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ASTERISK BACK TO BACK

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  1. ##########################################################################################
  2. 1.1.1.1 = The IP Handset initiating the call (Original ip was the WAN IP of the office )
  3. 192.168.10.1 = This IP the internal (LAN) ip of the IP handset
  4.  
  5. 77.77.77.66 = This is the IP of the A2B server that the handset is directly connnected to.
  6. 77.77.77.88 = This is the IP of the second A2B server that is supposed to bill the calls made by the first server.
  7.  
  8. A2B-SECOND-SERVER  = Trunk Name to the second server.
  9.  
  10. 441234567890 is the Tel number called
  11. ##########################################################################################
  12.  
  13.  
  14. LOG
  15.  
  16.  
  17. Asterisk 1.6.2.24, Copyright (C) 1999 - 2010 Digium, Inc. and others.
  18. Created by Mark Spencer <markster@digium.com>
  19. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  20. This is free software, with components licensed under the GNU General Public
  21. License version 2 and other licenses; you are welcome to redistribute it under
  22. certain conditions. Type 'core show license' for details.
  23. =========================================================================
  24.   == Parsing '/etc/asterisk/asterisk.conf':   == Found
  25.   == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf
  26.   == Found
  27. Connected to Asterisk 1.6.2.24 currently running on server066 (pid = 6807)
  28. server066*CLI>
  29. Verbosity is at least 3
  30. Core debug is at least 3
  31.  
  32.  
  33.  
  34. server066*CLI> server066*CLI> server066*CLI>
  35. 
  36. <--- SIP read from UDP:1.1.1.1:53450 --->
  37. INVITE sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1029059428From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>Call-ID: 1953639537@192.168.10.1CSeq: 1 INVITEContact: <sip:0024476691@192.168.10.1:5063>Content-Type: application/sdpAllow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGEMax-Forwards: 70User-Agent: Yealink SIP-T26P 6.61.0.83Supported: replacesAllow-Events: talk,hold,conference,refer,check-syncContent-Length: 305v=0o=- 20295 20295 IN IP4 192.168.10.1s=SDP datac=IN IP4 192.168.10.1t=0 0m=audio 4016 RTP/AVP 0 8 9 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:9 G722/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=fmtp:101 0-15a=rtpmap:101 telephone-event/8000a=ptime:20a=sendrecv
  38. <------------->
  39. --- (14 headers 15 lines) ---
  40.   == Using SIP RTP TOS bits 184
  41.   == Using SIP RTP CoS mark 5
  42.   == Using SIP VRTP TOS bits 136
  43.   == Using SIP VRTP CoS mark 6
  44. Sending to 192.168.10.1 : 5063 (no NAT)
  45. Using INVITE request as basis request - 1953639537@192.168.10.1
  46. Found peer '0024476691' for '0024476691' from 1.1.1.1:53450
  47.  
  48. <--- Reliably Transmitting (NAT) to 1.1.1.1:53450 --->
  49. SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1029059428;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as13a779f6Call-ID: 1953639537@192.168.10.1CSeq: 1 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7d912090"Content-Length: 0
  50. <------------>
  51. Scheduling destruction of SIP dialog '1953639537@192.168.10.1' in 32000 ms (Method: INVITE)
  52.  
  53. server066*CLI>
  54. 
  55. <--- SIP read from UDP:1.1.1.1:53450 --->
  56. ACK sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1029059428From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as13a779f6Call-ID: 1953639537@192.168.10.1CSeq: 1 ACKContent-Length: 0
  57. <------------->
  58. --- (7 headers 0 lines) ---
  59.  
  60. server066*CLI>
  61. 
  62. <--- SIP read from UDP:1.1.1.1:53450 --->
  63. INVITE sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEContact: <sip:0024476691@192.168.10.1:5063>Authorization: Digest username="0024476691", realm="asterisk", nonce="7d912090", uri="sip:441234567890@77.77.77.66", response="d068bcf2769ef7b3699fee63538dc300", algorithm=MD5Content-Type: application/sdpAllow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGEMax-Forwards: 70User-Agent: Yealink SIP-T26P 6.61.0.83Supported: replacesAllow-Events: talk,hold,conference,refer,check-syncContent-Length: 305v=0o=- 20295 20295 IN IP4 192.168.10.1s=SDP datac=IN IP4 192.168.10.1t=0 0m=audio 4016 RTP/AVP 0 8 9 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:9 G722/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=fmtp:101 0-15a=rtpmap:101 telephone-event/8000a=ptime:20a=sendrecv
  64. <------------->
  65. --- (15 headers 15 lines) ---
  66. Sending to 1.1.1.1 : 53450 (NAT)
  67. Using INVITE request as basis request - 1953639537@192.168.10.1
  68. Found peer '0024476691' for '0024476691' from 1.1.1.1:53450
  69. Found RTP audio format 0
  70. Found RTP audio format 8
  71. Found RTP audio format 9
  72. Found RTP audio format 18
  73. Found RTP audio format 101
  74. Found audio description format PCMU for ID 0
  75. Found audio description format PCMA for ID 8
  76. Found audio description format G722 for ID 9
  77. Found audio description format G729 for ID 18
  78. Found audio description format telephone-event for ID 101
  79. Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
  80. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  81. Peer audio RTP is at port 192.168.10.1:4016
  82. Looking for 441234567890 in a2billing (domain 77.77.77.66)
  83. list_route: hop: <sip:0024476691@192.168.10.1:5063>
  84.  
  85. <--- Transmitting (NAT) to 1.1.1.1:53450 --->
  86. SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.66>Content-Length: 0
  87. <------------>
  88.  
  89. server066*CLI>
  90.     -- Executing [441234567890@a2billing:1] AGI("SIP/0024476691-00000004", "a2billing.php,1") in new stack
  91.  
  92. server066*CLI>
  93.     -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
  94.  
  95. server066*CLI>
  96.     -- AGI Script Executing Application: (DIAL) Options: (SIP/A2B-SECOND-SERVER/441234567890,60,S(5088000:))
  97.  
  98. server066*CLI>
  99.     -- Setting call duration limit to 5088000.000 seconds.
  100.  
  101. server066*CLI>
  102.   == Using SIP RTP TOS bits 184
  103.   == Using SIP RTP CoS mark 5
  104.   == Using SIP VRTP TOS bits 136
  105.   == Using SIP VRTP CoS mark 6
  106.  
  107. server066*CLI>
  108. Audio is at 77.77.77.66 port 13810
  109. Adding codec 0x4 (ulaw) to SDP
  110. Adding codec 0x8 (alaw) to SDP
  111. Adding non-codec 0x1 (telephone-event) to SDP
  112. Reliably Transmitting (no NAT) to 77.77.77.88:5060:
  113. INVITE sip:441234567890@77.77.77.88 SIP/2.0Via: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;rportMax-Forwards: 70From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>Contact: <sip:441234567890@77.77.77.66>Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEUser-Agent: Asterisk PBX 1.6.2.24Date: Sun, 11 Nov 2012 16:24:41 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Type: application/sdpContent-Length: 261v=0o=root 1044909343 1044909343 IN IP4 77.77.77.66s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.66t=0 0m=audio 13810 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
  114. ---
  115.  
  116. server066*CLI>
  117.     -- Called A2B-SECOND-SERVER/441234567890
  118.  
  119. server066*CLI>
  120. 
  121. <--- SIP read from UDP:77.77.77.88:5060 --->
  122. SIP/2.0 100 TryingVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;received=77.77.77.66;rport=5060From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.88>Content-Length: 0
  123. <------------->
  124. --- (11 headers 0 lines) ---
  125.  
  126. server066*CLI>
  127. 
  128. <--- SIP read from UDP:77.77.77.88:5060 --->
  129. SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;received=77.77.77.66;rport=5060From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>;tag=as0e885c15Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.88>Content-Type: application/sdpContent-Length: 261v=0o=root 2063983537 2063983537 IN IP4 77.77.77.88s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.88t=0 0m=audio 10016 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
  130. <------------->
  131. --- (12 headers 12 lines) ---
  132. Found RTP audio format 0
  133. Found RTP audio format 8
  134. Found RTP audio format 101
  135. Found audio description format PCMU for ID 0
  136. Found audio description format PCMA for ID 8
  137. Found audio description format telephone-event for ID 101
  138. Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  139. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  140. Peer audio RTP is at port 77.77.77.88:10016
  141.     -- SIP/A2B-SECOND-SERVER-00000005 is making progress passing it to SIP/0024476691-00000004
  142. Audio is at 77.77.77.66 port 13634
  143. Adding codec 0x4 (ulaw) to SDP
  144. Adding codec 0x8 (alaw) to SDP
  145. Adding codec 0x100 (g729) to SDP
  146. Adding non-codec 0x1 (telephone-event) to SDP
  147.  
  148. <--- Transmitting (NAT) to 1.1.1.1:53450 --->
  149. SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as2decf668Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.66>Content-Type: application/sdpContent-Length: 302v=0o=root 9081837 9081837 IN IP4 77.77.77.66s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.66t=0 0m=audio 13634 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
  150. <------------>
  151.  
  152. server066*CLI>
  153. 
  154. <--- SIP read from UDP:77.77.77.88:5060 --->
  155. SIP/2.0 200 OKVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK4b0bed95;received=77.77.77.66;rport=5060From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>;tag=as0e885c15Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.88>Content-Type: application/sdpContent-Length: 261v=0o=root 2063983537 2063983538 IN IP4 77.77.77.88s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.88t=0 0m=audio 10016 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
  156. <------------->
  157. --- (12 headers 12 lines) ---
  158. Found RTP audio format 0
  159. Found RTP audio format 8
  160. Found RTP audio format 101
  161. Found audio description format PCMU for ID 0
  162. Found audio description format PCMA for ID 8
  163. Found audio description format telephone-event for ID 101
  164. Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  165. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  166. Peer audio RTP is at port 77.77.77.88:10016
  167. list_route: hop: <sip:441234567890@77.77.77.88>
  168. set_destination: Parsing <sip:441234567890@77.77.77.88> for address/port to send to
  169. set_destination: set destination to 77.77.77.88, port 5060
  170. Transmitting (no NAT) to 77.77.77.88:5060:
  171. ACK sip:441234567890@77.77.77.88 SIP/2.0Via: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK3baec434;rportMax-Forwards: 70From: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eTo: <sip:441234567890@77.77.77.88>;tag=as0e885c15Contact: <sip:441234567890@77.77.77.66>Call-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 ACKUser-Agent: Asterisk PBX 1.6.2.24Content-Length: 0
  172. ---
  173.     -- SIP/A2B-SECOND-SERVER-00000005 answered SIP/0024476691-00000004
  174. Audio is at 77.77.77.66 port 13634
  175. Adding codec 0x4 (ulaw) to SDP
  176. Adding codec 0x8 (alaw) to SDP
  177. Adding codec 0x100 (g729) to SDP
  178. Adding non-codec 0x1 (telephone-event) to SDP
  179.  
  180. <--- Reliably Transmitting (NAT) to 1.1.1.1:53450 --->
  181. SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK1836708136;received=1.1.1.1From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859
  182. server066*CLI>
  183. To: <sip:441234567890@77.77.77.66>;tag=as2decf668Call-ID: 1953639537@192.168.10.1CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContact: <sip:441234567890@77.77.77.66>Content-Type: application/sdpContent-Length: 302v=0o=root 9081837 9081838 IN IP4 77.77.77.66s=Asterisk PBX 1.6.2.24c=IN IP4 77.77.77.66t=0 0m=audio 13634 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecv
  184. <------------>
  185.     -- Packet2Packet bridging SIP/0024476691-00000004 and SIP/A2B-SECOND-SERVER-00000005
  186.  
  187. server066*CLI>
  188. 
  189. <--- SIP read from UDP:77.77.77.88:5060 --->
  190. BYE sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 77.77.77.88:5060;branch=z9hG4bK38db1bd6;rportMax-Forwards: 70From: <sip:441234567890@77.77.77.88>;tag=as0e885c15To: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eCall-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 BYEUser-Agent: Asterisk PBX 1.6.2.24X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
  191. <------------->
  192. --- (11 headers 0 lines) ---
  193. Sending to 77.77.77.88 : 5060 (no NAT)
  194.  
  195. <--- Transmitting (no NAT) to 77.77.77.88:5060 --->
  196. SIP/2.0 200 OKVia: SIP/2.0/UDP 77.77.77.88:5060;branch=z9hG4bK38db1bd6;received=77.77.77.88;rport=5060From: <sip:441234567890@77.77.77.88>;tag=as0e885c15To: "441234567890" <sip:441234567890@77.77.77.66>;tag=as3fd73a0eCall-ID: 47bb4c505b70e695224506336ef57238@77.77.77.66CSeq: 102 BYEServer: Asterisk PBX 1.6.2.24Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0
  197. <------------>
  198.  
  199. server066*CLI>
  200.     -- <SIP/0024476691-00000004>AGI Script a2billing.php completed, returning 4
  201.  
  202. server066*CLI>
  203.   == Spawn extension (a2billing, 441234567890, 1) exited non-zero on 'SIP/0024476691-00000004'
  204.  
  205. server066*CLI>
  206. Scheduling destruction of SIP dialog '1953639537@192.168.10.1' in 32000 ms (Method: INVITE)
  207.  
  208. server066*CLI>
  209. 
  210. <--- SIP read from UDP:1.1.1.1:53450 --->
  211. ACK sip:441234567890@77.77.77.66 SIP/2.0Via: SIP/2.0/UDP 192.168.10.1:5063;branch=z9hG4bK553523559From: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859To: <sip:441234567890@77.77.77.66>;tag=as2decf668Call-ID: 1953639537@192.168.10.1CSeq: 2 ACKContact: <sip:0024476691@192.168.10.1:5063>Max-Forwards: 70User-Agent: Yealink SIP-T26P 6.61.0.83Content-Length: 0
  212. <------------->
  213. --- (10 headers 0 lines) ---
  214.  
  215. server066*CLI>
  216. set_destination: Parsing <sip:0024476691@192.168.10.1:5063> for address/port to send to
  217. set_destination: set destination to 192.168.10.1, port 5063
  218. Reliably Transmitting (NAT) to 1.1.1.1:53450:
  219. BYE sip:0024476691@192.168.10.1:5063 SIP/2.0Via: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK427777eb;rportMax-Forwards: 70From: <sip:441234567890@77.77.77.66>;tag=as2decf668To: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859Call-ID: 1953639537@192.168.10.1CSeq: 102 BYEUser-Agent: Asterisk PBX 1.6.2.24X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
  220. ---
  221. Scheduling destruction of SIP dialog '1953639537@192.168.10.1' in 32000 ms (Method: ACK)
  222. Really destroying SIP dialog '47bb4c505b70e695224506336ef57238@77.77.77.66' Method: BYE
  223.  
  224. server066*CLI>
  225. 
  226. <--- SIP read from UDP:1.1.1.1:53450 --->
  227. SIP/2.0 200 OKVia: SIP/2.0/UDP 77.77.77.66:5060;branch=z9hG4bK427777eb;rportFrom: <sip:441234567890@77.77.77.66>;tag=as2decf668To: "TEST 066" <sip:0024476691@77.77.77.66>;tag=592961859Call-ID: 1953639537@192.168.10.1CSeq: 102 BYEUser-Agent: Yealink SIP-T26P 6.61.0.83Content-Length: 0
  228. <------------->
  229. --- (8 headers 0 lines) ---
  230. SIP Response message for INCOMING dialog BYE arrived
  231. Really destroying SIP dialog '1953639537@192.168.10.1' Method: ACK
  232.  
  233. server066*CLI> exitExecuting last minute cleanups
  234.  
  235. ============(CALL DROPS HERE)=============
  236.  
  237.  
  238. Asterisk ending (0).
  239. ]0;root@server066:/etc/asterisk[root@server066 asterisk]#
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