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  1. cat pjsip_FreeVoipDeal.conf
  2. [fvd-trunk]
  3. type = registration
  4. ;transport = transport-udp-nat
  5. outbound_auth = fvd-auth
  6. server_uri = sip:sip.freevoipdeal.com
  7. client_uri = sip:sip.freevoipdeal.com
  8. retry_interval = 60
  9.  
  10. [fvd-auth]
  11. type = auth
  12. auth_type = userpass
  13. username = ********
  14. password = ********
  15.  
  16.  
  17. [FreeVoipDeal]
  18. type = endpoint
  19. aors = fvd-aor
  20. outbound_auth = fvd-auth
  21. transport = transport-udp-nat
  22. context = from-VOIP
  23. allow = !all,g722,ulaw
  24. direct_media = no
  25. from_domain = sip.freevoipdeal.com
  26.  
  27. [fvd-aor]
  28. type = aor
  29. contact = sip:sip.freevoipdeal.com
  30. qualify_frequency = 15
  31.  
  32. [fvd-identify]
  33. type = identify
  34. endpoint = FreeVoipDeal
  35. match = sip.freevoipdeal.com
  36.  
  37. //------------------------------------------------------------------------------------------
  38.  
  39. pjsip show registration fvd-trunk
  40.  
  41. <Registration/ServerURI..............................> <Auth..........> <Status.......>
  42. ==========================================================================================
  43.  
  44. fvd-trunk/sip:sip.freevoipdeal.com fvd-auth Registered
  45.  
  46. ParameterName : ParameterValue
  47. ===================================================
  48. auth_rejection_permanent : true
  49. client_uri : sip:sip.freevoipdeal.com
  50. contact_user :
  51. endpoint :
  52. expiration : 3600
  53. fatal_retry_interval : 0
  54. forbidden_retry_interval : 0
  55. line : false
  56. max_retries : 10
  57. outbound_auth : fvd-auth
  58. outbound_proxy :
  59. retry_interval : 60
  60. server_uri : sip:sip.freevoipdeal.com
  61. support_path : false
  62. transport :
  63.  
  64. -- Contact fvd-aor/sip:sip.freevoipdeal.com is now Reachable. RTT: 55.497 msec
  65. == Endpoint FreeVoipDeal is now Reachable
  66.  
  67. //-------------------------------------------------------------------------------------
  68. pjsip show endpoint FreeVoipDeal
  69.  
  70. Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
  71. I/OAuth: <AuthId/UserName...........................................................>
  72. Aor: <Aor............................................> <MaxContact>
  73. Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  74. Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
  75. Identify: <Identify/Endpoint.........................................................>
  76. Match: <ip/cidr.........................>
  77. Channel: <ChannelId......................................> <State.....> <Time.....>
  78. Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
  79. ==========================================================================================
  80.  
  81. Endpoint: FreeVoipDeal Not in use 0 of inf
  82. OutAuth: fvd-auth/****MY USER****l
  83. Aor: fvd-aor 0
  84. Contact: fvd-aor/sip:sip.freevoipdeal.com c86a228da8 Avail 55.717
  85. Transport: transport-udp-nat udp 0 0 0.0.0.0:5060
  86. Identify: fvd-identify/FreeVoipDeal
  87. Match: 77.72.174.128/32
  88.  
  89.  
  90. ParameterName : ParameterValue
  91. =========================================================
  92. 100rel : yes
  93. accountcode :
  94. acl :
  95. aggregate_mwi : true
  96. allow : (g722|ulaw)
  97. allow_subscribe : true
  98. allow_transfer : true
  99. aors : fvd-aor
  100. auth :
  101. bind_rtp_to_media_address : false
  102. call_group :
  103. callerid : <unknown>
  104. callerid_privacy : allowed_not_screened
  105. callerid_tag :
  106. connected_line_method : invite
  107. contact_acl :
  108. context : from-VOIP
  109. cos_audio : 0
  110. cos_video : 0
  111. device_state_busy_at : 0
  112. direct_media : false
  113. direct_media_glare_mitigation : none
  114. direct_media_method : invite
  115. disable_direct_media_on_nat : false
  116. dtls_ca_file :
  117. dtls_ca_path :
  118. dtls_cert_file :
  119. dtls_cipher :
  120. dtls_fingerprint : SHA-256
  121. dtls_private_key :
  122. dtls_rekey : 0
  123. dtls_setup : active
  124. dtls_verify : No
  125. dtmf_mode : rfc4733
  126. fax_detect : false
  127. fax_detect_timeout : 0
  128. force_avp : false
  129. force_rport : true
  130. from_domain : sip.freevoipdeal.com
  131. from_user :
  132. g726_non_standard : false
  133. ice_support : false
  134. identify_by : username
  135. inband_progress : false
  136. language :
  137. mailboxes :
  138. media_address :
  139. media_encryption : no
  140. media_encryption_optimistic : false
  141. media_use_received_transport : false
  142. message_context :
  143. moh_suggest : default
  144. mwi_from_user :
  145. mwi_subscribe_replaces_unsolicited : false
  146. named_call_group :
  147. named_pickup_group :
  148. one_touch_recording : false
  149. outbound_auth : fvd-auth
  150. outbound_proxy :
  151. pickup_group :
  152. record_off_feature : automixmon
  153. record_on_feature : automixmon
  154. rewrite_contact : false
  155. rpid_immediate : false
  156. rtp_engine : asterisk
  157. rtp_ipv6 : false
  158. rtp_keepalive : 0
  159. rtp_symmetric : false
  160. rtp_timeout : 0
  161. rtp_timeout_hold : 0
  162. sdp_owner : -
  163. sdp_session : Asterisk
  164. send_diversion : true
  165. send_pai : false
  166. send_rpid : false
  167. set_var :
  168. srtp_tag_32 : false
  169. sub_min_expiry : 0
  170. subscribe_context :
  171. t38_udptl : false
  172. t38_udptl_ec : none
  173. t38_udptl_ipv6 : false
  174. t38_udptl_maxdatagram : 0
  175. t38_udptl_nat : false
  176. timers : yes
  177. timers_min_se : 90
  178. timers_sess_expires : 1800
  179. tone_zone :
  180. tos_audio : 0
  181. tos_video : 0
  182. transport : transport-udp-nat
  183. trust_id_inbound : false
  184. trust_id_outbound : false
  185. use_avpf : false
  186. use_ptime : false
  187. user_eq_phone : false
  188. voicemail_extension :
  189.  
  190. //---------------------- CLI
  191. PBX*CLI>
  192. -- Executing [<MYNUMBER>@parsis:1] Dial("PJSIP/23-00000000", "pjsip/<DIALED_NUMBER>@FreeVoipDeal") in new stack
  193. -- Called pjsip/<MYNUMBER>@FreeVoipDeal
  194. == Everyone is busy/congested at this time (1:0/0/1)
  195. -- Auto fallthrough, channel 'PJSIP/23-00000000' status is 'CHANUNAVAIL'
  196.  
  197. //---------------------- PJSIP Debug ON
  198. PBX*CLI>
  199. <--- Received SIP request (1225 bytes) from UDP:192.168.200.170:5062 --->
  200. INVITE sip:<MYNUMBER>@192.168.200.104 SIP/2.0
  201. Via: SIP/2.0/UDP 192.168.200.170:5062;branch=z9hG4bK1489984851;rport
  202. From: "a" <sip:23@192.168.200.104>;tag=399299417
  203. To: <sip:<MYNUMBER>@192.168.200.104>
  204. Call-ID: 1783996928-5062-40@BJC.BGI.CAA.BHA
  205. CSeq: 320 INVITE
  206. Contact: "a" <sip:23@192.168.200.170:5062>
  207. X-Grandstream-PBX: true
  208. Max-Forwards: 70
  209. User-Agent: Grandstream GXP2140 1.0.7.25
  210. Privacy: none
  211. P-Preferred-Identity: "a" <sip:23@192.168.200.104>
  212. P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=44-87-FC-D0-B0-B3
  213. P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-7A-5A-11
  214. Supported: replaces, path, timer
  215. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  216. Content-Type: application/sdp
  217. Accept: application/sdp, application/dtmf-relay
  218. Content-Length: 406
  219.  
  220. v=0
  221. o=23 8001 8000 IN IP4 192.168.200.170
  222. s=SIP Call
  223. c=IN IP4 192.168.200.170
  224. t=0 0
  225. m=audio 5008 RTP/AVP 0 8 4 18 9 97 2 101
  226. a=sendrecv
  227. a=rtpmap:0 PCMU/8000
  228. a=ptime:20
  229. a=rtpmap:8 PCMA/8000
  230. a=rtpmap:4 G723/8000
  231. a=rtpmap:18 G729/8000
  232. a=fmtp:18 annexb=no
  233. a=rtpmap:9 G722/8000
  234. a=rtpmap:97 iLBC/8000
  235. a=fmtp:97 mode=30
  236. a=rtpmap:2 G726-32/8000
  237. a=rtpmap:101 telephone-event/8000
  238. a=fmtp:101 0-15
  239.  
  240. <--- Transmitting SIP response (497 bytes) to UDP:192.168.200.170:5062 --->
  241. SIP/2.0 401 Unauthorized
  242. Via: SIP/2.0/UDP 192.168.200.170:5062;rport=5062;received=192.168.200.170;branch=z9hG4bK1489984851
  243. Call-ID: 1783996928-5062-40@BJC.BGI.CAA.BHA
  244. From: "a" <sip:23@192.168.200.104>;tag=399299417
  245. To: <sip:<MYNUMBER>@192.168.200.104>;tag=z9hG4bK1489984851
  246. CSeq: 320 INVITE
  247. WWW-Authenticate: Digest realm="asterisk",nonce="1473235781/ee168b2d616c5bba787a1c6600fd64e1",opaque="5bb6fd153e4c47e8",algorithm=md5,qop="auth"
  248. Server: Asterisk PBX 13.11.0
  249. Content-Length: 0
  250.  
  251.  
  252. <--- Received SIP request (303 bytes) from UDP:192.168.200.170:5062 --->
  253. ACK sip:<MYNUMBER>@192.168.200.104 SIP/2.0
  254. Via: SIP/2.0/UDP 192.168.200.170:5062;branch=z9hG4bK1489984851;rport
  255. From: "a" <sip:23@192.168.200.104>;tag=399299417
  256. To: <sip:<MYNUMBER>@192.168.200.104>;tag=z9hG4bK1489984851
  257. Call-ID: 1783996928-5062-40@BJC.BGI.CAA.BHA
  258. CSeq: 320 ACK
  259. Content-Length: 0
  260.  
  261.  
  262. <--- Received SIP request (1502 bytes) from UDP:192.168.200.170:5062 --->
  263. INVITE sip:<MYNUMBER>@192.168.200.104 SIP/2.0
  264. Via: SIP/2.0/UDP 192.168.200.170:5062;branch=z9hG4bK1799936555;rport
  265. From: "a" <sip:23@192.168.200.104>;tag=399299417
  266. To: <sip:<MYNUMBER>@192.168.200.104>
  267. Call-ID: 1783996928-5062-40@BJC.BGI.CAA.BHA
  268. CSeq: 321 INVITE
  269. Contact: "a" <sip:23@192.168.200.170:5062>
  270. Authorization: Digest username="ruied", realm="asterisk", nonce="1473235781/ee168b2d616c5bba787a1c6600fd64e1", uri="sip:<MYNUMBER>@192.168.200.104", response="678180fe626d2b773b6f199ea2286092", algorithm=md5, cnonce="16572374", opaque="5bb6fd153e4c47e8", qop=auth, nc=00000004
  271. X-Grandstream-PBX: true
  272. Max-Forwards: 70
  273. User-Agent: Grandstream GXP2140 1.0.7.25
  274. Privacy: none
  275. P-Preferred-Identity: "a" <sip:23@192.168.200.104>
  276. P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=44-87-FC-D0-B0-B3
  277. P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-7A-5A-11
  278. Supported: replaces, path, timer
  279. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  280. Content-Type: application/sdp
  281. Accept: application/sdp, application/dtmf-relay
  282. Content-Length: 406
  283.  
  284. v=0
  285. o=23 8001 8000 IN IP4 192.168.200.170
  286. s=SIP Call
  287. c=IN IP4 192.168.200.170
  288. t=0 0
  289. m=audio 5008 RTP/AVP 0 8 4 18 9 97 2 101
  290. a=sendrecv
  291. a=rtpmap:0 PCMU/8000
  292. a=ptime:20
  293. a=rtpmap:8 PCMA/8000
  294. a=rtpmap:4 G723/8000
  295. a=rtpmap:18 G729/8000
  296. a=fmtp:18 annexb=no
  297. a=rtpmap:9 G722/8000
  298. a=rtpmap:97 iLBC/8000
  299. a=fmtp:97 mode=30
  300. a=rtpmap:2 G726-32/8000
  301. a=rtpmap:101 telephone-event/8000
  302. a=fmtp:101 0-15
  303.  
  304. <--- Transmitting SIP response (322 bytes) to UDP:192.168.200.170:5062 --->
  305. SIP/2.0 100 Trying
  306. Via: SIP/2.0/UDP 192.168.200.170:5062;rport=5062;received=192.168.200.170;branch=z9hG4bK1799936555
  307. Call-ID: 1783996928-5062-40@BJC.BGI.CAA.BHA
  308. From: "a" <sip:23@192.168.200.104>;tag=399299417
  309. To: <sip:<MYNUMBER>@192.168.200.104>
  310. CSeq: 321 INVITE
  311. Server: Asterisk PBX 13.11.0
  312. Content-Length: 0
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