Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: Seb Reeve on June 13, 2007, 09:51:56 AM
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Hi everyone,
I am interested in deploying Genesys SIP Server as a core part of a customer deliverable and as part of my analysis am interested to get a feel for who/how this technology is being used in the real-world.
If you have a moment - can you please select from one of the options in the poll so we can all get a view of how this particular technology is being used by everyone.
Many thanks!!
Seb
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Seb - you forgot the poll :)
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What you mean Victor? I see the poll...
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[quote author=victor link=topic=2301.msg8427#msg8427 date=1181815040]
Seb - you forgot the poll :)
[/quote]
wow.... Yes, I see it now too!!! Weird :) Just to make sure that enough people answer it, I have set it to be at the top of the forum until Monday.
We are using SIP server on quite a few sites, and it seems to be pretty good, except for a few quirks working with RTC and the fact that when using HA, you need to use Windows Cluster or Network Load Balancer. I think Genesys really needs to address this short-coming and fast!
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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
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Hi Seb,
Currently I'am working on 2 projects with SIP Server 7.5. We choose Audiocodes GW - Mediant1000 for simple outbound Call Center, and Mediant 2000 as we need SS7 signaling. For endpoints I'am still experimenting with different hardphones form Genesys supported hardware list.
For the basic setup if You follow documentation for 7.5 there shouldn't be any problems - for treatments and moh.
Problems starts when You begin transfering calls, making conferences and some multi site routing.
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Pawel,
Thanks for the input - useful and backs up our lab experiences - seems all is OK with simple callflows but gets difficult when trying MCU and Transfers etc.
Did you manage to get this stuff working - or does that still elude you?!
Cheers!
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Transfers are working ok but I need to change option in URS, route_consult_call to true. By default it has value false. I will be testing MCU next week - I will post results.
Paul
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As for endpoints - we will be using Polycom 501 - You can do some interesting thing with it - like deciding in strategy what ringtone agent will hear or wheter call will be answered automaticly. Also TAnswer method works with Polycom endpoint.
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[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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Hi Vic,
I was wondering why You didn't use internal-registrar-persistent for generating contact field based on register message ? - were there some problems with it ?
When writing own SIP client - did You based in on Genesys Endpoint block or You started from scrach ?
Could You write a little more about treatment headache ? - the only thing I found is that with some GW sm changes codes during playing treatments and sound is disorted after such change
Paul
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Plus to what Vic states above, afaik, fully Microsoft .Net Framework-compatible RTC (versioned as 1.4) will be released by October 2007, which is expected to solve many issues such as VAS, or SIP/2.0-related session problems.
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[quote author=bublepaw link=topic=2301.msg8653#msg8653 date=1183441817]
Hi Vic,
I was wondering why You didn't use internal-registrar-persistent for generating contact field based on register message ? - were there some problems with it ?
When writing own SIP client - did You based in on Genesys Endpoint block or You started from scrach ?
Could You write a little more about treatment headache ? - the only thing I found is that with some GW sm changes codes during playing treatments and sound is disorted after such change
Paul
[/quote]
Hi, Paul,
to be frank - I did not know about it. I just set it to true, since without it all my non-CTI phones were useless.
I will test and see what this internal-registrar-persistent do and get back to you, but I hope it will do just what I imagine it will and remove the need for contact field!
I will write about treatment tomorrow!
Best regards,
Vic
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Vic,
It worked for me :) but remeber to change security settings for SYSTEM account on SIP switch object - by default SYSTEM has read-only access - but when this option is set to true SYSTEM accounts needs write access to modify ANNEX tab for DN's.
Paul
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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.
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[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
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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...
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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 :)
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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
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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.
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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.
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Hi René,
Indeed, I was missing that idea. I applied the same idea from OXE to SIP Server and worked fine.
Thanks!