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:
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.
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
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?
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.
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.
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
You configuration is correct. Just be sure that MS GSM codec is used otherwise Stream Manager won't be able to play the announcement.
When I save the routing strategy, it give me an error of
Warnings:
Type: CFGVoicePrompt name: 1 - not available
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).
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.
DN 1900 (Routing Point) - no strategy is loaded on this RP
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
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)
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.
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.
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.
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”).
Post here log and I will check it again. It isn’t worth to waste time on log analysis
wonder can announcement be actually play out on a virtual server image…
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.
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.
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”.
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?
Could you try to upload it again?