Genesys SIP Server

[victor]

[quote author=Sylvainsjc link=topic=2301.msg8690#msg8690 date=1183761259]
Hi all from france,
We are deploying Sip Server 7.5 with SoftSwitch Asterisk in front (PSTN gateways and softphones are connected to asterisk).
Genesys SipServer and Stream Manager are running on linux redhat.
The first tests are good (skill routing, custom desktop, ccpulse, etc)
I'll keep you in touch.
[/quote]


Hi!

Can you please tell me the reasoning behind having Asterisk on top of SIP server?

thank you!
Vic

[Sylvainsjc]

Yes, we want the audiocodes gateway to be registered on Asterisk and not on Genesys Sip Server to be able to distribute as we want the free PSTN numbers.
You can also imagine some external agencies registered on Asterisk and a central Genesys Call Center Agent with an agent desktop that tell him that employee x localized in distant agency is "present".
Result : No PBX licenses, no Genesys licenses for the agencies...

[cavagnaro]

Guys one question,
Do I have to register the RP also on Asterisk? I mean, create an extension for this object and then create a RP on CME on SIP Switch DNs and configure it to be also monitored? How does this part works?
Also Vic, can you please complete the idea of the problems you found on treatments??

Thanks 

[René]

Hi Cav,

There is new document - Framework 7.6 SIP Server Integration Reference Manual - available on Genesys TechSupport describing integration between SIP Server and Asterisk. That documents is related to SIP Server 7.6 but the principles are the same even for release 7.5.

René

[cavagnaro]

Hi René,
Thanks for answering, I don't know why this post was marked as already read when never saw your answer...
I have read the mentioned document and it is basically a resume of all the other white papers available, nothing new. It doesn't mention how to involve URS on the call flow...I'm stuck at that point right now.

[René]

Hi Cavagnaro,

I think you're confused by the difference between agent DNs and routing point DNs.

In case of agent DNs the SIP endpoints are registered on Asterisk side and created as object of type "Extension" in Genesys Configuration. SIP Server issues SUBSCRIBE message for these and Asterisk updates SIP Server with status of these objects using NOTIFY messages then.

On the other hand, routing points are configured as SIP endpoints (=RP in CME) on SIP Server. On Asterisk side, there is a dial plan only that specifies what calls have to be transferred to SIP Server and to what number (=RP on SIP Server).

Please read the chapter "Call Flows" on page 60 of reference guide again and I'm sure you'll notice the difference now. Especially compare SIP message on figures 45 and 46 with these on figure 47.

Hope it helps you
R.

[cavagnaro]

Hi René,
Indeed, I was missing that idea. I applied the same idea from OXE to SIP Server and worked fine.
Thanks!