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CUCM 8.5 and 'Secondary Dialling' with SIP phone

We have CUCM 8.5.1.x The standard 'skinny' Cicso phones all work fine. The issue is with the SIP endpoints on an Avtec dispatch console. They work fine- BUT once a call is placed no more DTMF tones are sent across the line. I can hear the sidetone in the headset- but it does nothing on the other end. Example: Place a call- get to a 'voice mail tree'. More entries are required to go anyplace. Pressing the button gives a sidetone, but nothing happens. Looking at the Wireshark capture it appears it is not enabled on the Cisco side (so the tech guys from Avtec say). I'm not exactly sure what Cisco calls that - I saw 'Secondary Dialling' once. What needs to be 'turned on' in the Call Manager to allow this to happen. HELP! Mark
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Advisor

Sounds like those phones

Sounds like those phones support RFC2833. Without a view into your system and what you have, it's hard to tell you in the forums where things may need to be tweaked.  In this case, TAC is a good call I think.

Hailey

9 REPLIES 9
Advisor

What you are talking about is

What you are talking about is DTMF relay.  What is the voicemail system you are calling into and how does it integrate with CUCM?

It could be any voicemail

It could be any voicemail system- whatever the Dispatchers are calling.

I've tested it calling into the Cisco CUE for testing, but it needs to work with any system.

Where can I look for information on DTMF relay and basic SIP phones?

 

thanks

Mark

Advisor

Well, some of what you need

Well, some of what you need to look at depends on the call flow.  You need to take a look at the voice gateway (if issues with external calls or inbound calls to your own systems) to see which DTMF option is applied (i.e. dtmf-relay h245-alphanumeric vs. dtmf-relay ntp-nte.  If you are calling into a system (such as CUE) that is internal but integrated via SIP trunk, then you'd have to look there as well.  There are a two types of DTMF in that it can be in-band or out-of-band.  Out-of-band passes via signaling (i.e. H323, SIP, etc.) and in-band is RFC2833.  This can also be complicated by the actual endpoints involved - such as in your case, you are using a (presumably) 3rd-party SIP endpoint and having issues.  It may be beneficial for you to open a TAC case as they can help troubleshoot the call flow and have a better look at the actual problem.

Hailey

Thank you. Knowing what Cisco

Thank you.

 Knowing what Cisco calls it is a big help in opening a TAC.

The vendor of the basic SIP phone (Avtec) seemed to say they would normally use 2833...at least from the Wireshark capture I did showing what was sent. 

Looks like something is not properly enabled.

There is no separate SIP trunk enabled for those phones (at least none show up).

They do make normal calls just fine, the only limitation is entering DTMF during the call.

 

thanks for the ideas!

Mark

Advisor

Sounds like those phones

Sounds like those phones support RFC2833. Without a view into your system and what you have, it's hard to tell you in the forums where things may need to be tweaked.  In this case, TAC is a good call I think.

Hailey

Beginner

Mark - We are having this

Mark - 

We are having this issue as well with our new Avtec consoles. Did you get this resolved, and if so, can you point me in the right direction?

Thanks

Adam

No, I never did get this

No, I never did get this resolved.

I didn't open a TAC and got busy putting other fires out and just haven't gotten back to it.

 

I did get the phones (and radio enpoints) recording with our Eventide recorder from the Avtec console.

 

Mark

Beginner

Mark - I had a Cisco tech

Mark - 

I had a Cisco tech help out with this. It looks like Avtec is using KPML (and I don't think that's in keeping with RFC2833). We had to change the "DTMF Signaling Method" on the trunk configuration to "OOB or RFC2833". I think you also need to enable an MTP but I'm not positive on that. Hope that helps!

 

Adam

Thanks!  I have not had time

Thanks!

  I have not had time to revisit this issue at all.

I'll have to see our current setup.

I didn't make a SIP trunk for those console phones (unlike the recording, which I needed to do). So I'll have to remember where to make trunk changes.

Might be above my permissions, but I'll start to look into it.

 

You have it working now?

 

thanks

Mark

 

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