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Offline hendro

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Gensys SIP Server with Alcatel OXE SIP Trunk
« on: May 27, 2008, 01:44:04 AM »
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Dear All,
We are implementing genesys SIP Server using Alcatel OXE as the gateway for outbound PSTN call.

We need help here about the OXE or SIP Server configuration to make it work.
The problem is that if we can call the PSTN number from the SIP Softphone and have no problem with it, but when we can't do it from agent desktop using the "TLine.TDial" command.
So when we dialed from the agent desktop, the agent SIP Phone will ringing, but when we accept the call, it just hangup.

Could anyone help us about what is the standard configuration for this kind of solution. I mean what should I setup inside the OXE and genesys. Actually, It is only a preview outbound.

We will very appreciate any kind of help, since we already have no idea how to do it.

Thanks,

Best Regards,
Hendro

Offline cavagnaro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #1 on: May 27, 2008, 02:10:27 AM »
Are you using OXE agents of SIP Agents? If OXE, if a phone set can do it, an agent should be able too. Remember that in OXE an agent is only taking the ProACD phoneset number. What happens if you dial manually from agent?
Is authentication configured on OXE? Is the Sip agent registered on OXE?

Any way, can you post SIP Traces to try to understand better the problem?

Offline hendro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #2 on: May 27, 2008, 09:29:38 AM »
Thanks Cavagnaro for your kind reply,

I am using Genesys SIP agent, the agents is registered to Genesys SIP Server. So in this case OXE will be acts as a media gateway. I mean we use OXE for external PSTN connections.

I have created a DN with TRUNK type, and setup the prefix for it. So everytime sip user dial this prefix, it will go to OXE and OXE will route it to the PSTN trunk.

When I dialed it directly, using the sip softphone by dialing "9" + [PSTN number], it can connect to the PSTN number, but if I am using the agent desktop application to dial, it cannot connect.

It looks like a conference call for me, when I dialed it from the agent desktop application.

I attached here the log I got from genesys sip server, and from OXE ("traced" command).

The configuration is like this :
Genesys SIP Server IP Address : 172.9.2.85
Genesys Configuration Server : 172.9.2.80
OXE IP Address : 172.9.2.95

Client/Agent IP Address : 172.9.2.213
SIP Agent DN : 3085
Dialled PSTN Number : 71683992

Thank you very much for your help,
Really appreciate it.

Best Regards,
Hendro



Offline cavagnaro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #3 on: May 27, 2008, 05:22:24 PM »
The 9 is a professional trunk group seizure or ARS?
Which Public Network category has the SIP trunk group?
Is the Genesys SIP Server address as trusted host?
Can you also post a successful outgoing call logs so I can compare them? For what I can't see is the OXE that is forbidding the outgoing call, check the entities and the connection category.
Is authentication on?

Offline hendro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #4 on: May 28, 2008, 03:48:55 AM »
Thanks Cavagnaro,

"9" is the DN Trunk type inside the genesys.
This DN number is 5000 with type = TRUNK. The TServer Annex :
contact = 172.9.2.95
oosp-transfer-enabled = true
password = 1234
prefix = 9
reuse-sdp-on-reinvite = true

The Public Network Category is : 31 (I saw it inside the trunk group category setup).
Yes, Genesys sip server is already a trusted inside the OXE SIP.
I will ask our PBX guys to check the entity and the authentication now.
I have attached the succesful log.


I have the DN with Extension type inside the genesys, with TServer Annex :
refer-enabled = false
reuse-sdp-on-reinvite = true
sip-hold-rfc3264 = true

So, last night when I set the "refer-enabled" to "True", I can do the call from the agent desktop, but unfortunately when we do call from several agents, then it become like a conference call, agents can hear the other agents conversation, also the other agents can hear other agents conversation.

I am sorry for this long question, I know I should go to genesys for it, but I still can't get the support.

Thank you very much,
Really appreciate your help

Best Regards,
Hendro

Offline cavagnaro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #5 on: May 28, 2008, 03:52:38 PM »
Please the OXE traces too...

Offline cavagnaro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #6 on: May 28, 2008, 08:05:34 PM »
Oh! It was inside the ZIP...sorry... ;D

Offline hendro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #7 on: May 29, 2008, 03:14:51 AM »
Thanks Cavagnaro,
Really appreciate your help,

Thank you,

Offline cavagnaro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #8 on: May 29, 2008, 03:51:03 PM »
The effect of others listening the same conversation is because all of them received the sip events as one user (Genesys) and was distributed to other agents....weird.
CAn you show how is the extension and agent are configured? (Annex Section)

Offline hendro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #9 on: May 30, 2008, 05:48:33 AM »
Thanks Cavagnaro,

This is our DN Extension Annex Configuration :
refer-enabled = false
reuse-sdp-on-reinvite = true
sip-hold-rfc3264 = true

And we have 1 DN Trunk Annex Configuration :
contact = 172.9.2.95
oosp-transfer-enabled = true
password = 1234
prefix = 9
reuse-sdp-on-reinvite = true

Do you thing we miss some of the setup ?

Best Regards,
Hendro


Offline hendro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #10 on: May 30, 2008, 06:02:02 AM »
Hi Cavagnaro,
I have tried to change the "refer-enabled" to "true" and set the "contact" of the DN to the client IP Address, and It works, but do you thing it's a right way to do it.

Thanks

Best Regards,
Hendro

Offline cavagnaro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #11 on: May 30, 2008, 10:47:45 PM »
Hi Hendro, sorry for the delay but have been very busy and little bit sick.

Here from Genesys documentation:

[quote]
Note: The following settings are recommended when configuring Genesys
SIP Phones:
• The refer-enabled option should be either absent or contain the
value true.
• The reinivte-requires-hold option should contain the value true.
• The transfer-complete-by-refer option should contain the value
false.
[/quote]

[quote]
Annex
TServer/contact
Contains the contact URI. This field specifies the end
point’s IP address, if this address is fixed. This object is
necessary only for stand-alone configuration, and only if
the end point does not register itself in the SIP Server
registrar. It is part of the persistent registrar feature.
For example, if an end point sends a REGISTER request to a
SIP Server and this request is accepted, SIP Server uses
the contact information from the REGISTER request and
updates (or creates) in Configuration Manager the option
contact in the TServer section of the Annex tab of the
corresponding DN object.
The URI format is:
[sip:][number@]hostport[;transport={tcp/udp}]
Where:
• sip: is an optional prefix.
• number is the DN number. This is currently ignored.
• hostport is a host:port pair, where host is either a
dotted IP address or a DNS-resolvable hostname for the
end point.
• transport=tcp or transport=udp is used to select the
network transport. The default value is udp.
[/quote]

So I would say yes, this is ok now :)

Offline hendro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #12 on: May 31, 2008, 07:49:49 AM »
Thanks Cavagnaro,
IT really a honour for us that you treat us like one of your customers,
Thank you, what a really nice support.

I will try it again next week, since I am also in the middle of the other project.
Hopefully, it's work, and we can go live, because we already 3 weeks late from the schedule.

Success for your jobs and hopefully you already recover now :)
Thanks,
Really appreciate it.

Best Regards,
Hendro ;D ;D ;D ;D ;D ;D

Offline cavagnaro

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Re: Gensys SIP Server with Alcatel OXE SIP Trunk
« Reply #13 on: May 31, 2008, 05:17:11 PM »
Please tell me did you check what allows that category 31?
And thanks for the comments but I only do what we all who support this forum do, help others as someone else also helped us once.  :D