Alcatel Unleashed

HI Guys

I need some direction please. I think I am missing something somewhere.

I have a new customer with SIP trunk setup but I am struggling to get the calls to DDI extensions.

So the Default DDI translation is set up as

First external number :27116893600
First Internal number :3600

SIP GW 1 is created and it is linked to TG 10 on the system.

I can see in the SIP Invite the provider is sending +2711689xxxx and the NPD for incoming is set up as follow:
It does not matter what I change the Calling Numbering plan to, it does not change the outcome.
-----------------------------------------------------------------------------------
Name : GBN
│ Calling Numbering plan ident. + NPI/TON ISDN International
│ Called numbering plan ident. + NPI/TON: ISDN Unknown
│ Authorize personal calling num use + False
│ Install. number source + None used
│ Default number source + None used
│ Called DID identifier : 0
│ Calling/Connected DID identifier : 0
│ Location ID Source + None
│ Location ID : 0
----------------------------------------------------------------------------------

But all calls end up in the Default entity and follow the default routing in there. Currently I am just sending all calls to a Hunting group.

I asked the provider to remove the + in front of the number in the invite but they say they cannot strip it as it will affect other customers on their network and I need to accommodate the +.

Please see my SIP GW setup

Node Number (reserved) : 101
│ Instance (reserved) : 1
│ SIP External Gateway ID : 1
│
│ Gateway Name : GBN
│ SIP Remote domain : 172.31.20.35
│ PCS IP Address : ---------------------------------------
│ SIP Port Number : 5060
│ Transport type + UDP
│ Belonging Domain : xxxxxxxxxxxxx
│ Registration ID : 27116893600
│ Registration ID P_Asserted + True
│ Registration timer : 1800
│ SIP Outbound Proxy : 172.31.20.35
│ Supervision timer : 0
│ Trunk group number : 10
│ Pool Number : -1
│ Outgoing realm : ---------------------------------------
│ Outgoing username : ---------------------------------------
│
│ Outgoing Password : --------------------
│ Confirm : --------------------
│
│ Incoming username : ---------------------------------------
│
│ Incoming Password : --------------------
│ Confirm : --------------------
│
│ RFC 3325 supported by the distant + True
│ DNS type + DNS A
│ SIP DNS1 IP Address : 172.31.22.42
│ SIP DNS2 IP Address : ---------------------------------------
│ SDP in 18x + False
│ Minimal authentication method + None
│ INFO method for remote extension + False
│ To EMS + True
│ SRTP + RTP only
│ Ignore inactive/black hole + False
│ Contact with IP address + False
Dynamic Payload type for DTMF : 101
│ Outbound Calls 100 REL + Supported
│ Incoming Calls 100 REL + Not Requested
│ Gateway type + Standard type
│ Re-Trans No. for REGISTER/OPTIONS : 2
│ P-Asserted-ID in Calling Number + False
│ Trusted P-Asserted-ID header + False
│ Diversion Info to provide via + History Info
│ Proxy identification on IP address + False
│ Outbound calls only + False
│ SDP relay on Ext. Call Fwd + Default
│ SDP Transparency Override + False
│ RFC 5009 supported / Outbound call + Not Supported
│ Nonce caching activation + NO
│ FAX Procedure Type + T38 only
│ DNS SRV/Call retry on busy server : 0
│ Unattended Transfer for RSI + NO
│ Redirection functionality + NO
│ Attended Transfer + NO
│ Send BYE on REFER + YES
│ Support Redirection response + NO
│ OPTIONS required + YES
│ Support UTF8 characters set + NO
│ Support CSTA User-to-User + NO
│ DDI destination number + TO
│ Video Support Profile + Not Supported
│ UPDATE in Allow header/INVITE + Optional
│ RFC 4904 supported + NO
│ Bulk registration (RFC 6140) + NO
│ RFC3264 m-line + True
│ Sendonly for hold + False
│ In Band DTMF + NO
│ SIP trunk Recording + NO
│ Send user name in SIP + User name else user number
│ Session Timer : 1800
│ Min Session Timer : 900
│ Session Timer Method + RE_INVITE
│ Allow Direct RTP + True
│ IP Domain : -1
P-ANI Header + None
│ Support G722 + YES
│ Support G711 + YES
│ Support G729 + YES
│ Support Re-invite without SDP + True
│ Registration on proxy discovery + False



The SIP invite attached (motortrace 3):

Dialing to 0116893674 from 0123589944

did you managed the trunk 10, "DID transcoding" to True ?

Hi Sadim

Yes the DID transcoding is set to True.

Node Number (reserved) : 101 │
│ Trunk Group ID : 10 │
│ │
│ Trunk Group Type + T2 │
│ Trunk Group Name : GBN SIP │
│ UTF-8 Trunk Group Name : --------------------------------------- │
│ Number Compatible With : 0 │
│ Remote Network : 11 │
│ Shared Trunk Group + False │
│ Special Services + Nothing │
│ Node number : 1 │
│ Transcom Trunk Group + False │
│ Auto.reserv.by Attendant + False │
│ Overflow trunk group No. : -1 │
│ Tone on seizure + False │
│ Private Trunk Group + False │
│ Q931 Signal variant + ISDN all countries │
│ SS7 Signal variant + No variant │
│ Channel selection type + Quantified │
│ Auto.DTMF dialing on outgoing call + NO │
│ T2 Specification + SIP │
│ Homogenous network for direct RTP + NO │
│ Public Network COS : 31 │
│ DID transcoding + True │
│ Can support UUS in SETUP + True │
│ Associated Ext SIP gateway : -1 │
│ │
│ Implicit Priority │
│ │
│ Activation mode : 0 │
│ Priority Level : 0 │
│ │
│ Preempter + NO │
│ Incoming calls Restriction COS : 10 │
│ Outgoing calls Restriction COS : 10 │
│ Callee number mpt1343 + NO │
│ Overlap dialing + YES │
│ Call diversion in ISDN + NO

┌─Review/Modify: Network Routing Table───────────────────────────────┐
│ │
│ Node Number (reserved) : 101 │
│ Instance (reserved) : 1 │
│ Network Number : 11 │
│ │
│ Rank of First Digit to be Sent : 1 │
│ Incoming identification prefix : -------- │
│ Protocol Type + ABC_F │
│ Numbering Plan Descriptor ID : 100 │
│ ARS Route list : 1 │
│ Schedule number : -1 │
│ ATM Address ID : -1 │
│ Network call prefix : -------- │
│ City/Town Name : -------------------- │
│ Send City/Town Name + False │
│ Associated Ext SIP gateway : 1 │
│ Enable UTF8 name sending + True │
│ │
└────────────────────────────────────────────────────────────────────┘

and the parameter mgr/system/other system/sip parameters/Via Header_ Inbound Calls Routing is it at True?

Yes it is set to true

and what's the selected ddi for that Trunkgp? and how it's selected auto mode , normal or expert one ?

Hi Haroun

Are you referring to NPD?

┌─Review/Modify: Trunk group NPD selector────────────┐
│ │
│ Node Number (reserved) : 101 │
│ Trunk Group ID : 10 │
│ Instance (reserved) : 1 │
│ │
│ Public NPD ID : 100 │
│ Private NPD ID : 0 │
│ Management Mode + Normal │
│ │
└────────────────────────────────────────────────────┘

NPD 100 is what is set up

│
│ Node Number (reserved) : 101 │
│ Instance (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Description identifier : 100 │
│ │
│ Name : GBN │
│ Calling Numbering plan ident. + NPI/TON ISDN International │
│ Called numbering plan ident. + NPI/TON: ISDN Unknown │
│ Authorize personal calling num use + False │
│ Install. number source + None used │
│ Default number source + None used │
│ Called DID identifier : 0 │
│ Calling/Connected DID identifier : 0 │
│ Location ID Source + None │
│ Location ID : 0

Try to change those two parameters cuze I see sip private call in trace
Associated Ext SIP gateway : -1 to 1
Private NPD ID : 100