IVR Transcoding on CUBE

Victor Elizarov · ‎10-07-2014

Hello!

I've got CUBE on 2901 with IOS c2900-universalk9_npe-mz.SPA.154-3.M.bin.

SIP provider uses g711 alaw codec for SIP trunk.

I need to transcode incoming calls from g711a to g711u before playing IVR script.

I've configured transcoding profile on CUBE(in telephony-service, also I can use LTI as I know), but how can I enable transcoding on CUBE, so that it works in SRST mode too?

How many dial-peers do I need for solving this problem?

One incoming from provider with hard-coded g711a and another for IVR with g711u?

I know that transcoding turns on when one call leg is in one codec and second call leg is in another codec. How can I terminate incoming call from one to another dial-peer?

I tried with voice translation rules, but it doesn't work, router chooses incorrect DP or don't chooses outgoing call leg at all....

Sample of config:

application
service test flash:/ivrfax.vxml
!
voice translation-rule 3
rule 1 /308042/ /555/
!

voice translation-profile 308042_income
translate called 3

dial-peer voice 1 voip
description To_CUCM
destination-pattern [1-8]...
session protocol sipv2
session target ipv4:10.128.2.2
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 2 voip
description To_CUCM
destination-pattern [1-8]...
session protocol sipv2
session target ipv4:10.128.2.3
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 308042 voip
description ---Outgoing to SIP 308042---
translation-profile outgoing 308042
destination-pattern 9T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 30804242 voip
description from_Provider_incoming
translation-profile incoming 308042_income
session protocol sipv2
session target sip-server
incoming called-number 308042
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 555 voip
description IVR
service test
session protocol sipv2
incoming called-number 555
dtmf-relay rtp-nte
codec g711ulaw
no vad

Thanks for any help or information!

Jayanth Velkuri · ‎10-07-2014

it should work in SRST

What is your call flow and what is your current issue apart from xcoder

Victor Elizarov · ‎10-08-2014

My issue is problem with xcoding of incoming calls to IVR. I don't understand how can I do it using dial-peers or voice-translation rules. In debug ccsip mess it always chooses dial-peers 1 or (wrong) or there is not any outgoing DP when I change some settings in config. What I am missing for?

Victor Elizarov · ‎10-21-2014

So, can anybody help me?

I tried to only one one incoming dila-peer, but I don't hear anything in my audiofile via call, call establishes in g711a codec.

https://supportforums.cisco.com/discussion/11978696/transcode-incoming-calls-and-ivr

Chris Deren · ‎10-07-2014

Have you reviewed this for dial-peer matching understanding:

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html

Where is the call going to connect under SRST, will the IVR still be available?

You can use telephony-service (CME) and register xcoders to it, keep in mind that to transcode between u-law to a-law you will require universal transcoders and not traditional transcoders, so review config for universal transcoders.

Chris

Victor Elizarov · ‎10-08-2014

Yes, I read this document.

I've created ccm group, sccp local interface, dspfarm profile, and registered it to telephony-service. It's ok.

Yes, IVR still be availible under SRST, it runs as vxml script on router flash.

I configured router for inbound/outbound calls in g711a, and it works fine. But when using IVR with .wav file in g711u codec I've got a problem. I decided to use translation-profile on incoming dial-peer, from incoming number to internal IVR number, but it doesn't work.

I attached config of xcoder, dial-peers and translation rules.

In normal mode (without IVR) I receive calls in dial-peer voice 30804242 voip and after it translate them to dial-peers 1 or 2 to CUCM phones.