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  1. ### [Asterisk debug messages ] ############################################################
  2.  
  3. <--- SIP read from UDP:217.10.79.9:5060 --->
  4.  
  5. <------------->
  6. redox*CLI>
  7. <--- SIP read from UDP:217.10.79.9:5060 --->
  8. INVITE sip:30@192.168.0.3 SIP/2.0
  9. Record-Route: <sip:217.10.79.9;lr=on;ftag=as76f4a6ba>
  10. Record-Route: <sip:172.20.40.3;lr=on>
  11. Record-Route: <sip:217.10.79.9;lr=on;ftag=as76f4a6ba>
  12. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8de5.58857c54.0
  13. Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8de5.58857c54.0
  14. Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK543a2aa3
  15. Via: SIP/2.0/UDP 217.10.67.4:5060;branch=z9hG4bK543a2aa3;rport=5060
  16. From: "0160xxxxxxxx" <sip:0160xxxxxxxx@sipgate.de>;tag=as76f4a6ba
  17. To: <sip:0049381xxxxxxx@sipgate.de>
  18. Contact: <sip:0160xxxxxxxx@217.10.67.4>
  19. Call-ID: 54ed359c6887a20414d10dcc7dff90f2@sipgate.de
  20. CSeq: 102 INVITE
  21. Max-Forwards: 67
  22. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  23. Supported: replaces
  24. Content-Type: application/sdp
  25. Content-Length: 402
  26.  
  27. v=0
  28. o=root 14171 14171 IN IP4 217.10.67.4
  29. s=session
  30. c=IN IP4 217.10.77.21
  31. t=0 0
  32. m=audio 58052 RTP/AVP 8 0 3 18 112 101
  33. a=rtpmap:8 PCMA/8000
  34. a=rtpmap:0 PCMU/8000
  35. a=rtpmap:3 GSM/8000
  36. a=rtpmap:18 G729/8000
  37. a=fmtp:18 annexb=no
  38. a=rtpmap:112 G726-32/8000
  39. a=rtpmap:101 telephone-event/8000
  40. a=fmtp:101 0-16
  41. a=silenceSupp:off - - - -
  42. a=ptime:20
  43. a=sendrecv
  44. a=direction:active
  45. a=nortpproxy:yes
  46.  
  47. <------------->
  48. --- (18 headers 19 lines) ---
  49. Sending to 217.10.79.9 : 5060 (NAT)
  50. Using INVITE request as basis request - 54ed359c6887a20414d10dcc7dff90f2@sipgate.de
  51. Found peer 'outgoing' for '0160xxxxxxxx' from 217.10.79.9:5060
  52. Found RTP audio format 8
  53. Found RTP audio format 0
  54. Found RTP audio format 3
  55. Found RTP audio format 18
  56. Found RTP audio format 112
  57. Found RTP audio format 101
  58. Found audio description format PCMA for ID 8
  59. Found audio description format PCMU for ID 0
  60. Found audio description format GSM for ID 3
  61. Found audio description format G729 for ID 18
  62. Found audio description format G726-32 for ID 112
  63. Found audio description format telephone-event for ID 101
  64. Capabilities: us - 0x14e (gsm|ulaw|alaw|slin|g729), peer - audio=0x90e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
  65. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  66. Peer audio RTP is at port 217.10.77.21:58052
  67. Looking for 30 in default (domain 192.168.0.3)
  68. list_route: hop: <sip:217.10.79.9;lr=on;ftag=as76f4a6ba>
  69. list_route: hop: <sip:172.20.40.3;lr=on>
  70. list_route: hop: <sip:217.10.79.9;lr=on;ftag=as76f4a6ba>
  71. redox*CLI>
  72. <--- Transmitting (NAT) to 217.10.79.9:5060 --->
  73. SIP/2.0 100 Trying
  74. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8de5.58857c54.0;received=217.10.79.9
  75. Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8de5.58857c54.0
  76. Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK543a2aa3
  77. Via: SIP/2.0/UDP 217.10.67.4:5060;branch=z9hG4bK543a2aa3;rport=5060
  78. Record-Route: <sip:217.10.79.9;lr=on;ftag=as76f4a6ba>
  79. Record-Route: <sip:172.20.40.3;lr=on>
  80. Record-Route: <sip:217.10.79.9;lr=on;ftag=as76f4a6ba>
  81. From: "016095346819" <sip:0160xxxxxxxx@sipgate.de>;tag=as76f4a6ba
  82. To: <sip:0049381xxxxxxx@sipgate.de>
  83. Call-ID: 54ed359c6887a20414d10dcc7dff90f2@sipgate.de
  84. CSeq: 102 INVITE
  85. Server: Asterisk PBX 1.6.2.5
  86. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  87. Supported: replaces, timer
  88. Contact: <sip:30@192.168.0.3>
  89. Content-Length: 0
  90.  
  91.  
  92. <------------>
  93. [Apr 8 15:51:48] WARNING[13966]: pbx.c:3675 pbx_extension_helper: No application '' for extension (default, 30, 2)
  94. Scheduling destruction of SIP dialog '54ed359c6887a20414d10dcc7dff90f2@sipgate.de' in 6400 ms (Method: INVITE)
  95.  
  96. <--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
  97. SIP/2.0 603 Declined
  98. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8de5.58857c54.0;received=217.10.79.9
  99. Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8de5.58857c54.0
  100. Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK543a2aa3
  101. Via: SIP/2.0/UDP 217.10.67.4:5060;branch=z9hG4bK543a2aa3;rport=5060
  102. From: "0160xxxxxxxx" <sip:0160xxxxxxxx@sipgate.de>;tag=as76f4a6ba
  103. To: <sip:0049381xxxxxxx@sipgate.de>;tag=as2ef9a557
  104. Call-ID: 54ed359c6887a20414d10dcc7dff90f2@sipgate.de
  105. CSeq: 102 INVITE
  106. Server: Asterisk PBX 1.6.2.5
  107. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  108. Supported: replaces, timer
  109. Content-Length: 0
  110.  
  111.  
  112. <------------>
  113. redox*CLI>
  114. <--- SIP read from UDP:217.10.79.9:5060 --->
  115. ACK sip:30@192.168.0.3 SIP/2.0
  116. Max-Forwards: 10
  117. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8de5.58857c54.0
  118. Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8de5.58857c54.0
  119. From: "0160xxxxxxxx" <sip:0160xxxxxxxx@sipgate.de>;tag=as76f4a6ba
  120. Call-ID: 54ed359c6887a20414d10dcc7dff90f2@sipgate.de
  121. To: <sip:0049381xxxxxxx@sipgate.de>;tag=as2ef9a557
  122. CSeq: 102 ACK
  123. Content-Length: 0
  124. X-hint: rr-enforced
  125.  
  126.  
  127. <------------->
  128. --- (10 headers 0 lines) ---
  129. redox*CLI>
  130. <--- SIP read from UDP:217.10.79.9:5060 --->
  131.  
  132. <------------->
  133.  
  134.  
  135.  
  136. ### [sip.conf] ############################################################################
  137.  
  138. [general]
  139. port=5060
  140. bindaddr=0.0.0.0
  141. localnet=192.168.0.0/24
  142. qualify=yes
  143. disallow=all
  144. allow=alaw
  145. allow=ulaw
  146. allow=g729
  147. allow=gsm
  148. allow=slinear
  149. context=default
  150. srvlookup=yes
  151. language=de
  152. nat=yes
  153. match_auth_username=yes
  154. register => callid:passwd@sipgate.de/30
  155.  
  156. [outgoing]
  157. type=friend
  158. username=24xxxxx
  159. fromuser=24xxxxx
  160. secret=passwd
  161. host=sipgate.de
  162. fromdomain=sipgate.de
  163. insecure=invite
  164. context=default
  165. insecure=port,invite
  166. canreinvite=no
  167. dtmfmode=rfc2833
  168.  
  169. [incoming]
  170. type=friend
  171. host=dynamic
  172. context=default
  173.  
  174. [30]
  175. type=friend
  176. username=24xxxxx
  177. fromuser=24xxxxx
  178. fromdomain=sipgate.de
  179. user=30
  180. mailbox=30
  181. host=dynamic
  182. secret=passwd
  183. insecure=very
  184. context=default
  185. callerid="Bjoern <30>"
  186. canreinvite=no
  187. qualify=yes
  188. allow=ulaw
  189. nat=yes
  190.  
  191.  
  192.  
  193. ### [extensions.conf] #####################################################################
  194.  
  195. [general]
  196. static=yes
  197. writeprotect=no
  198.  
  199. ; --------------------------------------------------------------------
  200. ; Es hat sich als gute Praxis erwiesen, die Inhalte der Datei
  201. ; extensions.conf modular aufzubauen. Diese Praxis wollen
  202. ; wir auch hier anwenden
  203. ;
  204.  
  205. [echotest]
  206. exten => 81,1,Answer
  207. exten => 81,n,Wait(1)
  208. exten => 81,n,Playback(demo-echotest)
  209. exten => 81,n,Echo
  210. exten => 81,n,Playback(demo-echodone)
  211. exten => 81,n,Hangup
  212.  
  213. [30]
  214. ;exten => 30,1,Dial(SIP/30,15)
  215. exten => 30,2 Voicemail(30)
  216. exten => 30,3,Hangup
  217.  
  218. [mailbox]
  219. exten => Mailbox,1,Answer
  220. exten => Mailbox,n,Wait(1)
  221. exten => Mailbox,n,VoiceMailMain(30@default,s)
  222. exten => Mailbox,2,Hangup
  223.  
  224. [apfelmus]
  225. exten => 8888,1,Answer()
  226. exten => 8888,2,Playback(hello-world)
  227. exten => 8888,3,Hangup()
  228.  
  229. [home]
  230. ; Erreichbarkeit der Nebenstellen 30-39
  231. ; untereinander herstellen
  232. exten => _3X,1,NoCDR()
  233. exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr
  234.  
  235. [outgoing]
  236. exten => _0.,1,Set(CALLERID(num)=24xxxxx)
  237. exten => _0.,n,Dial(SIP/${EXTEN}@outgoing,60,trg)
  238. exten => _0.,n,Hangup
  239.  
  240. [incoming]
  241. ; alle Anrufe mit einer ID 2473867 sollen an das SIP Endgeraet 30 geroutet werden
  242. exten => 30,1,Dial(SIP/30,30)
  243. exten => 30,2,VoiceMail(30,u)
  244.  
  245.  
  246. [default]
  247. include => echotest
  248. include => mailbox
  249. include => 30
  250. include => home
  251. include => outgoing
  252. include => incoming
  253. include => apfelmus
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