Panasonic Call Limiting with SIP

[CMUK]

Hi All,
I have a customer with an NS700 and I'm trying to limit one number to only receive one call at a time.
The call should firstly route to the TVM50 then it points it to the extension it needs to go to, however I am having major issues achieving this,
I am made to believe the TVM50 cannot give back a busy signal also (I have found this to be true so far),
Originally I had 6x Gamma SIP Trunks. My next stage was making a new SIP account (2 channels minimum for some reason, however I have only set 1 on up PBX) and port the number.
Now we have the number on the new SIP account but i'm now getting 2 calls in at a time despite the system only having one trunk with this number on it, I'm going to site next Friday however this one is a strange one, has anyone else overcame this problem?
Thanks

Calum M
ACSS

 

[AkraTEL]

we have this problem too,
we have NS500 Panasonic and with 2 LIC for SIP Extension,
after 20 sec we cant continue our call with these sip extension. but for SIP trunk every thing is ok,
but for terminate one call that recieved by sip trunk we cant use from our PSTN, in TDE model we could active D05 in 2.9 system option recerved bit, resolve out problem but for NS series, this account *PCCKXTDA* cant work, is there any one that know how we cant resolve those ?

many thanks

 

[AJ22]

You don’t need two SIP Provider accounts you can split it.

Put in Your base number from provider set that as a basic channel only normally is set that for outbound

Put your other ddi in as the same

You will get busy i have done this a few times

Tvm will give busy when ports are busy

 

[CMUK]

Thanks for the reply
I have tried that just there and not getting correct result.
When I OUS the main SIP account I get my busy after one call as I would expect, it's as if it jumps over to the other channels of which it isn't even registered to receive calls

Calum M
ACSS

 

[AJ22]

Do you have 2 Gamma accounts pointing to the same WAN? Last time I checked you couldnt do that, but achievable with sip registrar.

I would work with the one sip account only and get rid of the second. I would put the ddi you want to limit in as the user name etc and set to 1 or 2 ch depending what you want.

Then I would do exactly the same for the base no and and set that to be used for outgoing.

It does work, as said for testing get rid of the 2 accounts use the one and re test.

If you have one ch of sip against a certain no yuubwill get one call of you have 2 your third call will be busy and so on licence /provider permitting

 

[CMUK]

Yes 2 gamma accounts to same WAN, I think this is definitely the problem as the number will ring on either circuit.
I have never used SIP Registrar for Panasonic SIPS, I usually just disable it - this is Nimans that told me to do it this way a while back and been doing it ever since.
How does the SIP Registrar work, is it just the username/password Gamma provide?

Calum M
ACSS

 

[AJ22]

to get it working the way you want this what you do it on the system . worked for me every time. use same sbc server address and registrar disabled.

gamma endpoint wont let you enter a WAN IP that is already in use by them on another account.. this must be a new thing if it letting you do that.. as said i would test with just one account so you know pbx works then find out what has been done on the gammas side as doesn't sound right and you dont need the 2 accounts to make it work. i see the logic though.

 

[CMUK]

Okay I have got the second gamma account out the equation, I have done as you have shown, however I am still getting more than 1 call at a time, I have attached photos

Calum M
ACSS

 

[AJ22]

Just ran another test with one of the systems we have out there and this setup should work for you. Are the 2 numbers you are testing with on the same gamma account and on the same Sig serv address ? What version you on and have you done a re boot since changes ?

 

[CMUK]

2 numbers im testing are on same address and same gamma account, I tried a reboot this morning but no joy.
System is version 004.10050

Calum M
ACSS

 

[phones001]

Not sure if you ever resolved this but I would suggest disabling call waiting on the voice mail ports as I think this is enabled by default.