CME Redundancy?

[ra_jeshkalra_2]

In my setup I have Cisco3845 CME routers(C3845-VSEC-CCME/K9), one with CUE module, other with non-CUE(The customer is o.k. not to have voicemails if the CUE-module CME router fails). Both the CME are on same localtion/Lan.

Each router will have one E1 line from PSTN.The Telco sends call to both CMEs in round robin.

How can we achieve CME redundancy in this setup?

1)configuring Redundant Router: Reference CME Admin Guide-Page 117

OR

2)HSRP

>>For CME Admin guide, it says on page 144 For configuration information, see the "SCCP: Configuring a Redundant Router" section on page 117.

But under the section:SCCP: Configuring a Redundant Router as a pre requisite it mentions following:

The physical configuration of the secondary router must be as described in the "Redundant Cisco Unified CME Router" section on page 104. >> which means fxo port with Splitter:

2)HSRP>> I heard some issues with phone registration to standby router if active router fails. Also what would happen to incoming calls coming from PSTN>> will they be able to go to IP Phones, since this router is in Standby mode?

Bearing in mind that the Telco is sending calls to both CMEs in round robin, since each CME has one E1 line from Telco.

Thanks

[paolo bevilacqua]

What a shame, seeing so much insisting questioning, many excellent answers from a certified professional, but not even a post rated.

Some people think they're are entitled to have everything for free, and even cheaper.

[Wilson Samuel]

Originally not my question.. however thanks Nick and Paolo for excellent in depth answers.. my 5 Points to both of you guys..

Being a CCIE-V wanna be.. I gain a lot.. every day.. with expert answers like you guys!

God Bless

Sam Wilson

[Nicholas Matthews]

PSTN--PRI--(pots dial peer)--CME2---(voip dial peer)--H323---(voip dial peer)--CME1--(IP phones)

No need for IPIPGW. It's just a simple voip dial peer.

(thanks for the ratings, Paulo)

-nick

[ra_jeshkalra_2]

Hi Nic,

thanks for your help so far.Just few question:

1.Is it a must for more than two party conf. call I should create OCtal Dn, I created dual-line DN, and the call gets dropped if I want to involve the third party.

Thanks

[Nicholas Matthews]

If you want to have more than 3 people in a conference you need to configure hardware conferencing. This will require DSPs. Take a look at 'Configuring Conferencing' in the CME Admin guide.

-nick

[ra_jeshkalra_2]

Hi Nick,

Thanks for the direction.

If I have to configure fax on FXS port on CME. Is it a must that I control the fax similar to IP Phone by using ephone-dn.

Can't I just configure like a traditional was fax with dial-peers and without SCCP in a normal cisco voice router? The idea is to keep the config simple. I am just looking for sending /receiving one fax at a time.

I just want to implement fax here, like with a normal voice gateway.

Thanks

[Nicholas Matthews · Accepted solution]

The only reason to run MGCP or SCCP on an FXS port is to have increased line-side features (hold, conference, park, etc) or ease of management (usually with CUCM).

In the case of a fax machine, you will never use the advanced features, and with CME SCCP FXS ports are harder to manage. For a CME system, you're correct and should just use normal Router controlled FXS ports instead of SCCP.

This is an example of a working fax configuration:

dial-peer voice 1 pots

destination-pattern 9.T

port 0/0/0

(outgoing pots dial peer)

dial-peer voice 2 pots

destination-pattern 1000

port 0/1/0

(fxs port with fax machine)

-nick

[ra_jeshkalra_2]

Hi Nick,

Thanks for all your brilliant answers.

1.Is it possible to use E1 of CME2 for outgoing calls if the E1 in CME1 is down(CME2 is failiover to CME1). I was trying to use a VOIP dial-peer to transfer calls to CME2 IP to achieve, with 9T as session target:

a)Surprisingly, if I bring E1 on CME1 down(phons still registered to CME1), on pressing 9, there is no dial tone on phones.

b)Also there seems to be clash with dest-pattern on VOIP dial peer. All callers with their CLID starting with 9 unable to call in to CME from PSTN.

2.How to resolve Music on Hold issue between phones internally, currently on hold music is non-continues and garbled( for a sec I hear garbled music then blank space & then garbled again) between phones for internal calls. But MOH works o.k. from/to calls to PSTN. The phones are enabled with moh-multicast command on CME.Also Lan side, I am told multicast is enabled.

How to resolve this issue of Garble MOH music for internal calls?

Do I have to check on IP phone something for registation to multicast group?

Thanks in advance.

[Nicholas Matthews]

1) Yes this is possible.

You would simply create a pots and voip dial peer with the same destination pattern, and a lower preference value for the one you're wanting to use as the primary.

There should always be dial tone on phones. You may mean secondary dial tone? Secondary dial tone is with "secondary-dialtone 9"

b) I would 'debug voip ccapi inout' and figure out your dial peers. You probably have an incoming called-number 9.T or something similar on a dial peer which you don't need.

2) I would use unicast MoH. In this case, you're likely multicasting from both CMEs on the same LAN. If you disable MoH on one you will probably not see this problem.

hth,

nick

[ra_jeshkalra_2]

1)I noted secondary dialtone is as a result of PRI signalling(when PRI up) from the CME1. The moment E1 on CM1 goes down, the secondary dial-tone also goes down, since phones still registered to CME1.

But with the pots, voip config as suggested, how can we get secondary dial-tone from E1 in CME2?

2)what is do be done for using unicast MOH & not multicast MOH. shouldn't it work by default, if I disable multicast everywhere.

>>Instead of disabling multicast in CME2, can't we give a new/different MOH source IP compared to that in CME1, since phones will register to only one CME at a time.

3)For extn mobility:

a)I have create the extn mobility profile and associated to the phone.But while loggin in extn mobility, I get login failed authentication error.

b)my customer has 150 physical ip phones, but CME3845 can support 250. So is it possible to create additional 100 profiles for people who don't have physical phone, but want to login to any of the above phone without affecting their profiles/association with IP Phones.

Basically the customer wants any user can sit/login on any phone.

Thanks

[Nicholas Matthews]

1) This is a common misconception. The secondary dial tone is sent to the phone before anything is ever sent to the PRI. The secondary dial tone is a SCCP message that is sent to the phone, not anything related to the PRI. As long as you have the secondary-dialtone command in, you will have this functionality.

2) This command:

telephony-service

no multicast moh

You may be able to get two different multicast streams from two different CMEs to work together. If you can do unicast I would, just for simplicity. I believe two streams will work for you also.

3) Extension mobility is a bit different. I had to file a bug for this specifically because of the problems with the URL when you fail over. Since both CMEs use the same primary address, the Mobility URL is built off of the same IP, which doesn't help for mobility. The bug ID is this : CSCsy21851. It isn't fixed yet, and won't make it into IOS for a while. The option right now is to manually edit the file so that each CME points to itself instead of pointing at a single IP.

I'm not quite sure about the EM part of it.

hth,

nick

[ra_jeshkalra_2]

Hi Nick,

Thanks for your help so far.

As per the attached config, the ad-hoc and meetme conference was working, but now we get an error message "Cannot complete conference" when we try to do meetme or ad-hoc conference.

I suspect this might be due to introduction of some command under telephony-service section.

Any suggestions.

Any debug I can run?

[Nicholas Matthews]

I would compare your adhoc dial peers against those in the CME conferencing section. Specifically, the use of 'huntstop' and 'preference' in terms of using the same DN for the ad-hoc conferences.

-nick

[ra_jeshkalra_2]

Hi Nick,

What is adhoc dial peer here?

I am using the ephone-dn as Octal for adhoc/meetme conference, I am not using preference and huntstop.

(I want only 8-party conference)

The meet-me and adhoc was working earlier, but suddently stopped working.

The change I did was to bring the pstn incoming calls from pots->pots(ip phones), instead of pots->VOIP dial-peer(session target self CME IP Add->pots(ip phones.

But this shouln't cause the problem.

Also can transcoding cause an issue?

(pls see under telephony-service)

I am getting error "Cannot complete conference".

>.I am using ephone-dn 198 & 199 for adhoc confernce(pls see config)

& 186,187,188 for meetme

thanks

[Nicholas Matthews]

I mean ephone-dn and not dial peers, sorry. I believe you may need to do change the preference/huntstop issue.

-nick