Is anyone out there done the network sip server implementation?
What kind of soft switch do you use and how is it ?
Thanks in advance…
You would just use Genesys SIP Server. Then connect your agents through a MPLS or like network, so it doesn’t matter where your agents are or your servers, as long as you have a data feed between them.
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What kind of soft switch do you use? Asterisk?
How about access code to get to soft switch, what do you use?
Asterisk will work, I have an implementation guide for Asterisk and Genesys if your interested. Just provide me with an email and I will send it over.
Far as the access code, your current switch needs to be able to support SIP. So if you had an Avaya, you would need AES. Alternatively, you could route the call to a new 800 # that bridged your non SIP switch to the SIP.
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Hi, CTIGem,
we have been rolling out one of the largest Genesys SIP installations (1000+ seats) using both HA, multi-site and DR. We are using Genesys SIP version 7.5 with HA, URS and CCA. Everything was done using Genesys (except for Windows Cluster Server for HA).
I have never heard of Network SIP Server other than NexIPT that NTT Comware is offering as ASP. I would guess that depending on your layout a normal Genesys SIP Server would be just more than plenty. What exactly are you trying to do?
Daimonas:
Regarding Asterisk, I like the idea and I have played with it on more than one occasion; however, given that there is no full time support except for online user group, lack of ability to ask for SR I would advise against its rollout as something to be used by more than 20+ call center. It was not designed for that. With time, it will, but if people just need a basic VoIP infrastructure without the bells and whistles of Genesys, they might as well get NET.COM - at least that way they would be supported, right?
I tried connecting Genesys with Asterisk last year using JTAPI and writing an adapter server to act as a proxy imitating G3 protocols and got most of it working, but then some other things came up, so I have never bothered to play with it again.
Best regards,
Vic
Hi Vic, our sister company has built an Asterisk and is using Genesys SIP and Media Server in our lab, and it looks good so far. Last i heard he was working on getting a ring generated to the customer end when ringing a SIP phone. I will have him join the board and share his thoughts.
The Genesys Network SIP Server is used to scale beyond 2000 users. The Network SIP Server is designed to provide load balancing between N+1 SIP Servers as opposed to a HA configuration with a warm/hot standby. There is a document in the Genesys Document Library entitled Network SIP Server Deployment Guide, which explains in detail how to deploy. With that said, I have interfaced an Asterisk PBX with Genesys SIP Server and Stream Manager. To do so all you need is the following:
Genesys Configs:
1.) Genesys SIP server running
2.) In CME, in your SIP Switch DN, create a DN of type Trunk,
3.) In the annex tab create a section called TServer
4.) In the TServer section, create an option called contact and make it equal to asterisk@ i.e. asterisk @ 10.1.0.99
Asterisk Configuration:
I just set up a couple of DID’s to send calls to Genesys. So in my extensions_did.conf file I created the following:
[from-did]
exten=> 2287000,1,Dial(SIP/70000 @ 10.1.0.15:5065,20,rt)
exten=> 2287001,1,Dial(SIP/10000 @ 10.1.0.15:5065,20,rt)
so here we have:
exten=> ,1,Dial(SIP/ @ <SIP Server IP:SIP Port,20,rt)
I hope this helps.
Wow… this is something to definitely try!!!
BTW, can you please tell me the reason for using Asterisks? It sounds like something I might need to reconsider after all!
Daimonas: just post your doc here for all of us to see. No reason to mail it privately and then have twenty “me too’s” !!! 
Vic
Thanks for the info.
I think Network Tserver would be overkill for us.
What we want to do is route calls to other site with ISCC.
But other site might not have PBX switch and that’s where IP switch (Asterisk) comes into play.
Robb suggested having a brige and that might be a solution.
But we have Nortel.
So, is this mean I have to have some kind of gateway (possibly Asterisk with T1 trunk interface to CO)?
Origination Tserver send calls to some number (this would be access code) that’s hitting Asterisk gateway…?
Daimonas,
Yes, I’d like to your doc.
CTI Gem,
Does the other call center need to have a PBX? If not, you can try to just install SIP Softphones on the agent desktops and have them register to the remote SIP server. But to answer your question, Asterisk can be an inexpensive gateway if you have some available hardware.
I emailed the doc to Victor, did you get it? As I am unable to provide the doc due to my network security. Please upload it 
Hi, everybody,
last week was pretty hectic (I am in Thailand now after a visit to Japan and UK), so I apologize for the delay. I am looking at this document and thinking… wow… how does it affect Genesys SIP Server plan. This pretty much opens a myriad of new possibilities of rolling out SIP with the minimum set of Genesys tools.
Regarding the docs, I will hold off on posting stuff that says “Copyright Genesys”, so I am only adding Word Document that I assume you have obtained permission for posting from Phillip Raise (who is the guy?)
Document is very very very interesting to say the least. BUT, does Genesys support it?
Best regards,
Vic
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Yes, Genesys officially supports Asterisk.
It was on their web site one point.
Good catch victor on the copyright stuff, if anyone interested in more info you can contact your Genesys Account Rep and they can provide you with more information. Far as the white paper, I checked with my internal contact I received the document from and its free to publish (not sure who Phillip is, probally a tech writter )
New to the boards so just saw this posting - With Digiums acquisition of SwitchVox you can have a “business edition” of Asterisk which is fully supported. I believe the largest switchvox appliance supports 400 extensions - of course with clustering you could scale it as you want. I’ve deployed Asterisk in production environments as a stand alone PBX with much success. The SIP/Asterisk integration is new to me however. I have, with little effort, got it running in a lab environment but I have not heard of any prod deploys with this setup. I’ve seen Asterisk work with very few issues and of course Genesys, if deployed properly, is rock solid. So as long as the integration is tight, I think this could be a very good solution for mid-sized contact centers. Of course - now that Genesys has SIP - any media gateway will do. The role of the “switch” as we have known it, definitely is changing. Really all you need is a device to interface to the PSTN. For that matter - if you utilize a voip provider, you dont even need voice circuits - just data please! 
Probably preaching to the choir (hopefully) but I’ve been talking to every client who will listen the possibilities that the Genesys SIP server could bring them.
what is with this post that it is like honey to flies?? maybe some words on it???
(just curious…delete if necesary)
I have no idea but this is weird…
Gee, Is something wrong with me? ???
I really do not know why all spam bots target this particular post, but my natural guess would be: Tony. With Wyn helping a bit.