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cjoseph23
Beginner

Multiway and audio only participant

VCS Control Running 7.0.1

VCS Expressway Running 7.0.1

Codian 4501 MCU   running 4.3(2.18)

CUCM running 7.1.5

MX200/EX90 running 5.1.3

I can initiate multiway calls between my MX200s and EX90 just fine.   But if I try to bridge in a audio only call which connects via the SIP trunk to CUCM then i get a merge failed.  For Example:

I call an IP endpoint on CUCM from the MX200

Place this call on hold

Call an EX90 and get connected

Hit merge and it fails.

I can't seem to get the calls to merge

If I log into the Codian and dial out to the CUCM IP Endpoint, EX90, and MX200 then i am able to get the calls into a single multipoint conference.

Does multiway not support getting an audio only call in?

Regards,

CJ               

1 ACCEPTED SOLUTION

Accepted Solutions
Alok Jaiswal
Cisco Employee

Hi CJ,

yes, you can take a participant on CUCM into a multiway call.

i don't think we have any such limitation that says audio participants can't join in multiway call. for sip multiway you will see sip refer method.

what i found with CUCM in this kind of scneario's is tht the multiway uri comes to CUCM and since cucm doesn't to the domain routing and it fails to join the multiway call.

i see you are running cucm 7.1. can you add a sip domain route on CUCM for the multiway uri and then route it back on VCS sip trunk, should make the call work.

try it.

thnx

Alok

View solution in original post

10 REPLIES 10
Alok Jaiswal
Cisco Employee

Hi CJ,

yes, you can take a participant on CUCM into a multiway call.

i don't think we have any such limitation that says audio participants can't join in multiway call. for sip multiway you will see sip refer method.

what i found with CUCM in this kind of scneario's is tht the multiway uri comes to CUCM and since cucm doesn't to the domain routing and it fails to join the multiway call.

i see you are running cucm 7.1. can you add a sip domain route on CUCM for the multiway uri and then route it back on VCS sip trunk, should make the call work.

try it.

thnx

Alok

HI CJ.  I'd also check under the SIP trunk configuration called - Rerouting Calling Search Space on the SIP Trunk Configuration in the CUCM.  Make sure that the CSS is listed instead of None as well, as from what I can gather is required for the SIP REFER to take effect in order for the phone to make the proper call to the factory alias at which its being referred to.  I find if i have this set to NONE versus having my CSS in which my SIP trunk is a part of, will break my multiway configuration on my VCS.  Hope this helps. 

VR

Patrick

Thanks for the suggestions.  I was wondering how the CUCM would use the multiway URI to acutally place the call.  I was thinking that the MX200 which was joining the call would be the only one using the multiway URI but this makes sense.

I will try these over the weekend and report back what i find.

Regards,

Cyril

Thanks both of you for the help.  After adding the SIP Route Pattern for our SIP dialing domain (and a few other things) i was able to successfully use multiway to bridge our MX200s/EX90s and and CUCM audio only participant.  If any one else is trying to do this here is a doc I found which helped me out.

http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X7-0.pdf

Cheers,

cj

One more question - I am able to get a couple of our MX200s and a CUCM audio participant into a multiway call.  When all the video participants hit the END on the touch panel they will get disconnected but then the Audio participant stays connected to the MCU.

Is there a way to disconnect this audio only participant other than logging into the MCU?  The audio only participant I am connecting is a conference bridge number from our CUCM.

Regards,

Cyril

HI Cyril.  Not sure if it can be done.  The ad hoc conference on the MCU won't tear down unless everyone leaves.  If Telepresense Conductor was involved, could possibly build something in the conference template to destroy it after the chairman leaves the meeting etc, but without that, it would be tough to setup. 

If your running MCU 4.3, the last person before leaving can activate the inband DTMF (if enabled - Level 3 Advanced) by hitting "*" on the remote, and navigating to the endpoint and disconnecting it from there.  That would be another option if the end user would know to that,and that the last person to leave would do it. 

These are the only two options I can think of at the moment.  Maybe someone else here would have another to offer. 

VR

Patrick

Cyril,

You mentioned that you are able to connect your CUCM conference bridge into your ad-hoc conference.  I am wondering how you did that and what I might be missing.  I am assuming that you are refereing to a Meet-Me number.  I can call the meet-me from an EX90, but when I try to merge it with another EX90 endpoint, I get a "Cannot join when connected to more than one MCU" message.

Any insight would be greatly appreciated!

Thanks,

Arras

Hi,

We're having some similar problems with a customer, who's running;

- CUCM 8.6.2

- VCS 7.2

- MCU 4505 v4.3

and MX300, EX90 and CUVA (with 7942) as endpoints.

The situation is, that all TP endpoints are registered to VCS and IP phones are registered to CUCM. TP endpoints can call each other and start multiway without problem. TP endpoints can also call IP phones. In this case, CUVA automatically switches on and a point-to-point video call can be performed.

In the opposite direction, IP phones can (of course) call each other, and TP endpoints as well. We have set the SIP domain routing and route-patterns towards VCS.

The problem we face is about multiway and dialing into the MCU.

As described above, initiating multiway works on TP endpoints, but IP phones fail. Even further, if I set a temp meeting room in the MCU and register it to VCS, IP phones can dial into the static meeting room, but CUVA doesn't kick in and the call proceeds as an audio-only call.

Any thoughts on where I shall start toubleshooting?

Thanks,

Emre

Hi emresumengen.  Where is this failing for you? If your running Multiway and trying to REFER IP phone to Multiway Meeting, is the IP phone failing to enter? Or is it working and only get Audio only from CUVA?

Is your Multiway setup SIP only? H323 only?

When the originator of the Multiway over SIP here sends REFER to CUCM side and CUCM will return 202 Accepted on behalf of the phone.  CUCM will then search and send SIP NOTIFY to VCS indicating its progress.  If the join fails, CUCM may send 404 Not Found as the Rerouting CSS on the SIP Trunk to VCS may not be configured.  May be selected to NONE or have a CSS that the phone is not in and the search fails. 

H323 equivalent is FACILITY message indicating RouteToMC which interworks to SIP REFER. 

I'm curious to know what a log would say when you try to REFER the phone or CUVA to the meeting.  If its failing with 404 Not Found, do a check down on the Rerouting CSS on the profile for the trunk.  If it does join and get audio only, would be curious to see the exchange between VCS, MCU and CUCM as to why. 

Hope this helps.

VR

Patrick