Get Digits ext script not working. DTMF issue with Audio Codes Mediant 1000.

[Ritesh Desai]

Hi team,

I have UCCE Lab setup and working on ICM Script Editor. I've basic call flow, Snap attached for reference. Am beginner in UCCE.

I tried 3 digit Menu, capturing it and route to agent. My basic call flow is working. My call hits Get Digit Ext Script element. Prompt plays to press 1,2 & 3. DTMF is not passing and doesnot moves to next node....

Nyone have experienced before please help... TIA

regards,

Ritesh Desai

[Chintan Gajjar]

Hi,

please share the CVP logs for the test calls.. what have you set input_type ECC variable?

share your VXML gateway config...

Chintan

[Ritesh Desai]

Hi Chintan,

Thanks for reverting on this case... 

In my test environment CUBE and VXML are co-located. CUBE IP (192.168.1.1) and VXML GTW Loopback interface (192.168.2.1).

I've attached debug ccsip all having errors but am unluckily am not finding cause. Prior this, I have tested variants of dtmf-relay rtp-nte and atlast dtmf-rtp sip-notify is successfull and accepting DTMF digits but fails post that.

I've found 2 errors;

1. debug ccsip all: "Requested DTMF-RELAY option(s) not found in Preferred DTMF-RELAY option list!"

2. debug ccsip media: "stun is disabled for stream: 308E55CC"

Im checking for this and found will post the reason for cause. I've uploaded debug logs, VXML GTW config screensnaps of GD microapp & ECC variable.

Thanks in Advance...

regards,

Ritesh Desai.

[Chintan Gajjar]

hi Ritesh,

please share the CVP call server logs for the test calls.

and since you have ingress and VXML gateway co resident, i dont see inbound and outbound dial-peer for sending calls to CVP..

Chintan

[Chintan Gajjar]

ok i can see, 

inbound dial-peer

dial-peer voice 1001 voip
 description Incoming from ACM1000
 session protocol sipv2
 session transport tcp
 incoming called-number 8896
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

now num exp converts 8896 to 3011

and outbound dial-peer for CVP call server is

dial-peer voice 1002 voip
 description CVP VXML Standalone application dial-peer for a VOIP call
 destination-pattern 3011
 session protocol sipv2
 session target ipv4:192.168.3.142
 session transport udp
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

and now you have VRU dial - peer

dial-peer voice 811111 voip
 description CVP IVR dial-peer
 service bootstrap
 session transport udp
 incoming called-number 8111T
 voice-class sip rel1xx disable
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

in all three dial-peer dtmf-rely is not consistent and atleast not matching on VRU dial-peer.

i would request you to update dtmf-rely on all above three dial-peer witht below and test.

dtmf-rely rtp-nte sip-notfy cisco-rtp

should be on all three above.

[Chintan Gajjar]

there is more..

looking at the invite coming from Acme, its comming with fax as dtmf-relay

v=0

o=AudiocodesGW 1570355095 1570354782 IN IP4 192.168.1.233

s=Phone-Call

c=IN IP4 192.168.1.233

t=0 0

m=audio 6300 RTP/AVP 0 8 96

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

a=ptime:20

a=sendrecv

a=rtcp:6301 IN IP4 192.168.1.233

look at 

a=rtpmap:96 telephone-event/8000

and your Gateway and inbound dial-peer is not at all set for that.

you have two way to come over this,

ask your service provider to send sip dtmf methods like sip kmpl or sip notify

ask your service provide to send dtmf with nte payload 101 and not 96.

third one which i dont suggest but should work i guess is

have your all dial peer set to dtmf-relay rtp-nte

and use the command rtp payload-type nte 96 to force your gateway to use 96 as payload and not 101.

[Ritesh Desai]

Hi Chintan,

I've followed all the actions as requested by you and still unlucky. Please find the attached logs of CVP call server and ccsip messages.

Thanks for extending help.

regards,

Ritesh Desai.

[Chintan Gajjar]

i see everything working on CVP and ICM script side, its just you are receiving DTMF from your ACME with payload 96 and which is not supported by Voice gateway.

can you talk to your service provider and ask them to send your the DTMF payload as either sip kmpl, sip notify or nte payload 101?

there are techniques, did you try the new dial-peers i gave later with dtmf payload type 96?

if yes, please have a look at 

http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_book/vb_book/vb_10022.html

if that also doesn't work then you have to talk to your Service provider and get it fixed from there side to supported version.

the reason i see your setup is working is have a phone registered to VG ot CUCM and try calling same app, it will work.

[Ritesh Desai]

Hi Chintan,

Thanks... I tell you my Lab infra.

ISP provides me E1 Line which is terminated on AudioCode gateway. AudioCodes gateway converts TDM to SIP and connects to Ingress Gateway. Yes i see transport is Fax dtmf relay in Audio Codes.

I need to check with senior for changes.

I'll revert you on Monday.

regards,

Ritesh Desai. 

[Chintan Gajjar]

you got it correctly, that is something you have to change on your device who is ending calls to your gateway. you can change it to either rtp-nte (payload 101)

,sip-kmpl or sip-notify...

[Ritesh Desai]

Hi Chintan,

Please find attached 2 images from Audio Codes config. I feel everything is proper. Will you please have a look on this...?

But Chintan when am going through Cisco docs, I found Cisco IP Phones SCCP based accepts out of band signalling. "dtmf relay rtp-nte" command is for inbound signalling. This will not work in my opinion.

Refer rtp_rtcp image, RFC 2833 Tx and Rx payload is set to 101. In CUBE, on all DP's i configured "rtp payload nte 101" nothing worked. Its not capturing digits and announcement playing please press 1, etc.

So, I think this will not work. Correct me if am wrong.

I configured "dtmf-relay sip-notify" (out of band signalling) on 2 DP's but on VRU DP 8111111111 am having only 3 options (cisco-rtp, rtp-nte and h245). All these are in-band signalling except h245 which is for h323 network. I've removed dtmf-relay rtp-nte config from 3 DP'S VRU, 929292 AND 919191.

Test made and found call is disconnecting hitting Get Digits element in Script Editor. On AudioCodes syslog I found error "DtmfCapNegotiationAlgorithm :: TxDtmfMethod = DTMF_NOT_SUPPORTED "

Thanks in advance.

Regards,

Ritesh Desai

[Chintan Gajjar · Accepted solution]

as far as i know most of the ip phone supports in-band and out of band dtmf..

does Audio codes supports sip dtmf specifications? like sip-kpml and sip-notify?

i am first time having come across Audio Codes but looking at below link:

http://www.audiocodes.com/filehandler.ashx?fileid=1618188

if your search for DTMF in the pdf, you will see many options..look at the page 29 you would see the DTMF payload type, you can set it as 101 and check if that works.

regards

Chintan

[Ritesh Desai]

Chintan,

I found it! I have option to set DTMF Transmit as INFO, NOTIFY and RFC 2833. Along with this, I've option to set RFC 2833 Payload Type: 96 (currently defined). I need to check with Senior approval for changes and come back on this soon. Thanks for support.

Regards,

Ritesh Desai

[Ritesh Desai]

Hi Chintan,

Yeah mhan... It's successfull.

As suggested by you, changes in AudioCodes gateway were done.

1. DTMF Tx = RFC 2833 (In-band signalling) Prior successfull testing, it was changed to NOTIFY but nothing worked out. calls were failing.
2. RTP payload = 101.

- NOTIFY + RTP Payload 96 + CUBE DP's SIP NOTIFY combination din't worked.

- NOTIFY + RTP Payload 101 + CUBE DP's SIP NOTIFY combination din't worked.

- RFC2833 + RTP Payload 101 + CUBE DP's dtmf-relay rtp nte + rtp payload 101 combination finally worked out.

Many thanks and support for this case... Cheers!

regards,

Ritesh Desai

[Chintan Gajjar]

I am glad that its working, i think all the variants(SIP Notify and kpml) would also work, may be it requires minor config tweaks.

1 suggestion, please have your dtmf-rely on dial-peer uniformly defined on all the dial-peers.

and have redundant  option for dtmf rely configured on your dial-peer.

like below:

insted of having on rtp-nte support 

dtmf-relay rtp-nte

you can have  all the supported options, 

dtmf-relay rtp-nte sip-kpml sip-notify cisco-rtp 

in above case if call comes with any of above matching the first preferred will be matched and call will be processed with that.

Happy UCCE