Intercluster trunk restrictions

Dear All,
Is there a restriction in creating an intercluster trunk between 2 call managers. I'm trying to simulate this in a lab environment where i have one Call manager version 10.5 connecting to a Call Manager 7.1.X to route some calls outside my lab CUCM to another one.
Do intercluster trunk need to match CUCM versions? i'm not really seeing any calls getting through across the trunk, in call manager 7 there seems to be no way to check the uptime of the trunk nor see if it's active. I'm using DNA tests to see if the call gets routed over de trunk and it does.
Kr,
Yannick Vranckx

No, you can have an ICT between different version, hopefully you're not using a SIP trunk, we only recommend that if all clusters are 8.6(2) and above, if you need to trunk with a lower CUCM version, use the old ICT non-GK controlled. Aside from that, the rest of the config is the same, RP, RL, RG, CSS, partitions, etc.

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Intercluster trunk with Lync Integration

I have a customer who is looking to build a couple of intercluster trunks between 8.5 CUCM clusters. Originally I was looking at doing a non-gatekeeper h323 trunk, but the customer has stated that they will be moving to a Lync-CUCM integration in the near future. Question is is the h323 intercluster trunk still a viable option, and if so what do I gain or lose with a h323 trunk as opposed to a SIP intercluster trunk?

H323 trunk is still a viable option however imagine a call from Cluster A's 7945 SCCP phone to Cluster B's Lync client. It will be SCCP -> H323 -> SIP -> Lync client. Do you want all that protocol translations. I wouldnt just from a troubleshooting perspective but should work fine though.

One Way Calling on Intercluster trunk between 4.1.3 and 3.3.x

We've encoutered a bizarre situation, one which so far even TAC is stumped on.
We just turned up a new site, running 4.1.3SR3b and have set up the usual intercluster trunks (non-gatekeeper) between the new site and our other sites. Most of our other sites are 3.3.4 or 3.3.5. Calls from the other sites to the new 4.1.3 site work fine, but when you try to call from the 4.1.3 site to another site, the phone rings on the receiving end for a split second (not even a full ring), then the calling end gets a fast busy. Detailed traces on both ends show the info getting to the receiving end, but when the call drops, this appears on the sending end:
07/15/2006 11:30:38.499 CCM|StationD: (0000129) DEBUG- star_DSetCallState(6) State of cdpc(337) is 5.|<CLID::StandAloneCluster><NID::10.40.2.11><CT::0,0,0,0.0><IP::10.12.1.12><DEV::>
07/15/2006 11:30:38.499 CCM|H225D::restart0_TcpStopSessionInd: H225Cdpc(2,100,38,91), ConnPid(2,100,132,258), crv=12, mCiCrCrpEntries=1, mCrtoCiEntries=1|<CLID::StandAloneCluster><NID::10.40.2.11><CT::0,0,0,0.0><IP::><DEV::>
07/15/2006 11:30:38.499 CCM|Cc::wait_CfDataInd, total fie to handle is 0 |<CLID::StandAloneCluster><NID::10.40.2.11><CT::0,0,0,0.0><IP::><DEV::>
07/15/2006 11:30:38.499 CCM|ConnectionManager - wait_AuDisconnectRequest:NO ENTRY FOUND IN TABLE,CI(33555134,33555133),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)|<CLID::StandAloneCluster><NID::10.40.2.11><CT::0,0,0,0.0><IP::><DEV::>
We just tried the latest engineering special (63 I believe), but the issue remains. Has anyone encountered this before?
This occurs with either G711 or G729 codecs, with or without MTP's enabled on the intercluster trunks.

Hi,
Did you checked the H245 Options in the CCM trace options?
I dont see any H245 messages in CCM.
I see that this IP Phone (43867) is registered to node 2.
Is this the SUB in PHL cluster?
We sent a H225Setup sent to IP=10.12.1.12
In the other side:
H225Handler::SdlConnectionInd: from IP=10.40.2.11
If this non-gatekeeper-controlled intercluster trunk accesses the device pool of a remote non-gatekeeper-controlled intercluster trunk and that device pool has a second Cisco CallManager node, you must enter the second remote Cisco CallManager IP address/host name in this field. (ie. Cluster1 4 servers and Cluster2 2 servers
ICT1 has a CCM Group which contains 3 servers.
ICT2 has a CCM Group which contains the 2 servers
In ICT1 you need to enter the 2 server's IP addresses.
In ICT2 you need to enter the 3 server's IP addresses)
Make sure the order is correct.
In the incoming cluster I see IP Phone is registered to node 3.
Node selection for outgoing calls is controlled by which node the originating endpoint is registered with. If the originating endpoint is registered to the same node as the H.323 Trunk, the call is signaled from that node. If the originating endpoint is not registered to the same node as the H.323 trunk, a random selection is made between all nodes configured with the H.323 trunk
This error is the one which terminates the call taken from SDL in CCM SUB 3.
SdlSig-I | CcDisconnReq | restart0 | StationD(3,100,92,87) | LineControl(1,100,37,2258) | (1,100,101,34868).1-(*:*) | [NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP: 0] Cause=41
Temporary failure (cause code 41)
Normally this happens when GW, in this case the ICT is not registered correctly.
In CCM PHL, I see this:.
000221594| 06/07/15 11:30:38.499| 002| SdlSig | TcpStopSessionInd | restart0 | H225D(2,100,39,33) | H225Handler(2,100,40,1) | (0,0,0,0).0-(*:*) | [R:NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP: 0 DBP: 0] PID=(0,0,0,0)
000221595| 06/07/15 11:30:38.499| 002| SdlSig | TcpStopSessionInd | call_received7 | H225Cdpc(2,100,38,91) | H225D(2,100,39,33) | (0,0,0,0).0-(*:*) | [R:NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP: 0 DBP: 0] PID=(0,0,0,0)
This means that TCP connection was ended. By checking the SDL looks like destination node send a RST at TCP level and thats why call finishes. But problem seems to be the CMH cluster.
If ICT configuration is fine I would reboot|restart ccm.exe service in the CMH cluster.
If after that it doesnt work I would open a TAC case so we can get access and check whats going on.
HTH
//G

Intercluster trunk

This is a stupid question but I need to clarify it,
I have to seperate site running callmanager and I want to set up an intercluster trunk between them. Does Phone A at site A need to have an ip path to Phone B at site B, or is it only the callmanagers subnets that need to have a connected IP path? in other words do the callmanager at each site proxy the voice from the phones?

Hi,
You need to have an IP path between the two phones. CallManager is responsible of the signalin. Voice is transmitted from an end IP device (phone, gateway) to an another IP end divice (phone, gatewat).
Hope this helps,
Juan Luis

Video call from Cisco 8945 SCCP to Polycom VVX 1500 Third party Sip Phones between intercluster trunks is not working

Dear All,
We have two cucm Clusters in Different Locations between that clusters i created
Inter-Cluster Trunk (Non-Gatekeeper Controlled) Now all are working fine Between Clusters
audio calls & Video calls between sccp 8945 phones , but iam facing a Problem with third party
Video Phones (Polycom VVX 1500 ) Third Party SIP Phones located in second cluster, From 1 st cluster cisco 8945 Video
phone to 2nd cluster Polycom Video phone all calls are works for voice call only, but no video ,
Please Suggest me Solution.
Thank you,
Sriman

Try setting up a SIP trunk between the two clusters and set a route patten just to the VVX 1500 and check how that goes.
From memory inter-cluster trunks are a H.323 like protocol which might have video inter-op issues with the Polycom device.

Intercluster trunk problem

I have a trunk (Non-gatekeeper) between the legacy CCM 4.12 system and the new CCM 6.0. The trunk on the CCM 4 has the SUB then PUB listed, same on the CCM 6 side. Calls flow from the CCM 4 side to CCM 6 side no problem. Calls do not work from the CCM 6 to the CCM 4. Now here is the interesting part. This was working when I only had the PUB running on the CCM 6 side. I just added the SUB. So, I changed the server group to have the SUB listed first and moved the PUB to second. But I have updated the trunk config to match. But if I move the PUB up in the list and leave everything else the same and do a reset everything starts working again.

I have the same problem with the ICT.
from a single CCM 6.0 to a 4.2 cluster works fine, but not the other way.
Do you resolved your problem?If yes what you changed?

CUCM - trunking between 7 and 9

Hi,
Document needed to do a trunking between cucm 7 and cucm 9 ( intercluster trunking)
We are planning to route a PSTN call from IP phones which are registerd with old CUCM on version 7 from our new cucm 9 which has FXO- line active.
how do I achive it.
Regards

Hello
You have to use intercluster non-gatekeeper controlled . I do nothink there is special document for this , but you can see the below link:-
https://supportforums.cisco.com/discussion/11766291/intercluster-sip-trunk-between-cucm-version-71-and-91
Thanks
please rate all useful ifnromation

CWMS Call to IP Phone via ICT trunk, after answered no audio

CWMS ->CUCM 9.1 -> CUCM 8.6-> CUCM 8 IP Phone.
We have installed the CWMS which has SIP trunk to the CUCM 9.1 for audio to phone. The CUCM 9.1 has an Intercluster Trunk to CUCM 8.6. These two are separate CUCM clusters. The customer bought these systems from different vendors.
When the CWMS do a call back to the CUCM 8 IP Phone, the phone is ringing and can answer. However, on the CWMS meeting page, it shows call back failed and no answer. This is strange as when the CUCM 8 IP phone call to WebEx number the CWMS able to answer and user can join the meeting and works normally.
CWMS ->CUCM 9.1 -> CUCM 8.6-> Voice Gateway -> PSTN
In this scenario, when CWMS call to PSTN, e.g. mobile number, the mobile phone will ring and can answer. But the call drop after that without any audio heard. Same thing happens when the mobile phone call to CWMS DID number. It answered bu no sound, then the call drop.
The reason is connected above to the CUCM 8.6 as the PRI is connected to the voice gateway which is control by CUCM 8.6 via MGCP.
Any idea what's wrong in both scenario?? The weird issue is in both scenarios the call is ringing and after answered, there is no audio. IT seems the CWMS no getting any signal the call has been answered.
The CUCM 9.1 has IP phone registered to it as well and no issues for those IP phone to call WebEX CWMS and have conference.

Hi Yong,
What is the version of your CWMS system?
What is the size of the solution? (50, 250, 800, 2000 users)?
Do you have High Availability (HA)?
Keep in mind that for the appropriate setup of CWMS and CUCM integration you need to have at least 2 SIP trunks between CUCM and CWMS. If you have 2000 users system or smaller systems with HA, you may need to create more SIP trunks to fulfill the integration. Please, review the following documentation for configuring CUCM for CWMS deployment: http://www.cisco.com/c/en/us/td/docs/collaboration/CWMS/2_5/Planning_Guide/Planning_Guide/Planning_Guide_chapter_0111.html
If that is verified, and still doesn't work properly, without looking at the SIP negotiation traces for the affected calls, I can't tell you what the issue might be. Hence, if the configuration is verified, you may need to open a call with CUCM TAC to take a look at the SIP exchange between CWMS and IP phone and see if there is an issue with the call setup.
I hope this will help.
-Dejan

Intercluster communications

Hello,
I've configured two CM clusters and they are able to reach each other via inter-cluster trunk (no gatekeeper control). Both clusters have their own separate Unity servers. Only cluster 1 has a POTS connection, cluster 2 is able to dial out of the pots connection via the intercluster trunk. My question is; when someone dials in to the autoattendant on cluster 1 and wants to reach someone on cluster 2, how does the Unity server on cluster 1 need to be set up so that it can build connections to phones on cluster 2?
-Shikamaru

In large systems, you might have to configure more than one cluster to handle the call processing load.Refer URL
http://cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00800c4ca9.html#xtocid4

Inter cluster trunks

Is it possible to conect a intercluster trunk between the following 2 CUCM versions
version 5.1 and version 8.5?
All routing is correct and wokting fine however the inter cluster trunk will not allow calls to go through.
Dialed number analyzer confirms the path of teh call is reaching the inter cluster trunk
WHan adding a route pattern to the trunk the status and ip address changes to unknow within the version 5 cucm?

Yes, you can create ICTs from CUCM 5.1 and 8.5. When you create these trunks, it is as good as adding a H323 trunk. Therefore the trunk status would show unknown.
On calls not going through, you might want to consider the number of digits being dialed and what is being sent across the trunk. Also, what message do you get on the call when you dial through the trunk ?

CAC complex scenarios - please help!!!

I have read the CAC section of the SRND and I think I'm still confused on certain sections of the document. Would greatly appreciated anyone help on this :-)
Just assume RSVP is not a feasible option for now...
Re: Diagram 1
i) In this scenario, how many CCM clusters would you have & why? Would you put a cluster to each hub and remote site (i.e. to all sites)?
Ii) If each site has it's own cluster, how many GK would you use and why?
Re: Diagram 2
Iii) Assume it is a "Centralized model" where the HO hosts the only CCM cluster, do you see any problems in terms of CAC?
Iv) What would you configure for the location for the Non-GK controlled trunk between hub office (site D) & HO (site G)?
V) Do you ever configure any locations for the HO site? Or just leave the HO phones and HO Intercluster trunk as "none"?
Vi) Say if site A makes 7 calls to site B, and if site C also makes 8 calls to site B simultenously, can site B still call HO? I think the answer is no. Location CAC is not really designed to support multi-tier environment? anything we can do to fix this problem if the scenario has to be in Centralized model?
(N.B:- I have changed the location to be 'number of calls' rather than 'bandwidth' for simplicity)
Re: Diagram 3
Vi) In the centralized model,.If we have to reserve a certain bandwidth along all links end-to-end by RSVP, how is that differenet from using CAC? Don't we still need to over-pump the links? Say if a certain path along the RSVP path reserves 60 calls, wouldn't we need to RSVP for 60 calls through the whole path just like CAC?
Re: Diagram 4
Vii) For Tail End Hop off, assume all the voip dial peer has been setup on all the VGs, and the Route Pattern at Centralized CCM has been setup to point to the local VG (at site G) as Route List. But If the bandwidth between the hub (site D) & HO (site G) are all used up, but I want the user at site H to still be able to call the PSTN, can I use AAR in this scenario?
Thanks!

Hi thisisshanky,
Re: Case 1 (Diagram 5)
i) At Hub (site D), shall the 'locations' for the 3 x Non-GK intercluster trunk be left as 'none'?
ii) For the 'locations' for GK Inter-cluster trunks, shall they be set to 'none' as the traffic passing thru the GK are already being managed by GK already?
iii) If Hub (site D) wants to make 30 calls to site E, F, G & H at any time, would the total interzone bandwidth for ZoneA be 70 calls?
iv) Now, assume Zone B (site E, F, G & H) only needs to make 20 calls in total back to Zone A at any time, but since Zone A needs to make 70 calls as per iii, what should we do at GK?
bandwidth interzone ZoneA '70 calls'
bandwidth interzone ZoneB '20 calls' (would this restrict ZoneA to only be able to make 20 calls to Zone B?)
Re Case 2:
Can you please explain more on what problems will be faced since you mentioned it is not a true hub & spoke?
Re Case 3:
Can anyone else please shed some light on RSVP?
Re Case 4:
I'm a bit confused between the difference between the following 3 ways of setting up PSTN fallback if WAN bandwidth is insufficient in GK scenario
Say I want to call extn 3xxx on the remote office. The intercluster prefixs are *801, and the PSTN digits for the site is 9231-3xxx.
Method 1:
Configure a route list that contains two route groups. The first route group contains the intercluster trunk that routes outgoing calls over the IP WAN. Second route group contains the Voice Gateway.
Method 2:
Just on the Voice Gateway:
dial-peer voice 1 voip
destination-pattern 8013xxx
session target ras
preference 0
dial-peer voice 2 pots
destination-pattern 092313xxx
port 1/0:23
preference 1
Method 3:
Use AAR
Another extra Q regarding bandwidth for GK.
According to Cisco SRND, GK uses double the bandwidth as the codec call rate, so...
G.711 - GK uses 128kbps
G.729 - GK uses 16kbps
However, this is excluding all Layer 2, IP & RTP overhead... so when i am planning for number of calls, what values should i use as 'bandwidth per call'?
Cheers,
Hunt

The difference between H323 and ICT?

Hi Team
What is the difference between a normal H323 trunk and an Inter cluster trunk in CCM413?
When connecting a CCM and a CCME which trunk should we use?
Thank you very much for your help.
Jun

Use Intercluster trunk:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00806a803d.html#wp1057456
Please rate helpful posts.

What is the difference?

I'm a bit confused between the difference between the following 3 ways of setting up PSTN fallback if WAN bandwidth is insufficient in GK scenario
Say I want to call extn 3xxx on the remote office. The intercluster prefixs are *801, and the PSTN digits for the site is 9231-3xxx.
Method 1:
Configure a route list that contains two route groups. The first route group contains the intercluster trunk that routes outgoing calls over the IP WAN. Second route group contains the Voice Gateway.
Method 2:
Do everything on the Voice Gateway:
dial-peer voice 1 voip
destination-pattern 8013xxx
session target ras
preference 0
dial-peer voice 2 pots
destination-pattern 092313xxx
port 1/0:23
preference 1
Method 3:
Use AAR
Another extra Q regarding bandwidth for GK.
According to Cisco SRND, GK uses double the bandwidth as the codec call rate, so...
G.711 - GK uses 128kbps
G.729 - GK uses 16kbps
However, this is excluding all Layer 2, IP & RTP overhead... so when i am planning for number of calls, what values should i use as 'bandwidth per call'?

The following calculations are used:
Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)
PPS = (codec bit rate) / (voice payload size)
Bandwidth = total packet size * PPS

Route-List not using 2nd Route-Group when PRI cable is removed from H323 gw

CCM 4.1.3sr3b
Router - 2811 - 12.4.4T2
1 site with 2 gateways, gw1 and gw2
RL has both gateways as RG's.
If gw1 has its PRI cable removed, CCM RL will not pass call into gw2 - Not working.
If gw1 is turned of/Ethernet removed RL will wait the h323 timeout before passing call to gw2 - Working OK.
Is this how H323 works if the PRI is down ?
I know MGCP would work but is not an option.
I would have thought that the H323 gw1 would report that there are no resources available in the cause code and the CCM when then use the second choice in the RL.
Now, this is a service parameter "Stop Routing on User Busy Flag*" which is set to true, but this only relates to intercluster trunks.
I've tried playing with the few voice hunt command on the router with no luck.
Any advice or pointers.
Thanks

Yes, saw this next to the user busy flag, but the help says this about "Stop routing on unallocated number"
"This parameter determines routing behavior for intercluster trunk calls to an unalloc ted number. An unallocated number represents a dialed directory number that does not exist in a Cisco cluster. Valid values specify True or False. When the parameter is set to True and a call that is being routed to a remote Cisco cluster through a route list is released by a remote Cisco CallManager because of the unallocated number, a local Cisco CallManager will stop routing the call to a next device in the route list. When the parameter is set to False, the local Cisco CallManager will route the call to the next device."
Again, it talks about intercluster links, I will try on Monday.
Thanks

Architectural question for CCM failover WAN best practices

I have a client that has a large CCM cluster in Texas. Approx 2000 phones register here over the wan from branch offices, HQ, etc. In Milwaukee, there is a call center that is going to go in for about 200 agents 24/7 operations. We are looking at the architecture of this design and are wondering if it would be wise to setup another cluster in Milwaukee just for the Call Center, then use Intercluster trunking between the two clusters.
Or, could we just place a (2)subscribers at the Milwaukee location for a DR between the two sites? (Texas and Milwaukee).
The WAN backbone is MPLS, so we could configure multiple T's back to the data center, etc. The problem we see is if what the Texax CCM cluster falls down... what happens then? Will the CallCenter be able to function?
Any good advice, simple is better.
Thanks much!

What type of callcenter is it? IPCC Enterprise or Express?

Intercluster trunk restrictions

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