call drops from application MX ONE

[teenu3]

Hi all,

here is a senario .

A. A PABX WITH SIP TELEPHONES
B. AN MX ONE PABX
C. PSTN

NOW BETWEEN A AND B THERE 2 E1 60 channel connection is there.
between b and c one E1 TO PSTN.

NOW 'A' CALLING TO 'C' FROM TELEPHONE . NO PROBLEM NO CALL DROPS.

THE PROBLEM IS 'A' HAVE A NEW APPLICATION WHICH IS CALLING 'C' AUTOMATICALY. THOSE CALLS DROPS SOME TIME.
IF 10 CALLS IS THERE 3 CALLS DROPS.
'A' IS THINKING THE DROP IS AT 'B' .

MY POINT IS THAT WHY IT IS NOT CUTTING WHILE CALLING FROM TELEPHONE.

HOW CAN I TRACE IT TRAICE IT.
WITH WHAT COMMAND AND WHERE. I HAVE ONLY ACCESS TO THE MXONE PABX.

PLEASE GIVE ME A DETAIL SOLUTION.


REGARDS

 

[jetsz2013]

trace -dir

 

[jetsz2013]

with start and stop

 

[teenu3]

Hi jetsz2013,

Thanks for replay but trace with what command pls let me know the command pattern.

regards

 

[snor]

maybe an Ethernet trace will show you a little bit more.
log on to mx-one as super user.
start tcpdump on the server where SIP phone is registered. (tcpdump -i eth0 -s 0 -w /tmp/tcpdump.pcap)
make test calls.
stop trace with "strg" "C"
copy file tcpdump.pcap to a PC, and analyse the trace with "wireshark"

best regards,
Norbert

 

[himdp]

If I understand correctly, the PABX A to which the SIP phones are registered is not an MX-One.
If this is the case, there is no point to do trace -dir or tcpdump as the only connections on the MX-One are E1 circuits.
Is the MX-One TSE or TSW and what release level?
You would be better off examining the lost signals and see if any relate to the dropped calls.

 

[teenu3]

Hi himdp,

you are right the pabx is only connected to sip system thru e1.
pabx is tsw sp8.
what i want is the command pattern to trace the e1 connection.
the e1 from sip system is connected to lim no 2,the out going e1 is connected to lim no 1. that means incomming call from sip system comming in lim to and going out to outside world thru lim 1 e1. i want to trace were the call is dropping.
please help me with the command sequence.

regards

 

[himdp]

It will not be easy to trace a failed call in a live system unless you can define the exact call case which makes the call drop.

What is the trunk block use in the E1 connections?
You will need to do signal tracing on those blocks (SLP60?) to determine where the call release signal is originating from.
i.e PSTN, TSW or PABX A.

As a start I would do HIREI; then wait for a report of a call release and then do HIEDP; HIMDP; to see if there are any lost signals to indicate a TSW software fault.

If the connections are ISDN (SLP60) to both PSTN and PABX A then do-
STISR;
STUNI:LIM=?,UNIT=SLP60;
wait for a call release error then do-
STTSE:TRI=1;
STRAP:TRI=1;
Log the output to a file and run it through STI (part of SAM toolbox) which will give you an idea of what is releasing the call.
Good luck

 

[teenu3]

Thanks himdp,

i will try my best
thanks once again .

regards

 

[teenu3]

Hi all,

on this subject i re-arranged the E1 between the two lims.before the re - arrange the sip system says the cause of disconnection is code 42,103 and 31. now 42 and 103 problem solved but still some disconnection is there under code 31.
any sugesion to solve it.

regards