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Unable to make conference call

B.log-after-I-enter-conference-I.log A.log-when-I-start-make-call-to-.log

Dear Experts,

We are using Cisco call Manager to make call and we have an issue making conference call with remote site.
We have a branch office in different country connected to it via Site to Site VPN. They also have Cisco call manager setup and we can make normal call to them using their extension number with no issue.

But they have a conference number which is 8989 and setup a meeting with us providing conference ID.
We can make call to 8989 but when I dial 8989, auto attendant message says “enter your conference ID followed by #” but if I enter conference ID, it says “sorry I didn’t get back” which means it doesn’t get my respond no matter what I type(correct or incorrect ID).

I have no control over Remote site CME.

I have enabled logging in CME and issued debugging command “debug voip ccapi inout” set to log to Syslog server I setup.
Attached is syslogs I captured when I make call to 8989.
The extension number I used to test from my office is 6531 (name -Server Room )
Would you be able to help me check from logs what is causing that they didn’t our respond ?
Appreciate your help.

Thanks
Regards
BMT

Hi mate,
So is the CUCM and CME connected via a trunk?
Can you please post your configuration showing how they are connected?
I suspect that you have an issue with how you are delivering DTMF.

ASKER

Hi Mihailpetreski

Confg is attached. Could you check and let me know which config is causing ?

Thanks
Rgds
CME-config.txt

Hi,
Just to confirm, are you using CME to CME or Call Manager (CUCM) to CME?

If it is CME to CME, What dial-peer are you matching on your side when you call the other site? On that dial-peer, you could try swapping dtmf-relay cisco-rtp with dtmf-relay sip-notify and testing again.

if you are using CUCM with CME, then please post your config so I can see how CUCM is configured to communicate with CME.

ASKER

Hi Mihailpetreski,

Thanks you for your quick response. Are you meaning that if remote site is CUCM ?
I'm using CME. I'm not sure what remote site is using.

To swap dtmf-relay cisco-rtp with dtmf-relay sip-notify , I have to issue that command only for that Location right ? As you can see in Config, I have 5 countries connected.

And the problem that I have now is with *LocationB* (dial-peer voice 103 voip)
¿You mean change to "dtmf-relay sip-notify" for LocationB ?

I'm sorry that I'm not so familiar with Cisco IP phone and appreciate if you can let me know a bit detail what command I should issue and test.

Thanks
Rgds

Ok, that clears it up a little more - thansks for the extra detail.

conf t
dial-peer voice 103 voip
no dtmf-relay cisco-rtp >> press enter
dtmf-relay sip-notify >> press enter
exit

Should look like this:
dial-peer voice 103 voip
description **LocationB**
destination-pattern [5,8]9..
b2bua
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.57.10
dtmf-relay sip-notify
no vad

Once you have done this, try and make another conference call and press numbers to see if it recognizes DTMF.

Please also capture the following debug command if you have any issues:
debug ccsip all
remember to turn the debug off later :)

ASKER

I just tested enabling "dtmf-relay sip-notify" command.
When I issue that, I can make call to 8989 but once I enter any ID (as conference ID)
call just dropped without getting any reply.

I have set the original command back for now.

I'll try to get the config file of remote site and see if we can get some.

which device is generating the auto attendant message in order to enter the conference ID?

A call drop could be codec related.
You will need to capture all the debugs to determine why the call is dropping and what is also in the sip messages.

ASKER

Hi Mihailpetreski,

You are right. I just get to know that they are using "Cisco Unified Communications Manager business edition, version 7.1.2. ) which is server based.

Up to now, the only test I have performed is disable dtmf-relay cisco-rtp and enable
dtmf-relay sip-notify as you suggested. But after I make call and as soon as I enter conference ID call dropped.

I tried to get debug log by issuing "debug ccsip all" but I didn't get chance as users were busy with making call.

I get some screenshot how they have configured.
Could you have a look and advice me the action required?

Thanks
BMT
1.JPG
2.JPG
3.JPG

the problem will not be solved without knowing which device is answering the call and playing the conference message. is it the unity connection?

ASKER

Hi Hus1984,

Their remote CUCM is answering the call. We can make regular call to them with no issue.
But I'm not sure why we can't conference call.

The remote IT guy provide me conference number they have set and provide me conference ID number.

I'm only familiar with Cisco router and firewall but not familiar with Cisco call manager configuration hence I can't know what could be the problem except collecting debug logs for now.

but the call manager does not have feature to answer calls and playing message to enter the conference ID. could please check with them about this point and post the dial peer that is going to their call manager

ASKER

Dial Peer going to their call manager is as stated below.
My CME config file is attached in my first comment if you need.
I'll check with them which device is answering the call to play conference message then.
Thank you to both of you

dial-peer voice 103 voip
description **LocationB**
destination-pattern [5,8]9..
b2bua
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.57.10
dtmf-relay sip-notify
no vad

it is clear. I am waiting for your response to know what is the device that answering the call

Have you managed to obtain any further logs to see what error messages are being presented?
I'd be surprised if this worked without MTP resources.

Also, does this feature work with other Locations?

ASKER

For other location, we don't need to enter any conference ID, just dial the conference number (example 8765 )and join conference.

You might find this article interesting if you have some time to go through it
http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1036903

I believe that you will need to use the RFC 2833 DTMF MTP Passthrough Feature.
This will require that you have MTP resources available.

The MTP or transcoding module on a gateway detects RFC 2833 (DTMF) packets from an IP endpoint. You can configure whether it should do either or both of the following:
•Generate and send an out-of-band signal event to the call manager
•Pass the packets through to the other IP endpoint (default)

Once again, we can help further once you let us know what the remote site Conference is (be it MeetMe, Untiy Con, MeetingPlace) and provide some logs for a failed call.

ASKER

Hi mihailpetreski,

Thanks for your info.
How can I make MTP resource available ? I'm using hardware CME 2851 (Cisco IOS Software, 2800 Software (C2800NM-IPVOICEK9-M), Version 12.4(24)T6)

After MTP resource available, how can I activate and configure "RFC 2833 DTMF MTP Passthrough Feature" in my CME ?

I have asked the guy from remote site what the conference is and waiting for his reply, may be he will reply tonight due to different time zone.

I'll gather more info today and perform to gather debug logs tomorrow morning as I'm not in office today.

ASKER

Hi mihailpetreski and Hus1984,

I got respond from remote guy that when I make conference call to 8989, it will go to the call manager (192.168.57.10), but there is a SIP trunk setup to pass the call onto the Lync server(Microsoft Lync server) (192.168.xx.xx) – The Lync server is the one which is playing conference message )

FYI, I can ping to Lync server IP from my network subnet.

So which command I should issue for it ?

Thanks
Rgds

so the remote Call Manager is final step before the Lync so it might change the call parameters. can you route the call directly to Lync?

ASKER

Hi Hus,

Thanks for your reply.

You mean change to below command?

dial-peer voice 103 voip
description **LocationB**
destination-pattern [5,8]9..
b2bua
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.xx.xx (Lync server IP )
dtmf-relay cisco-rtp
no vad

How about if I make regular call to remote users extension will still go through ?

Rgds

yes

ASKER

Could you let me know which command is correct from below as previously Mihailpetreski
suggested to change from dtmf-relay cisco-rtp to dtmf-relay sip-notify

dtmf-relay cisco-rtp or

dtmf-relay sip-notify ?

My original setting set is dtmf-relay cisco-rtp

Thanks

keep it dtmf-relay sip-notify, please test and let me know

ASKER

OK I'll let you know tomorrow morning

Thanks

ASKER

Hi Hua,

I've tried setting as Lync Server IP but doesn't help. When I make call ip phone says Failed.

I've set back to previous setting and issued "debug ccsip all" command and gathered debug logs.

Attached two files are
A.When I make call before entering conference ID and
B. After I enter conference ID

Would you be able to find any thing from the logs ?

Thanks
Rgds
A.Dial8989-and-Before-entering-c.log
B.After-entering-conference-ID.log

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ASKER

Hi Hus,

It is the same. Call failed. But I tried something as below as a test.

session target ipv4:192.168.57.10
dtmf-relay rtp-nte

Seems, it works, I get continuous reply from auto respond message player once I enter conference ID but I hang up.
But how come it seems to be working if I set call manager IP?

If I set as below, it doesn't work

session target ipv4:192.168.57.10
dtmf-relay cisco-rtp

I can ping to their Lync server and Call manager from Voice network sunbet.
I asked remote guy to ping to one my IP phones but he is not replying anything.

Seems the command "dtmf-relay cisco-rtp " is the cause
But not sure yet until I setup the conference meeting with remote guy

Thanks a lot Hus ..really.
Let me know any suggestion

on the remote Call manager they have sip trunk to Lync, because of this when you use the Call manager IP it works. but it should work directly from your CME.

the problem in using remote CUCM is:
CME-------CUCM-------Lync so in the call manager the call parameters will be changed which will cause difficult troubleshooting because of that I want to go directly to Lync.

the important command to go directly to Lync is the (session transport UDP)

if you can debug the csip messages this will help us find the correct answer

ASKER

Hi Hus,

Debug csip was attached in my first comment today I post.

Thanks

May 31 13:52:13 192.168.1.1 981538: Disconnect Cause (CC) : 16
May 31 13:52:13 192.168.1.1 981539: Disconnect Cause (SIP) : 200
16 and 200 suggest that you have normal call disconnection.

Seems, it works, I get continuous reply from auto respond message player once I enter conference ID but I hang up.
But how come it seems to be working if I set call manager IP?
I'm not sure I understand what you mean in the above, can you please describe what happens to the call when you press buttons to enter the conferenceID? Does the call immediatley disconnect or are you getting a response to the digits entered?