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Mr. Ankush Khare
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Mr. Ankush Khare
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RE: Genesys Nuance Integration TTS
Posted By
Ankush Khare
03-13-2020 00:39
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Hi All, The issue was fixed finally. I used Play Application block and passed the variable holding my text from workflow to callflow. Used below code: var input = new.Object(); input.vartextmsg = _data.textmsg; _genesys.ixn.setUdata(input); Thanks Ankush ------------------------------ ...
RE: Genesys Nuance Integration TTS
Posted By
Ankush Khare
03-09-2020 00:44
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Hi Ferri, I checked the Composer Help guide for Play Message Block and it says : Play Message Block Use to invoke/play audio or text-to-speech Announcement treatments. As described in the Genesys Voice Platform Deployment Guide. GVP supports automatic speech recognition (ASR) and speech synthesis ...
Genesys Nuance Integration TTS
Posted By
Ankush Khare
03-07-2020 07:22
Found In
Library:
PureEngage On-Premises
#Integrations #SIP/VolP
Genesys Nuance Integration TTS
Posted By
Ankush Khare
03-07-2020 07:22
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Hi All, I have a setup with Genesys Nuance integration. I have a composer strategy with couple of WebRequest block and at last I get below value in a variable: textmsg = 'Geht es bei Ihrer Frage um ein Thema zu dieser Mobilfunk-Nummer?' At the end of workflow I want to play this message as TTS ...
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-30-2020 23:28
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Hi Santosh, Thanks will try the same. :) ------------------------------ [Ankush] [Khare] ------------------------------
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-29-2020 04:55
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Hi Siva, Thanks for the response. Yes the problem is ACK is being sent to private IP of SIP server i.e. 172.31.17.148 instead of sending it to Public IP of SIP server which is on AWS. I am just trying to understand if there is a possibility to force SIP server to insert the Record-Route header with ...
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-28-2020 22:26
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Yeah I have opened the ticket :) Thanks for all your help ------------------------------ [Ankush] [Khare] ------------------------------
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-28-2020 13:35
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Below is the config on my TRUNK: [TServer] contact=vf-test.pstn.us1.twilio.com cpn=+17162355059 dual-dialog-enabled=true make-call-rfc3725-flow=1 prefix=00 refer-enabled=false replace-prefix=+ sip-enable-100rel=false ------------------------------ [Ankush] [Khare] ----------------- ...
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-28-2020 12:36
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18:46:00.960: SIPTR: Received [0,UDP] 938 bytes from 54.172.60.3:5060 ;tag=E040053C-9368-4B3C-965C-2D69914D5C7E-5 To: ;tag=78445343_6772d868_eacf0ee7-dce3-4709-99db-d4eb36610640 Via: SIP/2.0/UDP 172.31.17.148:5060;rport=5060;received=3.8.78.209;branch=z9hG4bK43B6382A-405F-43DE-B940-30EF404971AC-1 ...
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-28-2020 12:26
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Hi Ivan, Thanks for the explanation :) . But as I understand the ACK is sent directly to the Contact Address in 200OK. This is how ACK is formed right? Isn't the same thing happening. Can it be because I don't have SBC in between and I am talking directly to SIP trunk provider. I am just trying ...
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-28-2020 02:34
Found In
Library:
PureEngage On-Premises
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-28-2020 02:34
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Softphone logs : Call-ID: FCCEA55C-DCBD-4B0D-BEE1-8A619D4FD5F6-3@O1fW ------------------------------ [Ankush] [Khare] ------------------------------
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-28-2020 02:32
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Softphone logs attached: Call-ID: FCCEA55C-DCBD-4B0D-BEE1-8A619D4FD5F6-3@O1fW ------------------------------ [Ankush] [Khare] ------------------------------
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-27-2020 23:58
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The log is from SIP server and Wirshark traces are from the host where softphone is running. I will try to find out matching Softphone logs. ------------------------------ [Ankush] [Khare] ------------------------------
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-27-2020 23:32
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Thank you for the reply guys. I have tried with 4 different soft phone and every time it's same. I am not sure how to troubleshoot a problem where Soft phone is not sending ACK. See attached traces wireshark and SIP logs. ------------------------------ [Ankush] [Khare] ----------------------- ...
RE: Call disconnects in 32 sec.
Posted By
Ankush Khare
01-27-2020 23:31
Found In
Library:
PureEngage On-Premises
Call disconnects in 32 sec.
Posted By
Ankush Khare
01-27-2020 06:49
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Hi All, I have a scenario where SIP server is directly integrated with Twilio provider. The outbound call is always disconnected in 32 seconds. When i checked the SIP server logs I can see I am getting 200OK message and after that ACK is sent from SIP server and then immediate BYE. The provider ...
Call disconnects in 32 sec.
Posted By
Ankush Khare
01-27-2020 06:48
Found In
Library:
PureEngage On-Premises
#SIP/VolP #Telephony
Platform SDK - Who changed the config object
Posted By
Ankush Khare
01-09-2019 00:08
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Hi All, We are looking for a config server event which could tell us about the client details from which changes has been made to the config object. We have already subscribed to the config events for the type of change i.e. create/update/delete and what has been changed but we are struggling to ...
RE: Genesys Platform SDK
Posted By
Ankush Khare
11-16-2018 04:16
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Hi Jim, Thanks a lot for your reply. I went through the .xml file in templates folder and parsing the file to get the Options for any application template is pretty much available. But i couldn't find the info regarding the name , type and version in the xml file , which we usually set by importing ...
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