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PSTN calls for UCCX - No audio

Hi everyone,

We've been having issues with PSTN calls to UCCX.

This is the scenario

PSTN --- E1 --- CUBE --- SIP Trunk --- CUCM --- UCCX

For incoming calls from PSTN to UCCX trigger, the greeting message plays and caller selects an option of the menu that gets redirected to end user on CUCM, the ringback plays OK.
Once the call is established between PSTN caller and the end user, both ends don't hear each other.

We can make outbound calls from phones/softphones registered to Call Manager and there is audio both ways.
Inbound calls directly to CUCM extensions works well too, audio works for both sides.

But everytime a call comes to the UCCX application and the caller makes a selection or dials the desired extension, the call
is directed to CUCM and plays the ringback, but when the agent answers, the call goes no way audio.

This UCCX application has two different triggers, which comes from two PSTN numbers (but same provider) through two different gateways. So, one UCCX application has triggers 4300 and 7300. Calls that come through 7300 work fine for audio, and calls that come through 4300 are presenting the issue I mentioned. So I think the application configuration is correct.

I compared configuration for both gateways and configurations on CUCM for the two sites, and they are similar. Each one uses their respective Device Pool, Partition, CSS, MRGL and so on, but they are similar.

Looking at the logs for both calls, I could see that they both need to use MTP for DTMF mismatch. While the OK call uses hardware MTP from gateway, the NOT OK call uses Software MTP from CUCM subscriber (the call comes from Publisher).

I don't know if this is related to the issue, but the coincidence is that I found another site that uses SW MTP from CUCM Subscriber and it's not working either. And this was the only odd thing I could notice on the logs, because the packet captures for working and non-working calls shows that the end user sends RTP to the gateway and the gateway sends RTP to the end user.

All sites have all resources available: hardware resources from the respective gateways and software resources from CUCM publisher and subscriber). But they allocate different resources.

Did you guys have any similar issue? 

Thank you in advance.

32 Replies 32

Hello Dmytro,

I configured "dtmf-relay sip-kpml sip-notify" on router and "no preference" on SIP trunk and made tests, but it didn't recognize digits again.
In this case, the call never reaches the agent, it stays on the greeting message, because none of the digits I press are accepted.
This happened every time I changed SIP trunk to "no preference" and dial-peers to the ones below:
dtmf-relay sip-notify sip-kpml
dtmf-relay sip-kpml
dtmf-relay sip-kpml sip-notify
dtmf-relay rtp-nte sip-kpml

The only working scenario for DTMF is when I change SIP trunk to "RFC 2833" or "OOB and RFC 2833" and dial-peer to "dtmf-relay rtp-nte sip-kpml".
I attached logs from CUCM for this scenario, where MTP was allocated due to DTMF mismatch. I sent the password to you in a message.

TechLvr
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@KarolineSoares76078 Lastly, check if there is a SIP Normalization Script applied to your SIP trunk that modifies CUCM SIP messages before they are delivered to the gateway. 

Hi TechLvr, 

I checked the SIP trunk but there isn't any SIP Normalization Script applied to it.

Also did the test removing the software and hardware resources from the MRGL of the endpoint.
For this case, while the call is still on UCCX, it is using MTP because of the DTMF mismatch, then the MTP is deallocated and the call is transferred to the agent. At this point, it seems that MTP is no longer in use, and the issue of no way audio happens again.
I attached the CUCM logs here if you'd like to see, and sent the password to you in a message.

What happens now is that when changing configuration to eliminate the MTPs, I'm not able to choose options of the menu because digits aren't recognized. The audio keeps playing as if I haven't pressed any digits. The call is not transferred to agents.
And for the case where CUCM dinamically allocates MTP resource because configuration on SIP Trunk is RFC 2833, the MTP is used only while the call is reaching UCCX, once it is transferred to the agent, MTP is deallocated.
When call reaches agent, there's still something odd in the SIP message, because gateway receives something different than the information CUCM sent.

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