Caller ID with BC11 over analogue extensions is working with certain patch, so it works, but AASTRA (X-Ericsson) should provide you with SACOS and patch file for this.
Regards
Did any one try before SIP Extension over inetrnet or WAN Network.
I am using AASTRA SIP phones, the extension is registered to the telephony server (TSE 3.2) but the speech is one way.
What should I configure in TSE?? Is it only IP Extensions??
I know well that we have Redundancy on the network and server level in TSE but I would like to know if the below scenario could work.
Do we have hotstandy for the calls in MX-one Telephony server??
Let's say that I have a call (IP Phone)and my TS (LIM1) went down, is it possible for the IP...
Hi,
My system is TSW SP5 and have a problem with caller ID (CLIP)on the operator.
Im using ISDN E1, and no problem with the caller ID over digital extensions but there is a problem with the caller ID on the operator.
(No problem in caller ID on the operator if I am receiving an internal call)...
changed BCAP but the same!!!
Could you please post your parameters (ROCAP,RODAP,RODDP,nutrp) at the three systems also the extension category...
What releases do you have??
Did you do all translations at ystem B as required??
I have already did this and it is the same result.
what do you mean by TRC=R??!!
This is my NUTRP in SYSTEM B
<NUTRP;NUMBER CONVERSION DATA
ENTRY CNVTYP NUMTYP ROU TARDST PRE TRC NEWTYP CONT BCAP HLC
601 1 0 3...
Hi Daddy,
CNVTYP=2 couldn't be used together with TARDST..
CNVTYP must be 1 to be used with TARDST..
Daddy, does the scenario Im trying is acheivable?? do you have some sites with same scenario??
I have done the traces many times before for a transit call ,at the transit switch ofcourse...
system B (TSWSP04):
Route 1 is the PRI T-interface route to system A.
Route 2 is the Qsig route to the telephony server (System C).
<ROCAP:ROU=ALL;ROUTE CATEGORY DATA
ROU SEL TRM SERV NODG DIST DISL TRAF SIG BCAP
1 7130000000000010 4 2110000011 0 30 128...
Hello Daddy,
I tried that in the transit switch but the same!!! I always got 601 when I receive the call at system A (pstn)..
Again my parameters:
<RODDP:DEST=ALL;EXTERNAL DESTINATION ROUTE DATA
DEST DRN ROU CHO CUST ADC TRC SRT NUMACK PRE
2 2...
Hi,
I have TSW system connected with Mx-one telephony server over Q-sig...where the TSW is connected to another TSW using DSS protocol (T-interface).
System A :TSW which works like PSTN.
System B: is the TSW which is in between the two systems, which is connected Qsig with the telephony server...
Hello,
For testing, I made a new route in the system and added a trunk line to this.
I changed the TRM of route from 4 to 7 (by command ROCAI) , and for each route TRM value I changed the TRM value of an extension from 0 to 3 (by command KSCAC). i.e, a total of 16 options.
Then I made calls...
I typed LSCOP and found the ATTN address is 2... and as you aware that the attenuation value of this address is always 0 db and couldn't be changed...
Any comments..
Should I change the TRM for the extension category??
Many thanks.
The Applicationsystem is standard.
The system is a new sale...so it is the new hardware of TSW.
Below you find the Route parameters.
ROCAI:ROU=1,SEL=0110000001000010,TRM=7,SERV=3110000000,SIG=411100000000,TRAF=00151515...
Did any one experience in the low audio level on Analogue trunks in TSW Sp03??
I did the following:
1.cnanged TRM in ROCAI to 7
2.tried with D7 in VARC (short or long line)
3.SACOS:LIM=1,UNIT=SCP,SECTOR=REL,ADDR=58&89,DATA=6;
LSATI:LIM=1,AVAL=6-x,LAW=A;
x:0-15
But no improvemnet the Audio...
Hello Whorsedady,
Sorry to tell you that after the test I verified again that once the file becomes in the passivestat, it won't never back to ready status unless I end the CIL and reconfigure it again!!! Any comments...
I have been in a waiting for the file to come back in ready status for 15...
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