Creating announcement in Stream Manager & hit error

[optimizer]

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

[René]

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.

[.....]

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

[René]

Hi,

If you post here logs of SIP Server, URS and Stream Manager (debug level) I can check what is wrong for you...

R.

[optimizer]

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

[René]

Hi,

There is some issue with the attachment - size 0 bytes  Could you try to upload it again?

R.

[optimizer]

Noted, see if this one ok..

Regards

Optimizer

[René]

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.

[optimizer]

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…

Regards

Optimizer

[René]

[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.

[optimizer]

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

[René]

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.

[optimizer]

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

[René]

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.

[optimizer]

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.