[quote author=Seb Reeve link=topic=2301.msg8462#msg8462 date=1181895321]
Victor,
How are you using the technology - are you using StreamManager as an MCU/MOH/Recorder/AutoAttendant etc. How did you find the setup? Any major gotchas to getting the simple (call control/treatment) stuff working?
What gateway/endpoints are you using?
thanks!
Seb
[/quote]
Hi, Seb,
sorry for a belated reply. I was traveling in U.K. last week and did not get a chance to read it until now.
[u]How are we using SIP server:[/u]
First of all, regarding your question of how are we using SIP Server, we have several call centers that use it. The biggest one we have is for over 800 people situated in five different places using two SIP TServers 7.2. These five centers are connected together via a gigabit WAN, and use net.com ShoutIP VoIP gateways to connect to outside world.
The operators are using PC-based SIP phone with integrated Genesys interface that our company has developed using Microsoft RTC library. We are using Stream Manager for MOH and transfers. We tried and failed to use Stream Manager for monitoring because it would sometimes mute one of the parties for 10 second to a minute whenever we tried to start or stop monitoring.
We determined it to be due to RTC limitation and the fact that Genesys SIP Server is not fully compatible with RTC 1.3 as it is. So, we built in our own monitoring feature into all our SIP phones, which allows supervisors to monitor agents without the use of Stream Manager.
We are also using Quest Server (yes, the one that we are now offering for anyone outside Japan for free with the source code included) to display number of calls in queue both inside our SIP phone as well on large screens above.
We are using URS skill-based routing, and it is working fine, except for a problem we are having with sending calls between sites - we are not able to preserve ConnID and attach-data, which is a real bummer. We are currently trying to figure out what went wrong...
Instead of using NICE Recorder, we are also using own own VoIP Recorder integrated with Genesys to record all the conversations into mp3 files and make it available through the web interface with related attach-data. It does the same thing as Nice, but obviously is much cheaper.
[u]Regarding gettings things to work:[/u]
Getting SIP Server to work is easy. Getting it to work the way you need to is another story.
My biggest complaint with Genesys SIP server is that you really have to fine-tune your system based on the phones you are using. As you know, it is necessary to add Annex to each DN specifying its SIP signaling properties.
[attachimg=1]
After you figure out the combination of options for your particular endpoint you are set with Herculean (is it even a word) of setting it up! Imagine setting it for 700+ DNs - it will drive you nuts. There might be a way to do it all at once, and I just did not know it, but I frankly doubt it: you need to specify IP address per each DN and I did not see any official Genesys tool for it. I am really tempted to write a tool that would read from Excel sheet and add DNs directly into CME, so that I never have to waste my time like this ever again!
Stream Manager options are not that hard really except for the fact that you need to realize that sip-port in SIP Tserver should be set to 5060, sip-hold-rfc3264 needs to be set to true, and of course, last but not least, sip-enable-moh = true. Do not ask me why, but for some reason or other, I am in a habit of setting sm-port to 6669 event though sip-port in SM is set to 5061. I do not know why I do it, but I know that it is working when I do that.
Here is my TServer config file (remove .txt from the end):
[attachurl=2]
If you are planning on using HA (and all of you should) BEWARE! You will need NLB or Microsoft product, since Genesys SIP server does not really work without it. In SIP Server deployment manual, it says that for HA you "should" use network load balancer. What it really means is "must". You cannot have two SIP servers in HA mode and switch between them - it will not work. I really wish Genesys would fix it and fix it soon, because it adds at least 50,000 USD per installation.
Stream Manager configuration is pretty straight forward.
Mine is like this:
[attachurl=3]
Make sure that you define sip-port (in my case it is 5061).
Getting URS to work in SIP environment is not too hard (except for multisite attach-data). Your biggest headache will be treatments. If you want, I can write more about them too.
I hope it helped you. If you need more help - please do not hesitate to ask.
Best regards,
Vic
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