SIP Trunking OXE. Can not route incomming call

[pantisimus]

PCX is OXE. Release R9.1-i1.605-21
Connected to the Public Operator.
Calls from OXE to the Operator are going without problems.
But, I can not activate DDI for incomming calls. It is always ringing on the "INVITE" number - Extension 100. 100 from 381117159100
From trace :INVITE : 381117159100@10.1.1.21:5060 ; user=name

In the following trace, I can see the called number 381117159101, but finally is ringing 381117159100(This number is the main one and used for registration!!! Later in the log, I can see the conversion message and finally rings 381117159100. It is obvious from the log.

BUT, how to manage to have DDI???

Best Regards

Message received SIP ----> UA (neqt : 992)
| INVITE : 381117159100@10.1.1.21:5060 ; user=name
| From : <> 0648109051@10.0.0.2:5060 ; user=phone
| To : <"381117159101 381117159101"> 381117159101@ims.telekomsrbija.com:5060 ; user=phone


SIP Trace :
(074086:000308) SIP : [send_to_motor_eqt_release] ipcSend resultat : 0 sur eqt : 992
(074086:000309) SIP : [ipc_send] envoi eqt release 992
(074086:000310) SIP mise_a_jour_des_canaux ind=0 neqt=992 allocation=0
(074105:000311) event_init_screen_mem : line 1 tscreen '648109051 '
(074105:000312) message_enregistrement() box_state=6 ML_NOT_MSG=6
(074105:000313) message_enregistrement() mess_enreg=0 autre_mess=0
(074172:000314) INIT MEVO : uti_inidcom init mevo neqt 433
(074193:000315) INIT MEVO : uti_inidcom init mevo neqt 433
(074193:000316) INIT MEVO : uti_inidcom init mevo neqt 434
(074257:000317) SIP : Global_get_mcdu, unknown hostname
(074257:000318) SIP : Global_get_mcdu, unknown hostname
(074257:000319) SIP : message INVITE arrive sur le neqt : 992.
(074257:000320) INIT MEVO : uti_inidcom init mevo neqt 992
(074257:000321) SIP : ctrl_sip evt : 10752.
(074257:000322) +------------------------------------------------------------+
(074257:000323) | Message received SIP ----> UA (neqt : 992)
(074257:000324) | INVITE : 381117159100@10.1.1.21:5060 ; user=name
(074257:000325) | From : <> 0648109051@10.0.0.2:5060 ; user=phone
(074257:000326) | To : <"381117159101 381117159101"> 381117159101@ims.telekomsrbija.com:5060 ; user=phone
(074257:000327) +------------------------------------------------------------+
(074257:000328) | SDP :
(074257:000329) | @IP:port = 10.0.0.2:31660
(074257:000330) | ALGOS :
(074257:000331) | PCMA
(074257:000332) | G729
(074257:000333) | DTMF : 98
(074257:000334) | DIRECTION : SEND & RECEIVE
(074257:000335) | cac : false
(074257:000336) | Prack_Required: 1
(074257:000337) | Allow_UPDATE: 1
(074257:000338) | -->Call to an external gateway : 2
(074257:000339) | autoAnswer : false
(074257:000340) +------------------------------------------------------------+
(074257:000341) ctrl_payloads_on_reception_sdp payloads_recu[0]=0
(074257:000342) ctrl_payloads_on_reception_sdp payloads_recu[1]=17
(074257:000343) ctrl_payloads_on_reception_sdp dtmf_payload 98
(074257:000344) ctrl_payloads_on_reception_sdp payloads_recu[2]=255
(074257:000345) ctrl_invite-->Allow_INFO_received=TRUE
(074257:000346) ctrl_invite-->n_gw_ext=2
(074257:000347) is_ems_ext_gw-->ext_gw=2,Result=0
(074257:000348) ctrl_invite-->sip_info_enable=0,dtmf_payload=98
(074257:000349) SIP : ctrl_sip content_type : 1
(074257:000350) build_ie_redirec_from_initial_called: type_of_url = 3
(074257:000351) build_ie_redirec_from_initial_called: 381117159101@ims.telekomsrbija.com074257:000352) redirec_from_initial_called: URL_type_local_num


...

_____________________________________________________________________________
| (074257:000412) Concatenated-Physical-Event :
| long: 95 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0 term: 4 type a5
| tei: 0 <<<< message sent : SETUP [05] Call ref : 00 12
| SENDING COMPLETE
|______________________________________________________________________________
|
| IE:[04] BEARER_CAPABILITY (l=3) 80 90 a3
| IE:[18] CHANNEL (l=2) a0 90 -> T2 : No B channel
| IE:[6c] CALLING_NUMBER (l=12) -> 01 81 Num : 0648109051
| IE:[70] CALLED_NUMBER (l=13) -> 80 Num : 381117159100
| [9f] Non-locking shift. codeset : 7
| IE:[06] EI_IP_PAYLOADS (l=2) : (COMP/ECE/VAD) -> G711a/0/0 G729/0/0
| [97] Locking shift. codeset : 7
| IE:[0a] EI_RTP_INFO (l=30)
| -> stop_packet=0 stop_rtp=0 h323=0 wc=0 rf=0 udp=1 rqm=0
| -> Transm_Bande=1 detection_Q23=1 dtmf_payload=98
| -> Port RTP = 31660, IPv4 : 10. 0. 0. 2.
| -> Port RTCP SR = 31661, IPv4 : 10. 0. 0. 2.
| -> Port RTCP RR = 31661, IPv4 : 10. 0. 0. 2.
| -> Port Fax = 0, IPv4 : 0. 0. 0. 0.

[frank]

You will need two things:
- DID Translation turned ON in your trunk
mgr / trunk / your_sip_trunk / DID Transcoding [NO]->[YES]

- Default Translator configured.
mgr / Translator / External numbering plan / Default DID num. Translator /
Here you will have to create translation from your incoming number, to the number you want to ring internally.

Ex:
First external number: 381117159101
First internal number: 100

That will ring extension 100 when 381117159101 is called.

[pantisimus]

Well, this is a problem.

I have managed Default DID like 381117159100 - 100 Range 10
DDI is active in trunk group

But if I dial from outside 381117159101, or 102, or any 10x, extension 100 is ringing. Not Extension 101 or 102 like in DDI :
┌─Review/Modify: DID number translator rules────────
│ Node Number (reserved) : 1
│ Instance (reserved) : 1
│ Instance (reserved) : 1
│ DID num. transl. identifier : 0
│ First External Number : 381117159100
│ First Internal Number : 100
│ Range Size : 10
│ Unique Internal Number + NO

In the SIP trace, I see ringing and to Field is 381117159101, but ringing on 100!!!
Number 381117159100 is the main number used also for authentication/registration.
Please take a look to the traces. For calling 381117159100 and 381117159101.
Complete management is also attached.
Please, try to help me.

[haroun]

confusion you configure sip gateway 2 as local but in is declared as external
Review/Modify: Network Routing Table───────────────────────────────┐
│ │
│ Node Number (reserved) : 1 │
│ Instance (reserved) : 1 │
│ Network Number : 14 │
│ │
│ Rank of First Digit to be Sent : 1 │
│ Incoming identification prefix : -------- │
│ Protocol Type + Dependant on Trunk Group Type │
│ Numbering Plan Descriptor ID : 40 │
│ ARS Route list : 0 │
│ Schedule number : -1 │
│ ATM Address ID : -1 │
│ Network call prefix : -------- │
│ City/Town Name : -------------------- │
│ Send City/Town Name + False │
│ Associated Ext SIP gateway : 2 │
│ Enable UTF8 name sending + True

other thing why two sip trunk for same carrier (one for outgoing other one for incoming) ?
why creating a trunk group for oxe local sip ? no need for that in you case i will put trunk group and network number to -1
change network number in trunk group 26 and 27 to 14
in network routing table 14 associeted EXSIPTGW =1
npd selector for both trunk 26 & 27 pub 40 priv 0
and last thing
your 381117159100 range 100 are them local user aor network users ?

[pantisimus]

Thanks for help !!!!
I have modified everything according your suggestions.
Outgoing call is completely OK.
But incomming is again ringing on the extension 100 (external 381117159100) when I am dialling from operator the phone number 38117159101. Call arrives in OXE.
Default DID is 381117159100 to 100, range 10.

This is what I see in motortrace 3 trace.
INVITE mesage is for 381117159100 that is our main number used for registration, but I am dialling 381117159101!
"To" field is OK 381117159101!
Field : P-Called-Party-ID: <sip:381117159100@ims.telekomsrbija.com>
Again 381117159100, not 9101!
While ringing on 100, in trace it is "To" field correct with number 381117159101!... But ringing on 100!!!
Any idea?
Please...

1396515467 -> RECEIVE MESSAGE FROM NETWORK (10.0.0.2:5060 [UDP])
----------------------utf8-----------------------
INVITE sip:381117159100@10.1.1.21;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bKbfqb9e10corg9nk867g0.1
To: "381117159101 381117159101"<sip:381117159101@ims.telekomsrbija.com>;cscf
From: <sip:0648109051@mn3sip.telekom.yu;user=phone>;tag=2129635428-1396515484316
-
Call-ID: BW105804316030414904066754@10.10.1.137
CSeq: 325556559 INVITE
Max-Forwards: 27
Content-Length: 261
Contact: <sip:0648109051@10.0.0.2:5060;transport=udp>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
Accept: multipart/mixed
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-hotsip-FileTransfer+xml
Supported: 100rel, timer
P-Asserted-Identity: <sip:+381648109051@mn3sip.telekom.yu;user=phone>
Privacy: none
P-Charging-Vector: icid-value=044cbd910405d530b233380dd9b933
Min-SE: 180
Session-Expires: 1800
P-Called-Party-ID: <sip:381117159100@ims.telekomsrbija.com>

[murraya]

hi, have you looked at
system/DH/other system/DH/SIP parameter/Via Header
maybe try switching it

[haroun]

again same question your sets are locals or network's one.if they are local i don't understand why routing table is managed?
what about if uou change the lenght of external numebr from whole to the last for digit : FEN 9100 FIN 100 Rg 10
yeah check sip parmeter via header 'by default is switched off)

[pantisimus]

Extensions mentioned are local. No ABCF network here.
SIP parameter in System: Via Header_ Inbound Calls Routing + True

Incomming SIP trace is :
1396519272 -> RECEIVE MESSAGE FROM NETWORK (10.0.0.2:5060 [UDP])
----------------------utf8-----------------------
INVITE sip:381117159100@10.1.1.21;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bKd1dbaj00d8v02tk8l7o0.1
To: "381117159101 381117159101"<sip:381117159101@ims.telekomsrbija.com>;cscf
From: <sip:0648109051@mn3sip.telekom.yu;user=phone>;tag=1707959085-1396519286506
-

It seems that OXE Ver.9.1 is not looking at "To" field. Only the "INVITE".
I think that it is not possible to set this to OXE R9.1 to use "To" field like in Asterisk and Audiocodes.

But I am not sure.

[haroun]

and if you uncheck it what's happening

[haroun]

i supspect that oxe is not tarnscoding at all cause of network 14
i reproduce you case (similair) call from isdn to sipextension (localsip) and to sipdevice (network number) connected to externalgateway;
INVITE sip:51018@10.235.85.34:62506;rinstance=f8befafa2b5516ec SIP/2.0

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO

Supported: replaces,timer

User-Agent: OmniPCX Enterprise R9.1 i1.605.38

Session-Expires: 1800;refresher=uac

Min-SE: 900

Content-Type: application/sdp

To: "51018" <sip:51018@10.235.96.130;user=phone>

From: "038876668" <sip:038876668@10.235.96.130;user=phone>;tag=1c4692513eca35232960bad726a3453d

Contact: <sip:038876668@10.235.96.130;transport=UDP>

Call-ID: 3973399201bc9df06ae9fbd13f1f90fa@10.235.96.130

CSeq: 1423354222 INVITE

Via: SIP/2.0/UDP 10.235.96.130;branch=z9hG4bK0c69162a80b85d43e2c8c18fcbb63cc7

Max-Forwards: 70

Content-Length: 239