VXML-API

Enhancing Gateway capabilities

2013-05-10T07:05:04Z

[color=#525252]We would like to have the following capabilities included for CVP TCL scripts:[/color] [color=#525252] [/color] [color=#525252]1) Create a new SERVGUID which should be unique for ever. Pass the SERVGUID with existing GUIID until the call ends (standalone, VRU only and comprehensive). We would like to set this as a new SIP header and retrieve in CVP/ICM. Which are the TCL and VXML scripts (bootstrap.tcl, bootstrap.vxml) that need changes?[/color] [color=#525252] [/color] [color=#525252]2) When the G/W is unable to reach CVP Call Server, it should use some means to store that information. It can even be in a text file or send a SNMP message. [/color] [color=#525252] [/color] [color=#525252]3) What we also wanted to do is identify whether there are any call rejects in signaling to find out whether MSC/Service Provider wanted to give any calls to the G/W but G/W rejected it for want of bearer channels. [/color] [color=#525252] [/color] [color=#525252]Thanks![/color] [color=#525252]-Sethu[/color]

RE: Max call optimization

2013-04-30T16:19:40Z

Hi Adolfo, Have you tried checking the memory and threads on the app server? Recently we were able to resolve a couple of scaling up issues by tweaking config on the Tomcat server that we were using. Regards Anupam IBM Research

RE: Max call optimization

2013-04-30T16:07:47Z

Thanks for your reply. I wasn't until recently that this subject came up again. I am trying to increase the amount of concurrent calls, but at some point in 300 calls it stops playing some prompts. I have used the following: "ivr prompt memory" "vxml memory tree" I have adjusted to see how can I perform better, without positive results Also, I am running on default memory iomem, is there a setting that would help me do better? There is very little information about these commands... Thanks again! Adolfo

Re: New Message from Feroz Syed in Voice Gateway API (VGAPI) - VXML-API: RE

2013-04-28T17:10:07Z

Feroz, Maybe you could save the RTP packets somehow. This link may help: https://supportforums.cisco.com/thread/2150112 Regards Anupam From: Cisco Developer Community Forums To: "cdicuser@developer.cisco.com" Date: 04/27/2013 07:58 PM Subject: New Message from Feroz Syed in Voice Gateway API (VGAPI) - VXML-API: RE: Whole Call Recording Feroz Syed has created a new message in the forum "VXML-API": -------------------------------------------------------------- Thanks Anupam! We don't have speach recog. I'm looking for CISCO solution as we are on CVP 8.5 platform. I went through some documents and found something related to Media Forking which supported by CISO Media Sense. I'm not sure if this servers the perpose that I'm looking for. If anyone could provide more on this , that would be great. http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/mediasense/853/srnd/cms853srnd.pdf -- To respond to this post, please click the following link: http://developer.cisco.com/web/vgapi/forums/-/message_boards/view_message/14722809 or simply reply to this email.

RE: Whole Call Recording

2013-04-27T14:29:36Z

Thanks Anupam! We don't have speach recog. I'm looking for CISCO solution as we are on CVP 8.5 platform. I went through some documents and found something related to Media Forking which supported by CISO Media Sense. I'm not sure if this servers the perpose that I'm looking for. If anyone could provide more on this , that would be great. [url=http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/mediasense/853/srnd/cms853srnd.pdf]http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/mediasense/853/srnd/cms853srnd.pdf[/url]

RE: Whole Call Recording

2013-04-26T17:29:09Z

I think 'whole call recording' is supported on Nuance (if you are using that for speech recog) Regards Anupam

RE: Error no resource when using TTS

2013-04-26T17:22:30Z

I'd turn on some logs on Cisco router to see whats the issue. Also, if you dont see anything in the packet trace then no other VXML and TCL application is trying to get the TTS resource (otherwise you would see 'their' SIP INVITE atleast) Also, if you dont see the SIP INVITE of the current TTS request also, then probably its nothing to do with the Nuance Server or the licenses, since the request is not even reaching the Nuance server. Its something to do with the MRCP client on Cisco. Its not sending the request out and giving an error - turning on logs may help.

RE: Whole Call Recording

2013-04-26T16:28:54Z

[color=#525252]Hi,[/color] [color=#525252]I knew this is very old post now. I'm also looking for this functionality and wanted to check if this feature added/available in any of the latest version of IOS?[/color] [color=#525252]Regards,[/color] [color=#525252]Feroz Syed[/color]

RE: Whole Call Recording

2013-04-26T16:28:06Z

RE: Whole Call Recording

RE: Error no resource when using TTS

2013-04-26T09:26:45Z

Hello, Yes, I've done a packet trace, but no SIP INVITE is sent (by Cisco) and received (by Nuance server) when the VoiceXML is trying to get TTS resources. We currently don't have TAC support anymore, so that's not an option. I haven't looked at the License Manager when the problem occurs, so I'll do that next. Is there a possibility the other VoiceXML and TCL applications are trying to get TTS resources eventhough the com.cisco.tts-server property is not configured explicitly in those scripts and not configured in the CLI? I would say no, because they wouldn't know it exists. Thanks, Grant

RE: Error no resource when using TTS

2013-04-25T15:29:26Z

Hi Grant, With my experience, its always good to take a Wireshark packet trace on the Nuance machine. That will tell you how many requests are really reaching the Nuance machine (they may be more than you think - for whatever reason). Also, it might be good to poll the Nuance License Manager while this problem is happening to see how many licenses are indeed available. (I know you mentioned that 503 is not being received but still the above is worth checking I guess) Thanks Anupam

RE: Error no resource when using TTS

2013-04-25T13:32:33Z

Hi Grant, i dont think there is an issue with VXML, could you please contact TAC they might help you. Thanks, Raghavendra

Error no resource when using TTS

2013-04-25T12:45:59Z

Hello, I've configured our Cisco VoiceXML Gateway to communicate with the Nuance Speech Server for TTS resources. The com.cisco.tts-server property has been configured explicitly in the VoiceXML document, nowhere else, so only this application makes use of the TTS resource. I'm noticing very strange behavior where the TTS functionality occasionally works, but most of the time I'm receiving an error noresource. See below. We are still using the evaluation version of Nuance Vocalizer for Network 5 and we have a demo license for 4 ports. The machine I tested this on is a live machine, but I temporarily routed the traffic away to another machine. Once I started testing with 5 outbound calls, 4 of them successfully received TTS resources and the 5th naturally didn't get it since the license didn't allow this. And the Nuance Speech Server also returned 503 Service Unavailable for the 5th call. All subsequent tests were successful. But as soon as I let the other traffic pass through this machine as well and try testing the TTS most of the time the TTS doesn't work. In this case I'm also not receiving a 503. We are running several VoiceXML applications and like 20 instances of a TCL application which polls for calls to place (outbound calls). None of these applications are configured to use TTS and the global ivr tts-server configuration is not configured. Does anyone have an idea why this is happening? We are using a Cisco 3925 with IOS Version 15.2(3)T and MRCPv2 (SIP) for TTS resources. Regards, Grant Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: bargein=1 Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: timeout=0 Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: encoding_name=utf-8 Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: tts_server=sip:mresources@192.168.1.16 Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: media_logging_id= Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: ssml=error Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: xml:lang=en-US Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_tts: name=ssml:\\text882.vxml.77694 Apr 25 12:33:39.003: //-1//VXML:/vxml_vapp_tts: Exit Apr 25 12:33:39.003: //-1//VXML:/vxml_default_event_prompt: Exit Apr 25 12:33:39.003: //-1//AFW_:EE180EB428000:/vapp_session_exit_event_name: Exit Event error.noresource

RE: VXML SIP Transferaudio

2013-04-09T09:27:14Z

Thank you Raghavendra the missing codec in the dial-peer was the problem. Regards Heimo

RE: VXML SIP Transferaudio

2013-04-09T09:24:01Z

Hi Heimo, from the logs it shows that play audio failed because of codec mismatch, please configure "codec g711ulaw" to your dial-peer 52. Apr 9 08:54:20.737: //34//MSM :/ms_asDone_buginf: Stream Association Failed: Requested codec=0x5=g711ulaw, Negotiated codec=0x10=g729r8 Thanks, Raghavendra

RE: VXML SIP Transferaudio

2013-04-09T09:02:34Z

Hi Raghavendra, you can find the log in the attachement. Thanks =)

RE: VXML SIP Transferaudio

2013-04-09T09:02:07Z

Hi Heimo, please try to configure "codec g711ulaw" to your dial-peer 52.audio files also should be same codec. Thanks, Raghavendra

RE: VXML SIP Transferaudio

2013-04-09T08:49:13Z

Hi Heimo, could you please send us the logs by enabling below debugs. debug voip app Thanks, Raghavendra

RE: VXML SIP Transferaudio

2013-04-09T07:59:07Z

[b][b][b]The pots dial peer has the same configuration: dial-peer voice 52 voip description "SIP Test" service tvmsip incoming called-number 4321T The sip dial-peer: voice class codec 10 codec preference 1 g711ulaw codec preference 2 g729r8 codec preference 3 g711alaw dial-peer voice 51 voip destination-pattern [1-8]T voice-class codec 10 session protocol sipv2 session target ipv4: dtmf-relay rtp-nte[/b][/b][/b]

RE: VXML SIP Transferaudio

2013-04-08T16:11:24Z

Can you plrase attached the dial-peer configuration for this vxml service ? Thanks !