cli log

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  1. [root@localhost ~]# asterisk -rx "sip show settings"
  2.  
  3.  
  4. Global Settings:
  5. ----------------
  6.   UDP Bindaddress:        0.0.0.0:5060
  7.   TCP SIP Bindaddress:    Disabled
  8.   TLS SIP Bindaddress:    Disabled
  9.   Videosupport:           Yes
  10.   Textsupport:            No
  11.   Ignore SDP sess. ver.:  No
  12.   AutoCreate Peer:        Off
  13.   Match Auth Username:    No
  14.   Allow unknown access:   Yes
  15.   Allow subscriptions:    Yes
  16.   Allow overlap dialing:  Yes
  17.   Allow promisc. redir:   No
  18.   Enable call counters:   No
  19.   SIP domain support:     No
  20.   Realm. auth:            No
  21.   Our auth realm          asterisk
  22.   Use domains as realms:  No
  23.   Call to non-local dom.: Yes
  24.   URI user is phone no:   No
  25.   Always auth rejects:    Yes
  26.   Direct RTP setup:       No
  27.   User Agent:             FPBX-13.0.192.16(11.23.1)
  28.   SDP Session Name:       Asterisk PBX 11.23.1
  29.   SDP Owner Name:         root
  30.   Reg. context:           (not set)
  31.   Regexten on Qualify:    No
  32.   Trust RPID:             No
  33.   Send RPID:              No
  34.   Legacy userfield parse: No
  35.   Send Diversion:         Yes
  36.   Caller ID:              Unknown
  37.   From: Domain:
  38.   Record SIP history:     Off
  39.   Call Events:            On
  40.   Auth. Failure Events:   Off
  41.   T.38 support:           No
  42.   T.38 EC mode:           Unknown
  43.   T.38 MaxDtgrm:          4294967295
  44.   SIP realtime:           Disabled
  45.   Qualify Freq :          60000 ms
  46.   Q.850 Reason header:    No
  47.   Store SIP_CAUSE:        No
  48.  
  49. Network QoS Settings:
  50. ---------------------------
  51.   IP ToS SIP:             CS3
  52.   IP ToS RTP audio:       EF
  53.   IP ToS RTP video:       AF41
  54.   IP ToS RTP text:        CS0
  55.   802.1p CoS SIP:         4
  56.   802.1p CoS RTP audio:   5
  57.   802.1p CoS RTP video:   6
  58.   802.1p CoS RTP text:    5
  59.   Jitterbuffer enabled:   No
  60.  
  61. Network Settings:
  62. ---------------------------
  63.   SIP address remapping:  Enabled using externhost
  64.   Externhost:             betvoip.wespace-i.com
  65.   Externaddr:             179.113.255.186:0
  66.   Externrefresh:          300
  67.   Localnet:               192.168.1.0/255.255.255.0
  68.  
  69. Global Signalling Settings:
  70. ---------------------------
  71.   Codecs:                 (gsm|ulaw|alaw|g726|g729|h264|mpeg4)
  72.   Codec Order:            alaw:20,ulaw:20,g729:20,gsm:20,g726:20,h264:0,mpeg4:0
  73.   Relax DTMF:             No
  74.   RFC2833 Compensation:   No
  75.   Symmetric RTP:          Yes
  76.   Compact SIP headers:    No
  77.   RTP Keepalive:          0 (Disabled)
  78.   RTP Timeout:            30
  79.   RTP Hold Timeout:       300
  80.   MWI NOTIFY mime type:   application/simple-message-summary
  81.   DNS SRV lookup:         No
  82.   Pedantic SIP support:   Yes
  83.   Reg. min duration       60 secs
  84.   Reg. max duration:      3600 secs
  85.   Reg. default duration:  120 secs
  86.   Sub. min duration       60 secs
  87.   Sub. max duration:      3600 secs
  88.   Outbound reg. timeout:  20 secs
  89.   Outbound reg. attempts: 0
  90.   Outbound reg. retry 403:0
  91.   Notify ringing state:   Yes
  92.     Include CID:          No
  93.   Notify hold state:      Yes
  94.   SIP Transfer mode:      open
  95.   Max Call Bitrate:       384 kbps
  96.   Auto-Framing:           No
  97.   Outb. proxy:            <not set>
  98.   Session Timers:         Accept
  99.   Session Refresher:      uas
  100.   Session Expires:        1800 secs
  101.   Session Min-SE:         90 secs
  102.   Timer T1:               500
  103.   Timer T1 minimum:       100
  104.   Timer B:                32000
  105.   No premature media:     Yes
  106.   Max forwards:           70
  107.  
  108. Default Settings:
  109. -----------------
  110.   Allowed transports:     UDP
  111.   Outbound transport:     UDP
  112.   Context:                from-sip-external
  113.   Record on feature:      automon
  114.   Record off feature:     automon
  115.   Force rport:            Yes
  116.   DTMF:                   rfc2833
  117.   Qualify:                0
  118.   Keepalive:              0
  119.   Use ClientCode:         No
  120.   Progress inband:        Never
  121.   Language:               en
  122.   Tone zone:              <Not set>
  123.   MOH Interpret:          default
  124.   MOH Suggest:
  125.   Voice Mail Extension:   *97
  126.  

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