Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: optimizer on August 29, 2009, 08:07:14 AM
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This is great side, & Kudos to those who been helping in various questions.
I’m a newbie in Genesys, am implementing Stream Manager (7.6) with SIP TServer (7.6), & trying to play announcement using Stream Manager with setting in IRD, had search through all Stream Manager related post & have done steps as below:
1. Base on the 1st method show in http://www.sggu.com/smf/index.php/topic,2719.0.html, I’ve create a DN of type "Voice Over IP Service" in CME, but I can’t find this DN in IRD, thus can’t apply the routing strategy to this DN. I then turn to 2nd method.
2. Create a DN of type Voice over IP Service in CME, name is as "Treatment". Create in this DN's annex tab, a section called TServer and define the following options:
a. Contact = my stream manager IP & the listening port, in xxx.xxx.xxx.xxx:xxxx format
b. service-type = treatment
3. I already have a working routing strategy which will target available agents when caller call into a hotline.
a. I only need to play announcement (without digit collect), thus inside my IRD, routing -> selection icon -> busy tab, I select Play Announcement in treatment & Prompt tab, create a new line with ID 1.
b. There is a file named “1_msgsm.wav” in C:\GCTI\SIPSM\announcemen
When I save the routing strategy, it give me an error of
Warnings:
Type: CFGVoicePrompt name: 1 - not available
I’ve try putting in the full name (I know I need to only put in the index) it give the same error, has anyone encounter this error before?
Regards
Optimizer
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Hi Optimizer (great nickname ;)),
[quote]1. Base on the 1st method show in http://www.sggu.com/smf/index.php/topic,2719.0.html, I’ve create a DN of type "Voice Over IP Service" in CME, but I can’t find this DN in IRD, thus can’t apply the routing strategy to this DN. I then turn to 2nd method.[/quote]
DN of type "Voice over IP Service" is used to define source of VoIP services like treatment, recorder etc. It is used by SIP Server internally and has no connection to routing.
[quote]3. I already have a working routing strategy which will target available agents when caller call into a hotline.
a. I only need to play announcement (without digit collect), thus inside my IRD, routing -> selection icon -> busy tab, I select Play Announcement in treatment & Prompt tab, create a new line with ID 1.
b. There is a file named “1_msgsm.wav” in C:\GCTI\SIPSM\announcemen[/quote]
You configuration is correct. Just be sure that MS GSM codec is used otherwise Stream Manager won't be able to play the announcement.
[quote]When I save the routing strategy, it give me an error of
Warnings:
Type: CFGVoicePrompt name: 1 - not available[/quote]
As you wrote it is "just" a warning not an error. Your strategy will work even with such warning. If you want to get rid of it you have to create new object of type "Voice Prompt" in Genesys Configuration (CME / Resources (or tenant name) / Voice Prompts).
R.
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Hi René,
1. Thanks for the info on DN of type "Voice over IP Service”
2. Setting under “codec” section in Stream Manager related to msgsm are as below:
sip-annc-codecs=pcmu,pcma,msgsm,g729,g723
sip-annc-transcode=pcmu,pcma,msgsm
sip-conf-codecs=pcmu,pcma,msgsm
sip-pcap-codecs=pcmu,pcma,msgsm,g729,g723,h261,h263,h264=108
sip-record-codecs=pcmu,pcma,msgsm
3. Setting under “TServer” related to treatment found in TServerSIP option
sip-treatments-enabled=true
4. create new object of type "Voice Prompt" in CME resolved the warning message Thank You. Yes.
Announcement still not playing, when calling the Routing Point, call keep ringing until an agent login, look like I still new to learn….
as for the nickname…I’m still trying to be one….
Optimizer
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Hi,
If you post here logs of SIP Server, URS and Stream Manager (debug level) I can check what is wrong for you...
R.
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Thanks Rene,
The Routing Point DN is 1900, attached file include export of 1900 routing strategy, export option & log of SIPSM, TServerSIP & UniversalRoutingServer.
Host name is CMCVirtual, IP: 192.168.1.254
Another IP 192.168.1.254 is for AudioCodes.
The log show some errors which I do not know what it means, hope this will not take up too much of your time.
Appreciated
Optimizer
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Hi,
There is some issue with the attachment - size 0 bytes :( Could you try to upload it again?
R.
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Noted, see if this one ok..
Regards
Optimizer
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Based on very quick analysis of provided logs:
- DN "Treatment" - missing key "contact" in the section "TServer" on Annex tab - result: SIP Server is unable to use this VoIP service
[code][108] dn = 'Treatment' type = VoIPService xtype = Other cfgtype = 29 service-type = 'treatment' reg-mode = 0x1 = +force[/code]
- DN 1900 (Routing Point) - no strategy is loaded on this RP
[code]19:15:29.276_T_I_007001b954fd1001 [14:0c] EventRouteRequest is received for tserver TServerSIP[SwitchSIPCS] (this dn=1900)
_T_I_007001b954fd1001 [14:09] add DN TServerSIP 1900 <1900@SwitchSIPCS> (CDN 71 007001b954fd1001) to the call 1-01ad4c30 truly:11
_T_I_007001b954fd1001 [14:08] check delayed proc
19:15:29.276 Int 20001 interaction 007001b954fd1001 is started
_I_W_007001b954fd1001 [14:03] start without strategy -> default/ignore
19:15:29.276 Int 20003 interaction 007001b954fd1001 is routed to default[/code]
R.
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Ops.. I release the routing strategy to do some checking just now & capture the log, I’ll try your suggestion tomorrow in the actual server.
In the TServerSiP -> Annex -> TServer -> contact section belong to TSevrerSIP & I’ll put in host port info for SIPSM IP & port, is that right, base on my understanding referring to Framework 7.6 SIP Server Deployment Guide.
I’ll test I update tomorrow….
Few day of going nowhere on my side & for just 5min for you to spot the possible error…… hat off… 8)
Regards
Optimizer
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[quote]In the TServerSiP -> Annex -> TServer -> contact section belong to TSevrerSIP & I’ll put in host port info for SIPSM IP & port, is that right, base on my understanding referring to Framework 7.6 SIP Server Deployment Guide.[/quote]
That configuration isn't correct. You have to create TServer section with "Contact" key on Annex tab of "Treatment" DN object! Not on TServerSIP object.
R.
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Hi Rene,
With the routing strategy loaded & Annex -> TServer -> contact set to SIPSM IP:Port (192.168.1.245:5508)
The announcement still not playing.
I browse through the log & do see the routing server activate the play announcement task, but it didn’t play the announcement & the phone keep ringing.
_I_I_007001b96bda1001 [01:07] treatment added: type=PlayAnnouncement, param1=PROMPT.1.INTERRUPTABLE:1|PROMPT.1.ID:1|, param2=, tout=-1
I do notice there’re error on the later part of the log:
21:16:17.961_T_E_ [14:0c] EventError is received for tserver TServerSIP[SwitchSIPCS] - Invalid Destination DN
21:16:17.961_A_E_007001b96bda1001 [14:32] <-----------ERROR
_T_W_007001b96bda1001 [0E:0f] emergency: delete call due to this error
_I_I_007001b96bda1001 [01:06] treatment deleted: type=PlayAnnouncement, param1=PROMPT.1.INTERRUPTABLE:1|PROMPT.1.ID:1|, param2=##3511##, tout=-1
21:16:17.961_I_I_007001b96bda1001 [01:08] call deleting truly
know has been tourble you a lot by now...
Thanks again
Optimizer
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Hi,
URS won't even try to play the announcement as routing time isn't set on Selection object. Please open your strategy, open properties of Selection block and set some timeout on "Target Selection" tab (item "Timeout").
R.
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Hi Rene,
Had set some timeout to 10sec, still nothing play out, wonder can announcement be actually play out on a virtual server image…
Had trouble you enough, will try to lost some more sleep & see can find anything on the web.
Appreciated on your help a lot & hope can ask you again in the future.
Optimizer
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Post here log and I will check it again. It isn't worth to waste time on log analysis ;)
[quote]wonder can announcement be actually play out on a virtual server image…[/quote]
Really depends on machine configuration, performance of your computer etc. I'm using virtual machine as well and SIP Server+Stream Manager works even the voice quality isn't very good.
R.
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Hi Rene,
You’re right, I do a setup to a physical server, announcement still not playing
I did a test with setting announcement in the Selection icon, also with a separate announcement icon, announcement do not play on both scenario, I’ve uploaded the log with ConnID, will need to trouble you again….
Wav file that I’m target to play also inside the upload.
Regards
Optimizer.
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Forgot to put in the CONNID:
007001b983311004 Single icon
007001b983311005 separate announcement icon
Regards
Optimizer
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Hi Optimizer,
I don't know why you've changed parameters of VoIP service "Treatment" but it is configured incorrectly now. Please open properties of DN "Treatment", go to Annex tab to section "TServer", delete all existing keys and add only the ones below:
contact = "sip:CMCVirtual:5070" <-- you have to specify SIP port of Stream Manager here which is 5070 and not 5508
service-type = "treatment"
In your strategy you have to use "Play Treatment" object instead of "Music".
R.
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Hi Rene,
This may be the culprit, I’ve been using 5508 as the listening port, as show in the CME for SIPSM.
I’ll test it tomorrow & update the result tomorrow.
Can you share how to determine the Stream Manager port?
Don’t worry about using Music in the routing strategy, I’m just playing around…..
Regards
Optimizer
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Hi Rene,
Port 5070 works!!! tear almost flow down when we 1st hear the simple 1 word announcement over here....
You really are The Best, can you share how to determine the SM port.
Also, both format “sip:host:port” & “host:port” work for the contact setting, is there any advantage of using the sip:host:port format?
How many year of logs reading to make us as knowledgeable as you in Genesys….25?
Optimizer
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Go to the SM application on CME and see options tab and contact section you will see the port defined over there.
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Got it !!!
Thus port 5508 define in CME -> Server Info is for internal Genesys communication use, while 5070 is for SIP base communication?
Regards
Optimizer
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Hi Optimizer,
Yes, you're right. Stream Manager didn't support SIP protocol originally and used Genesys proprietary protocol for communication. Support for SIP protocol was added later and new option "sip-port" in "contact" section was introduced. So port specified on "Server Info" tab supports communication using original Genesys proprietary protocol and port specified in the option "sip-port" supports communication using SIP protocol.
R.
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RIGHT!!! now it make sense....if I continue like this, I'll change my nic to optiwiser.....or optiwhiter after growing more white hairs....
REALLY APPRECIATE Rene & Cavagnaro, with all your help, we're corrected this issue & move on on our SIT.
It might take 25 years for me to be at your level & start helping other, but I'm sure you'll see optiseeker asking more questions from now..
Regards,
opti...whatever...
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lol, I bet you can do it in less time, I'm 28 so I doubt that when I was 3 years old my mom had any idea of what Genesys was... ;D
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Hey, if your mom let you learn Genesys at the age of 3, now you’re the founder of the new Call Center Solution name Cavanesys…
Ok I should stop the joke in this threat and get back to work, my boss is reading this threat as well…. :P
But really glad to found this great website, to learn & joke together….
to the infinity…..
opti-joker
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Hum...we should put a banner: BOSSES NOT ALLOWED.
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My gosh, I did not realize how fast was this thread moving!
Here I was preparing a screenshot go-through manual for him and I wake up today and what do I see? Some guy from Brasil and René resolve all the problems on a fly. All while I was sleeping and dreaming how people would be thankful to me for being so helpful and knowing.
Dammit!
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Hi Victor,
This mean we’re really working hard & our bosses should see these too…
I believe there’ll be plenty of opportunities to thank you in the near future.
So, Thank you in advance first
Optimiser
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lol, a pity mercy thanks lol! Poor Vic! here boy! a thanks to you too...come boy...come....lol