DTMF for Incoming over SIP trunk is not working

[Anas Abueideh]

Dear Experts

Kindly find the above topology.I have issues with DTMF calls when they hit the IVR. the calls codec is G729, we have configured IOS transcoder to be used when the calls hits the UCCX.

The IVR is working but unfortunately the DTMF. it always give an error OOBand RFC2833 mismatch which is required MTP.

the cube dial-peer towards CUCM 10.5 has dtmf-relay rtp-nte. 

the transcoder configuration is

dspfarm profile 1 transcode  
 codec g729abr8
 codec g729br8
 codec g729ar8
 codec g729r8
 codec g711alaw
 codec g711ulaw
 maximum sessions 14
 associate application SCCP

the transcoder is registered and associated with the SIP trunk and CTI route point.

Thank you in advance

Anas

[Chris Deren]

Add MTP and see if that solves your issue.

[Anas Abueideh]

Hi,

One question regarding MTP and transcoder.

I have only transcoder, when I do show sccp I have the below output

Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: x.x.x.x, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 28, Reported Max OOS Streams: 0
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 30
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

is that enough as an MTP, or we need to configure dspfarm profile 3 mtp ?

Regards

Anas

[Chris Deren]

A transcoder can do some MTP functionality but not all, i.e. no translating, and I doubt it can do DTMF bridging, this should be described in SRND. I would definitely build MTP on the GW and register with CUCM and ensure it's in its own MRG listed ahead of the MRG that hosts the xocder in the MRGL applied to CTI ports/ingress trunk.

[Anas Abueideh]

Hi,

per the SRND, CTI route supports all OOB methods and doesn't support RFC2833

I have already tried configured MTP without success.

My traffic is G729 and mtp support only g711. I also add the transcoder in the call manager as transcoder not mtp

dspfarm profile 2 mtp

 codec g711ulaw

 codec pass-through

maximum sessions hardware 2

maximum sessions software 30

associate application sccp

no sh

any idea ?

Anas

[Ayodeji Okanlawon]

If you have configured an MTP, then your MTP has to be configured for g729, if that's the codec you are  using otherwise cucm will complain about capabilities mismatch between the mtp and the devices that need it..I am sure you can use your existing transcoder. Just ensure that the region between your transcoder and sip trunk is set to G729

[Humza Khan]

Hi Anas,

I also faced that sip trunk call hang issue, So i check Mtp option in sip trunk which snapshot is attached, might be that would be helpful for you.

Regards,

Humza Khan

[Anas Abueideh]

Hi Guys,

I have tried all options you given me. Unfortunately they didn't work.

I change my setup to use CUBE to CUBE connectivity. after that i start found issues with incoming calls, sometimes the IVR answer the calls and sometimes not.

I have posted that issue in a new thread.

https://supportforums.cisco.com/discussion/12401091/calls-not-answered-cube-randomly

thanks for your help

Anas

[Ayodeji Okanlawon]

There are two options you can try

1. Because UCCX jtapi ports only support out of band DTMF, you can configure your cube dial-peer pointing to cucm to use both rtp-nte and sip-kpml. SIP-KPML will be oob and hopefully you will not need MTP

dial-peer voice xxvoip

dtmf-relay rtp-nte sip-kpml

2. Use the transcoder that you have now, it looks like it is capable of doing dtmf-conversion in its capabilities. Assign it to the MRG of your sip trunk.

SRND states that a transcoder can do the job of MTP.

"A transcoder is also capable of performing the same functionality as a media termination point (MTP). In cases where transcoder functionality and MTP functionality are both needed, a transcoder is allocated by the system. If MTP functionality is required, Unified CM will allocate either a transcoder or an MTP from the resource pool, and the choice of resource will be determined by the media resource groups, as described in the section on Media Resource Groups and Lists.

[GHARIP MOHAMED MARZOK]

Hi Ayodeji:

i have CME integrated with Unity and there is Mobile line on the CME router when we dial it call transfered to Unity and hear the IVR asking if i know the extension or wait for help 

if i called 888 from internal IP phones i hear the IVR and can dial the extension successfully

but if i call the mobile number as external call it transfer my to the IVR  but when i dial the internal extension it don't feel it and after while transfer to the operator

the below dial peer which matches the external call and transfer me to Unity

dial-peer voice 888 voip
 description Voicemail
 destination-pattern 8[018][08]
 session protocol sipv2
 session target ipv4:X.X.X.X
 dtmf-relay sip-notify sip-kpml
 codec g711ulaw
 no vad

can you help me please ?

[Ayodeji Okanlawon]

What is the dtmf on the inboud dial peer set to?  Can you please post your full configuration? 

[GHARIP MOHAMED MARZOK]

thanks for your reply , find attached file , thanks

[Ayodeji Okanlawon]

Are you using ISDN for PSTN? What is the integration between unity and CCME?

[GHARIP MOHAMED MARZOK]

This is a mobile line on brimacell conected to the router on FXO port , and this is CME and Unity Express 

[Ayodeji Okanlawon]

Ok please do a test call and send the ff

debug ccsip messages

debug voip ccapi inout

Attach the logs here..include calling and called number