[asterisk-dev] bug 0016466: cidname and cidnum in output of "sip show peers"

Klaus Darilion klaus.mailinglists at pernau.at
Thu Mar 18 16:01:00 CDT 2010



Am 18.03.2010 21:35, schrieb Russell Bryant:
> On 03/18/2010 03:31 PM, Ron Arts wrote:
>> For this reason I propose to add CallerID info to all instances of:
>>
>> -- SIP/netland44-00000402 is ringing
>> -- IAX2/iax-out-3-10507 answered DAHDI/5-1
>>
>> and similar log messages.
>
> Improving these very short messages to also include CallerID information
> seems reasonable to me.  However, that's not what the patch on the bug
> is proposing.  That patch adds more info to the "sip show peers" CLI
> command.
>
> I don't particularly mind that much.  I hesitate because we obviously
> can not add _all_ peer information to that command or it will become
> completely unreadable on just about everyone's terminal.  So, we need to
> be really strict about deciding what can be added there.

Also showing the full phone number (at least +15 digits) would be 
extremely useful:

 >sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)
111.22.33.90     +492115664  6518e46472b  00103/00000
11.22.222.184    +437202052  2f3856ca692  00101/00102
11.22.222.184    +435123122  0252c5627d3  00103/00000
111.22.33.90     +492115664  304e783c40a  00103/00104
11.22.222.184    +437202052  53507de543a  00101/00102
11.22.222.184    +436991588  2c5987dd67d  00102/00000
11.22.222.184    +436644527  0db010f9512  00101/00102
111.22.33.90     +492115664  5a29f7e336b  00103/00105
11.22.222.184    +437202052  3cd44206485  00101/00102
11.22.222.184    +431348014  04a50587327  00103/00000


also "show channels" should be not truncate channel names and contexts, 
this makes debugging very difficult:

 >core show channels
Channel              Location             State   Application(Data)
SIP/gatew1-09bab0f0  +431234620763 at fromPs Down    AppDial((Outgoing Line))
SIP/app-asterisk-b59 +431234620763 at toPstn Ring 
Dial(SIP/+431234620763 at gatew1)
SIP/gatew1-09ae50b0  (None)               Up      AppDial((Outgoing Line))
SIP/app-asterisk-09a +4912345642324 at toPst Up 
Dial(SIP/+4912345642324 at gatew1
SIP/app-asterisk-09f (None)               Up      AppDial((Outgoing Line))
SIP/gatew1-09f2f508  +43666612281 at fromPst Up 
Dial(SIP/+43666612281 at app-aste
SIP/gatew1-09af9bc8  (None)               Up      AppDial((Outgoing Line))


regards
klaus



More information about the asterisk-dev mailing list