[asterisk-dev] RFC: pjsip show endpoints output format

George Joseph george.joseph at fairview5.com
Mon Nov 4 14:44:00 CST 2013


Comments please...

I tried to balance readability, usefullness, and the fact that the output
is a flat representation of a relational model.

Here's a pastebin if you can't read the output below.
http://pastebin.com/JiPKLY3A



ernie*CLI> pjsip show endpoints
Endpoints:
 Endpoint/CID--------------------------  State-----  Channels
   I/OAuth: AuthId/UserName------------
   Contact: Uri--------------------------------------------  Status---
 ---RTT(ms)
   Channel: ChannelId--------------------------------------  State----
 -Time(sec)
      Codec: Codec-- Exten: DialedExten---  ConnectedLine:
ConnectedLineCID------
==================================================================================
 1181/7205551212                         Not in use  0 of 5
    InAuth: 1181/1181
   Contact: sip:1181 at 192.168.147.181:5062                    Avail
 30.935

 1180/7205551212                         Invalid     0 of 5
    InAuth: 1180/1180

 voipms-backup                           Invalid     0 of 100
   OutAuth: voipms/999999_gw1sip1
   Contact: sip:dallas.voip.ms:5060                          Avail
 42.483
   Contact: sip:houston.voip.ms:5060                         Avail
 48.748

 1185                                    In use      1 of 5
    InAuth: 1185/1185
   Contact: sip:1185 at 192.168.147.49:39554                    Avail
 66.196
   Contact: sip:1185 at 192.168.147.49:41450;transport=udp      Unavail
  0.000
   Channel: PJSIP/1185-00000004/Dial                         Up
     9
      Codec: (ulaw)  Exten: 3035551212    ConnectedLine: "" <>

 voipms-primary                          In use      1 of 100
   OutAuth: voipms/999999_gw1sip1
   Contact: sip:denver1.voip.ms:5060                         Avail
 22.423
   Contact: sip:denver2.voip.ms:5060                         Avail
 22.007
   Channel: PJSIP/voipms-primary-00000005/AppDial            Up
     9
      Codec: (ulaw)  Exten:               ConnectedLine: "George Joseph"
<1185>

 1186                                    Not in use  0 of inf
    InAuth: 1186/1186
   Contact: sip:1186 at 192.168.147.61:48933;transport=udp      Avail
189.095
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131104/d91d3b81/attachment.html>


More information about the asterisk-dev mailing list