176.  
  177. ---
  178. Scheduling destruction of SIP dialog '2a74834251eb313523034d2a7b931c04@194.90.1.5:5060' in 6400 ms (Method: INVITE)
  179. == Spawn extension (default, 18885551212, 2) exited non-zero on 'SIP/2928-00000072'
  180.  
  181. <--- SIP read from UDP:77.72.169.134:5060 --->
  182. SIP/2.0 200 Ok
  183. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK06c93736;rport
  184. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  185. To: <sip:18885551212@sip.voipcheap.com>
  186. Contact: sip:18885551212@77.72.169.134:5060
  187. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  188. CSeq: 103 CANCEL
  189. Server: (Very nice Sip Registrar/Proxy Server)
  190. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  191. Content-Length: 0
  192.  
  193. <------------->
  194. --- (10 headers 0 lines) ---
  195.  
  196. <--- SIP read from UDP:77.72.169.134:5060 --->
  197. SIP/2.0 487 Request terminated
  198. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK06c93736;rport
  199. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  200. To: <sip:18885551212@sip.voipcheap.com>
  201. Contact: sip:18885551212@77.72.169.134:5060
  202. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  203. CSeq: 103 INVITE
  204. Server: (Very nice Sip Registrar/Proxy Server)
  205. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  206. Content-Length: 0
  207.  
  208. <------------->
  209. --- (10 headers 0 lines) ---
  210. Transmitting (NAT) to 77.72.169.134:5060:
  211. ACK sip:18885551212@sip.voipcheap.com SIP/2.0
  212. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK06c93736;rport
  213. Max-Forwards: 70
  214. From: "2928" <sip:12124504543@194.90.1.5>;tag=as6a083b37
  215. To: <sip:18885551212@sip.voipcheap.com>
  216. Contact: <sip:12124504543@194.90.1.5:5060>
  217. Call-ID: 2a74834251eb313523034d2a7b931c04@194.90.1.5:5060
  218. CSeq: 103 ACK
  219. User-Agent: Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  220. Content-Length: 0
  221.  
  222.  
  223. ---
  224. Really destroying SIP dialog '2a74834251eb313523034d2a7b931c04@194.90.1.5:5060' Method: INVITE
  225. Reliably Transmitting (NAT) to 77.72.169.134:5060:
  226. OPTIONS sip:sip.voipcheap.com SIP/2.0
  227. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK37539b02;rport1G
  228. Max-Forwards: 70
  229. From: "asterisk" <sip:asterisk@194.90.1.5>;tag=as5811fabd
  230. To: <sip:sip.voipcheap.com>
  231. Contact: <sip:asterisk@194.90.1.5:5060>
  232. Call-ID: 75b513667c0a94f9093334cb71b73f06@194.90.1.5:5060
  233. CSeq: 102 OPTIONS
  234. User-Agent: Asterisk PBX 1.8.8.0~rc4-1digium0+1~squeeze
  235. Date: Wed, 04 Apr 2012 00:30:47 GMT
  236. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  237. Supported: replaces, timer
  238. Content-Length: 0
  239.  
  240.  
  241. ---
  242.  
  243. <--- SIP read from UDP:77.72.169.134:5060 --->
  244. SIP/2.0 200 Ok
  245. Via: SIP/2.0/UDP 194.90.1.5:5060;branch=z9hG4bK37539b02;rport
  246. From: "asterisk" <sip:asterisk@194.90.1.5>;tag=as5811fabd
  247. To: <sip:sip.voipcheap.com>
  248. Contact: sip:77.72.169.134:5060
  249. Call-ID: 75b513667c0a94f9093334cb71b73f06@194.90.1.5:5060
  250. CSeq: 102 OPTIONS
  251. Supported: foo
  252. User-Agent: (Very nice Sip Registrar/Proxy Server)
  253. Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
  254. Accept: application/sdp
  255.  
  256. <------------->
  257. --- (11 headers 0 lines) ---
  258. Really destroying SIP dialog '75b513667c0a94f9093334cb71b73f06@194.90.1.5:5060' Method: OPTIONS
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