transcode incoming calls and IVR

[nmdc.kzstan]

Hi,

Transcoding is working fine for incoming SIP calls. ITSP supports only g711alaw so it is hardcoded on incoming dial-peer.

How do I transcode incoming g711alaw calls to g711ulaw (its-CISCO.2.0.3.0.tcl) IVR? For transcoder to be invoked incoming and outgoing

dial-peers are needed, but in IVR calls there is only incoming dial-peer.

Now if I call external number from PSTN IVR I don't hear welcome message and after some time forwarded to operator number.

TIA

[islam.kamal]

Hi

1- You have to enable transcoder .

2- under the dial-peer which go to IVR , use codec g711ulaw.

3-match transcodig under your telephony-service.

Thank you

Please rate all useful information

[islam.kamal]

Hi

To enable transcoding , please find the below:-

voice-card 0

dspfarm

dsp services dspfarm

sccp local (interface)    / preferred voice interface/

sccp ccm x.x.x.x x.x.x.x   identifier 1 version ?   /where the ip address of your CME/

sccp

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register  hwxcode

!

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 10

associate application SCCP

no shut

Thank you

please rate all useful information

[islam.kamal]

Hi

Under

telephony-service

sdspfarm units 3

sdspfarm transcode sessions 4

sdspfarm tag 1 hwxcode                            /must match the name above /

Thank you

Please rate all useful information

[nmdc.kzstan]

Hi,

The problem is there is no dial peer to IVR, only incoming dial-peer 210. I have working transcode config and if call is

forwarded to CUCM via SIP trunk call is transcoded. 

application

service ivr flash:/its-CISCO.2.0.3.0.tcl

  paramspace english index 0

  paramspace english language en

  paramspace english location flash:/

  param aa-pilot 295

  paramspace english prefix en

  param operator 217

dial-peer voice 210 voip

description SIP_IN

translation-profile incoming SIP_IN

service ivr

session protocol sipv2

session target ipv4:10.0.0.12

incoming called-number 112233445566

dtmf-relay rtp-nte

codec g711alaw

no vad

[islam.kamal · Accepted solution]

Hi

I know the issue , did you try the below config?.

voice class-codec 1

code preference 1 g711alaw

codec preference 2 g711ulaw

dial-peer voice 210 voip

no codec g711alaw

voice class-codec 1

Thank you

please rate all useful information

[nmdc.kzstan]

Thank you, it works now.

[bkwon]

i have similar issue, but my case is dtmf.

incoming SIP call using in-band dtmf but IVR using rtp-nte

dial-peer voice 4 voip
 service meetme  -----> IVR
 session protocol sipv2
 incoming called-number 3666
 dtmf-relay rtp-nte sip-kpml
 codec g711ulaw

transcoding is up and running, but dtmf is not working.

ivr-172-28-135-3#sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
        IPv4 Address: 172.28.135.3
        Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 172.28.135.3, Port Number: 2000
Priority: N/A, Version: 4.0, Identifier: 1
Trustpoint: N/A

Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 172.28.135.3, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 68, Reported Max OOS Streams: 0
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30