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  1. asterisk*CLI> sip show channels
  2. Peer User/ANR Call ID Format Hold Last Message Expiry Peer
  3. 198.8.60.11 (None) 5f7ddee93e09546 (nothing) No Rx: OPTIONS <guest>
  4. 1 active SIP dialog
  5.  
  6. asterisk*CLI> sip show channelstats
  7. Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
  8. 198.8.60.11 714afed0663 (inv state: None) -- -- No RTP active
  9. 0 active SIP channels
  10.  
  11. asterisk*CLI> sip show registry
  12. Host dnsmgr Username Refresh State Reg.Time
  13. sanfrancisco-1.vtnoc.net:5060 N 19254291724 105 Registered Wed, 22 Oct 2014 11:17:09
  14. 1 SIP registrations.
  15.  
  16. asterisk*CLI> sip show objects
  17. -= Peer objects: 2 static, 0 realtime, 1 autocreate =-
  18.  
  19. name: 6001
  20. type: peer
  21. objflags: 0
  22. refcount: 1
  23.  
  24. name: VIATALK
  25. type: peer
  26. objflags: 0
  27. refcount: 3
  28.  
  29. -= Peer objects by IP =-
  30.  
  31. name: VIATALK
  32. type: peer
  33. objflags: 0
  34. refcount: 3
  35.  
  36. -= Registry objects: 1 =-
  37.  
  38. name: 19254291724:lxev1nvno9@sanfrancisco-1.vtnoc.net/19254291724
  39. objflags: 0
  40. refcount: 2
  41.  
  42. -= Dialog objects:
  43.  
  44. name: 0effa88d6564433c2902c6296da24738@198.8.60.11:5060
  45. type: dialog
  46. objflags: 0
  47. refcount: 2
  48.  
  49. asterisk*CLI> sip show inuse
  50. * Peer name In use Limit
  51. 6001 0/0/0 2147483647
  52.  
  53. asterisk*CLI> sip show peer VIATALK
  54.  
  55.  
  56. * Name : VIATALK
  57. Description :
  58. Secret : <Set>
  59. MD5Secret : <Not set>
  60. Remote Secret: <Not set>
  61. Context : from-pstn
  62. Record On feature : automon
  63. Record Off feature : automon
  64. Subscr.Cont. : <Not set>
  65. Language :
  66. Tonezone : <Not set>
  67. AMA flags : Unknown
  68. Transfer mode: open
  69. CallingPres : Presentation Allowed, Not Screened
  70. FromUser : 19254291724
  71. FromDomain : sanfrancisco-1.vtnoc.net Port 5060
  72. Callgroup :
  73. Pickupgroup :
  74. Named Callgr :
  75. Nam. Pickupgr:
  76. MOH Suggest :
  77. Mailbox :
  78. VM Extension : *97
  79. LastMsgsSent : 0/0
  80. Call limit : 0
  81. Max forwards : 0
  82. Dynamic : No
  83. Callerid : "" <>
  84. MaxCallBR : 384 kbps
  85. Expire : -1
  86. Insecure : port,invite
  87. Force rport : No
  88. Symmetric RTP: No
  89. ACL : No
  90. DirectMedACL : No
  91. T.38 support : No
  92. T.38 EC mode : Unknown
  93. T.38 MaxDtgrm: -1
  94. DirectMedia : No
  95. PromiscRedir : No
  96. User=Phone : No
  97. Video Support: Yes
  98. Text Support : No
  99. Ign SDP ver : No
  100. Trust RPID : No
  101. Send RPID : No
  102. TrustIDOutbnd: Legacy
  103. Subscriptions: Yes
  104. Overlap dial : Yes
  105. DTMFmode : inband
  106. Timer T1 : 500
  107. Timer B : 32000
  108. ToHost : sanfrancisco-1.vtnoc.net
  109. Addr->IP : 198.8.60.11:5060
  110. Defaddr->IP : (null)
  111. Prim.Transp. : UDP
  112. Allowed.Trsp : UDP
  113. Def. Username: 19254291724
  114. SIP Options : (none)
  115. Codecs : (ulaw)
  116. Codec Order : (ulaw:20)
  117. Auto-Framing : No
  118. Status : OK (61 ms)
  119. Useragent :
  120. Reg. Contact :
  121. Qualify Freq : 60000 ms
  122. Keepalive : 0 ms
  123. Sess-Timers : Accept
  124. Sess-Refresh : uas
  125. Sess-Expires : 1800 secs
  126. Min-Sess : 90 secs
  127. RTP Engine : asterisk
  128. Parkinglot :
  129. Use Reason : No
  130. Encryption : No
  131.  
  132. asterisk*CLI> sip show settings
  133.  
  134.  
  135. Global Settings:
  136. ----------------
  137. UDP Bindaddress: 0.0.0.0:5060
  138. TCP SIP Bindaddress: Disabled
  139. TLS SIP Bindaddress: Disabled
  140. Videosupport: Yes
  141. Textsupport: No
  142. Ignore SDP sess. ver.: No
  143. AutoCreate Peer: Off
  144. Match Auth Username: No
  145. Allow unknown access: Yes
  146. Allow subscriptions: Yes
  147. Allow overlap dialing: Yes
  148. Allow promisc. redir: No
  149. Enable call counters: No
  150. SIP domain support: No
  151. Realm. auth: No
  152. Our auth realm asterisk
  153. Use domains as realms: No
  154. Call to non-local dom.: Yes
  155. URI user is phone no: No
  156. Always auth rejects: Yes
  157. Direct RTP setup: No
  158. User Agent: FPBX-2.11.0(11.9.0)
  159. SDP Session Name: Asterisk PBX 11.9.0
  160. SDP Owner Name: root
  161. Reg. context: (not set)
  162. Regexten on Qualify: No
  163. Trust RPID: No
  164. Send RPID: No
  165. Legacy userfield parse: No
  166. Send Diversion: Yes
  167. Caller ID: Unknown
  168. From: Domain:
  169. Record SIP history: On
  170. Call Events: Off
  171. Auth. Failure Events: Off
  172. T.38 support: No
  173. T.38 EC mode: Unknown
  174. T.38 MaxDtgrm: -1
  175. SIP realtime: Disabled
  176. Qualify Freq : 60000 ms
  177. Q.850 Reason header: No
  178. Store SIP_CAUSE: No
  179.  
  180. Network QoS Settings:
  181. ---------------------------
  182. IP ToS SIP: CS3
  183. IP ToS RTP audio: EF
  184. IP ToS RTP video: AF41
  185. IP ToS RTP text: CS0
  186. 802.1p CoS SIP: 4
  187. 802.1p CoS RTP audio: 5
  188. 802.1p CoS RTP video: 6
  189. 802.1p CoS RTP text: 5
  190. Jitterbuffer enabled: No
  191.  
  192. Network Settings:
  193. ---------------------------
  194. SIP address remapping: Enabled using externaddr
  195. Externhost: <none>
  196. Externaddr: 50.79.211.169:0
  197. Externrefresh: 10
  198. Localnet: 192.168.1.0/255.255.255.0
  199.  
  200. Global Signalling Settings:
  201. ---------------------------
  202. Codecs: (gsm|ulaw|alaw|h264)
  203. Codec Order: ulaw:20,alaw:20,gsm:20,h264:0
  204. Relax DTMF: No
  205. RFC2833 Compensation: No
  206. Symmetric RTP: Yes
  207. Compact SIP headers: No
  208. RTP Keepalive: 0 (Disabled)
  209. RTP Timeout: 30
  210. RTP Hold Timeout: 300
  211. MWI NOTIFY mime type: application/simple-message-summary
  212. DNS SRV lookup: No
  213. Pedantic SIP support: Yes
  214. Reg. min duration 60 secs
  215. Reg. max duration: 3600 secs
  216. Reg. default duration: 120 secs
  217. Sub. min duration 60 secs
  218. Sub. max duration: 3600 secs
  219. Outbound reg. timeout: 20 secs
  220. Outbound reg. attempts: 0
  221. Outbound reg. retry 403:0
  222. Notify ringing state: Yes
  223. Include CID: No
  224. Notify hold state: Yes
  225. SIP Transfer mode: open
  226. Max Call Bitrate: 384 kbps
  227. Auto-Framing: No
  228. Outb. proxy: <not set>
  229. Session Timers: Accept
  230. Session Refresher: uas
  231. Session Expires: 1800 secs
  232. Session Min-SE: 90 secs
  233. Timer T1: 500
  234. Timer T1 minimum: 100
  235. Timer B: 32000
  236. No premature media: Yes
  237. Max forwards: 70
  238.  
  239. Default Settings:
  240. -----------------
  241. Allowed transports: UDP
  242. Outbound transport: UDP
  243. Context: from-sip-external
  244. Record on feature: automon
  245. Record off feature: automon
  246. Force rport: Yes
  247. DTMF: rfc2833
  248. Qualify: 0
  249. Keepalive: 0
  250. Use ClientCode: No
  251. Progress inband: Never
  252. Language:
  253. Tone zone: <Not set>
  254. MOH Interpret: default
  255. MOH Suggest:
  256. Voice Mail Extension: *97
  257.  
  258. ----
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