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Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 08:04
by Joris06
haroun wrote:YES IT's
i see overfolow to assoct is ok
what about ByPass on forwarding (under rights)?
and default overflow type /adresse unde (abbv area)?
Bypass on forwarding = 0
Default overflow type = Forward on busy and no reply
Default overflow adressee = Associated
I think the last two are ok, but I don't know about the first one.
haroun wrote:btw if the caller to IVR is external one see cnx cos compatibility between external trunk (5 in normal case)of the caller and ivr sipTrunk ( =31) to see if transit is allowed
For now, all of my tests are internal (from user to user), however, IVR works well if accessed directly, or from a DDI number translation rule.
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 08:26
by haroun
try to change Bypass on forwarding = 1
Default overflow type = noforward
Default overflow adressee = nothing
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 08:32
by Joris06
It changes nothing

Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 09:09
by haroun
WHAT's the acces class of service of your extension 6042
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 09:17
by Joris06
I don't have any parameter named "Access class of service" in my user's settings.
This is what I have about the categories :
Code: Select all
│ Charging Category + Justified │
│ Public Network Category : 4 │
│ External Forwarding Category : 255 │
│ Tel.Facility Category Id : 0 │
│ Connection Category Id : 0 │
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 09:39
by haroun
IT always under classe of services -categories-
┌─Select an object─────────────────────────────────────────────────┐
│ │
│ Shelf Specific Telephone Services │
│ Media Gateway Atm │
│ Dect System Events Routing Discriminator │
│ System Security and Access Control │
│ Translator IP │
│ -> Categories SIP │
│ Attendant DHCP Configuration │
│ Users Alcatel-Lucent 8&9 Series │
│ Profiled Users SIP Extension │
│ Set Profiles Encryption │
│ Groups Passive Com. Server │
│ Abbreviated Numbering SNMP Configuration │
│ Phone Book │
│ Entities │
│ Trunk Groups │
│ External Services │
│ Inter-Nodes Links │
│ X25 │
│ DATA │
│ Applications │
│ │
└──────────────────────────────────────────────────────────────────┘
┌─Select an object─────────────────┐
│ │
│ -> Access Category │
│ Connection Category │
│ Transfer Category │
│ Private Calls Connect.Categ. │
│ Phone Facilities Categories │
│ │
└──────────────────────────────────┘
┌─Consult/Modify: Access Category────────────────────────────────────────────┐
│ │
│ [ All instances ] │
│ [ Set filters ] │
│ [ Select attributes ] │
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Public Network Category : 4 │
│ │
└────────────────────────────────────────────────────────────────────────────┘
┌─Consult/Modify───────────┐
│ │
│ running ... 1 │
│ │
└──────────────────────────┘
┌─Consult/Modify─┐
│ │
│ running ... │
│ │
└────────────────┘
┌─Consult/Modify: Access Category────────────────────────────────────────────┐
│ │
│ NDDI overflow on free set │
│ │
│ Night : 1 │
│ Day : 1 │
│ Mode 1 : 1 │
│ Mode 2 : 1 │
│ │
│ DDI overflow on free set │
│ │
│ Night : 1 │
│ Day : 1 │
│ Mode 1 : 1 │
│ Mode 2 : 1 │
│ │
│ Tie Line overfl. on free set │
│ │
│ Night : 1 │
│ Day : 1 │
│ Mode 1 : 1 │
│ Mode 2 : 1 │
│ │
└────────────────────────────────────────────────────────────────────────────┘
IWAS TRYIng to recrate you scenario, an each time i call the overflow works fine
all overflow authorised for extension to extension or to sip device (sip trkgroup) even call source is external or internal the overflow works!
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 09:51
by haroun
timer 4 &timer 9 values?
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 09:58
by Joris06
I have the exact same values in those fields...
I tried to set the associated number to the number of another phone set, it works.
I tried to create an external voicemail number and set it in the associated number field : the call overflows on the standard voicemail, even the number is not good !
And if I set a number starting with 61 : no overflow at all, I don't understand why...
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 10:16
by Joris06
If I set a permanent forwarding on the IVR directly from my phone set, it works fine ! With a forwarding on busy, it works to !
If on the same phone set, I set a forwarding on no answer with a random extension, it works to.
But, if the forwarding on no answer number is the one of my IVR : nothing happens.
Am I going crazy ????
Re: Overflow on SIP trunk if no answer
Posted: 25 Nov 2014 10:21
by Joris06
haroun wrote:timer 4 &timer 9 values?
In system -> timers ?
4 = 50
9 = 0