H.323 Gateway between CME 10.5 and CUCM 12

[Rene MUGIRANEZA]

Hello,

I configured H.323 Gateway between CME 10.5 and CUCM 12, unfortunately I have the following challenges with calls:

CUCM to CUCM = Successful

CME to CME = Successful

CUCM to CME = Successful

But

CME to CUCM = Unsuccessful. for this, If I call from CME to CUCM, the phone registered under CUCM rings and when I answer the call, the voice is not passing through (If I answer call from CME to CUCM, there is no voice and calling phone i.e. phone from CME gives a busy tone after some 10 seconds. Note that If I answer call from CME to CUCM, the phone from CME still giving a tone that indicates the outgoing call then after 10 seconds it gives a busy tone).

Any documentation regarding this will highly be appreciated.

Your Kind support will highly appreciated

[Chris Deren]

Why not use SIP? With the versions you listed it would preferred protocol, and troubleshooting will be much simpler.

[Rene MUGIRANEZA]

Dear Chris Deren,
Thank you for advise. I will appreciate if you share with me any documentation (step by step) regarding integrating cucm 12 and cme 10.5 with SIP.
Thank you for your support again.
Regards,

[Chris Deren]

You just need to convert your dial-peers to be SIP by adding "session protocol sipv2" you should also update DTMF method to i.e. dtmf-relay rtp-nte sip-kpml, and define some global config, i.e.:

voice service voip
ip address trusted list
allow-connections sip to sip
sip
bind control source-interface <interface>
bind media source-interface <interface>
min-se 360 session-expires 360
ds0-num
header-passing
error-passthru
registrar server expires max 600 min 60
options-ping 90
early-offer forced
midcall-signaling passthru
no call service stop

then on CUCM create a SIP trunk that points to the CME IP address.

[Maren Mahoney]

I agree with Chris that moving to SIP is the way to go.

In the meantime, I have a question about your scenario:

When you call from CME to CUCM and the CUCM-based phone rings and you pick it up, does the CUCM phone immediately drop the call but the CME phone waits for 10 seconds before dropping the call? Or do both phones show active, and then both drop after 10 seconds?

The first one indicates a likely codec-mismatch issue. The second is some other problem.
Maren

[Rene MUGIRANEZA]

Hi Maren Mahoney
When I pick a call from CME, CUCM phone shows active and wait for 10 seconds then after gives a busy tone.
For CME based phone, when it calls CUCM based phone and when I pick the call, CME based phone continues showing the outgoing call and ringing then after 10 seconds it gives a busy tone.
Thank you

[Maren Mahoney · Accepted solution]

Then it sounds to me like you have a problem with session progress messages being negotiated when CME calls CUCM. The CME is not receiving the H225 Connect message sent by CUCM when the call is picked up on the CUCM side, or CUCM is expecting H245 negotiation at call pickup which the router is not responding to. I'm thinking this is a mismatch between H323 Fast Start (on the router) and Slow Start (on CUCM).

On the CUCM H323 Gateway configuration page, in the "Call Routing Information - Inbound Calls" section, look for a checkbox "Enable Inbound FastStart". If that is not checked, check the box and reset the gateway and see if that fixes the problem.

If that doesn't work, we'd need to look at your config.

Maren

(But I'd still plan on converting to SIP at some point. It is your best long-term plan.)

[Rene MUGIRANEZA]

Dear Maren Mahoney,
I come up to solve the issue with your guidance and I works now. I really thank you.
Do I really need to convert it to SIP? Kindly share with me a full guidance for SIP between CUCM 12 and CME 10.5. I appreciate your support.
Kind regards.

[Maren Mahoney]

Must you convert to SIP? No. However, the industry as a whole has been standardizing on SIP over the last few years and that trend will only continue. In addition, at some point you will likely have an IP-based PSTN connection and that will most certainly be SIP. So, should you plan to convert to SIP at some point? Yes.

First, you will want to learn about SIP so read up on it. There is an ocean of documentation out there. In addition, Cisco has posted a series of really excellent videos which cover SIP in-depth:

Collaboration Training Videos

And, as always, feel free to ask questions here.

Maren

(Glad it's working!)