OXO sip trunk with Genesys

[luis.molinas]

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.

[giovanni.attolini]

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

[luis.molinas]

Hi gio,

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

some any other idea?

Best Regards.

[giovanni.attolini]

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

[luis.molinas]

I attached my configuration, can you see please?

[giovanni.attolini]

Please, attach the configuration of ARS -> Gateway Parameters

[luis.molinas]

I attached.

[giovanni.attolini]

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.

[tot3nkopf]

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

Is 2000 in an attendant group?

[luis.molinas]

Hi tot3nkopf,

the numbering 2000 is an extension of the SipServer.

I will try to declare the sip trunk as public.

Regards.