Alcatel Unleashed

Hi all,

I need someone to help me with this configuration:
I have installed a OXOR9.0 and am wanting to integrate with Genesys 8.1 SipServer
I created the sip trunk on both sides.

objects:
OXO IP = 192.168.92.246
OXO endpoint number = 4016
SipServer IP = 192.168.11.120
SipServer endpoint number = 2000


When I make a call from one endpoint of OXO to an endpoint of SipServer I have the following result:

SIP/2.0 404 Not Found

In the INVITE see that the number of the endpoint of SipServer is modified.

INVITE sip: E2000@192.168.11.120, user = phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
Supported: 100rel, from-change, timer, histinfo
User-Agent: OXO_GW_820/032.007
Session-Expires: 43200
P-Asserted-Identity: "Soledad Blazquez" <sip:6014@192.168.92.246;user=phone>
History-Info: <sip:E2000@192.168.11.120;user=phone>; index = 1
To: <sip:E2000@192.168.11.120;user=phone>
From: "Soledad Blazquez" <sip:6014@192.168.92.246;user=phone>, tag = 1f2b56ca5fbb7398397079e52a323e23
Contact: "Soledad Blazquez" <sip:6014@192.168.92.246;transport=UDP;user=phone>
Content-Type: application / sdp
Call-ID: 836705304fefacd978e8e38d753f3e42@192.168.92.246
CSeq: 245760281 INVITE
Via: SIP/2.0/UDP 192.168.92.246; rport; branch = z9hG4bKd08aeae6ef33272fb96b9efb89965207
Max-Forwards: 70
Content-Length: 211

v = 0
o = 1358340180 1358340180 IN IP4 default 192.168.92.246
s = -
c = IN IP4 192.168.92.54
t = 0 0
m = audio 32000 RTP / AVP 18 106 4 8 0
a = sendrecv
a = rtpmap: 106 telephone-event/8000
a = fmtp: 106 0-15
a = maxptime: 60


Anyone know how I can do to make the OXO no longer sending the letter E before the number of the endpoint?

thank you very much and best regards.

Verify:
OMC -> Numbering -> Automatic Routing Selection -> SIP Public Numbering -> Called Prefix (Outgoing)

Hi gio,

Thks for your answer, I checked and there isn't nothing on this option configured.

some any other idea?

Best Regards.

Are you changing this on ARS? You can accidentaly modifying the dialing before send the call.

I attached my configuration, can you see please?

Please, attach the configuration of ARS -> Gateway Parameters

I attached.

Try this:
1 - Make sure that the ARS route is associated to the Gateway Parameters Index 1,
2 - In Gateway Parameters associate SIP Numbers Format Index to Index 1 too;

So you will associate that ARS route to that SIP Public Numbering.

try to declare SIP trunk as public.
Play also with relevant noteworthy addresses.

Is 2000 in an attendant group?

Hi tot3nkopf,

the numbering 2000 is an extension of the SipServer.

I will try to declare the sip trunk as public.

Regards.