08-28-2009 10:24 AM - edited 03-15-2019 07:33 PM
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
Solved! Go to Solution.
09-08-2009 01:06 PM
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
08-28-2009 10:41 AM
Yes, you can have redundacy in this manner, however you will need to configure standby CME to send calls to active CME via a lower preference voip DP.
HSRP is not necessary.
Note if you never done this it may take a while for the configuration to be perfect, and you will notice that in practice, router almost never fail.
08-28-2009 03:09 PM
Hi,
Can you please elaborable, may be tell the related command if possible for
"you can have redundacy in this manner, however you will need to configure standby CME to send calls to active CME via a lower preference voip DP".
1.My also worry is about incoming calls from E1 on standby CME, will they not get busy tone, or these will go to IP Phones without any problem.
2.What about voicemails for such calls if phone doesn't answer.
Thanks
07-01-2019 10:11 PM
Hi Paolo,
Could you have look at my post and comment?
Saif
08-28-2009 11:37 AM
Use the 'secondary' keyword on the ip source-address command.
It looks like this:
CME1:
interface 1
10.10.10.1
telephony-service
ip source-address 10.10.10.1 secondary 10.10.10.2
CME 2:
interface 1
10.10.10.2
telephony-service
ip source-address 10.10.10.1 secondary 10.10.10.2
This way all the phones register to the secondary by default.
-nick
08-28-2009 03:01 PM
So Nick, all the phones will register to CME1 first, if CME1 fails it will register to CME2. >> Ok
Scenario: The phones are registered to CME1:
Q1.For outgoing calls, will it use CME1 E1 PRI for all calls, or it will load-balance across gateways?
Q2.What happens to the calls coming on CME2 E1 PRI(refer: my setup explaination initially), will they go perfectly to IP Phones directly(as the IP Phones registered to CME1).
Q3.If in Q2 end phone doesn't answer, what about voicemail, will the go to CUE of CME1 or it will try to look for CUE on CME2?? >>Note CME1 has CUE , CME2 has no CUE.
Q4.Suppose CME2 ethernet down(CME1 is still up), what happens to calls coming in from E1 on CME2.
08-28-2009 03:08 PM
Good questions.
1 - this depends on your dial peer configuration. If multiple dial peers have the same destination-pattern configured, it will load balance. If you prefer one to the other, you can place a 'preference' statement on the dial peer to have one used first.
2 - This CME will have dial peers pointing towards CME1. When the phones register to CME2, the calls will come directly to the phones.
3 - The whole idea of redundancy revolves around if the ethernet or router goes down. In this case, CUE is completely inaccessible. You would not have voicemail in this failover scenario, unless you still had IP reachability to CUE. This is unlikely if you do not have reachability to CME. Even if you had two CUE's, you would not have centralized voicemail. Users would not be able to check the voicemail from the other CUE while in failover, etc. This is a product for this, the UMG. It's unified messaging - but it's more for widely distributed CUE modules and isn't really designed for just 2 CUE modules.
4. Calls that come in to CME2 if the ethernet is down will fail. This is of course, unless you place some type of TDM link (FXO/FXS, T1 PRI) between the two CME in case you would like additional redundancy if the ethernet goes down. This would also solve the CUE problem.
Hope this helps.
-nick
08-28-2009 03:33 PM
Hi Cick,
My setup consist of CM1(1E1+CUE), CM2(1E1). Telco sends incoming call in round robin
For 2. If phones registered to CME1, i need help to clarify, what happens to incoming calls on CME2(via E1), since Telco always send calls to round Robin.
What happens to voicemail , in same case when phones reg to CME1, and few incoming calls come to CME2.
For 4., won't the calls hairpin back to Telco & come as incoming to CME1, >> this scenario I tested with CCM server cluster ans two voice gateways, only thing we will lose the original CLID.
>>So similarly will it work for CMEs setup also??
Thanks
08-28-2009 04:30 PM
2 - You create dial peers on CME2. Half the calls will come directly, the other half will come either H323/SIP from the other CME. This is not a problem.
As long as voicemail is reachable, this scenario will have no impact on voicemail.
4 - Generally telco will not allow you forward your own number back out to the PSTN. It may be possible for your telco to re-route the call if we send 'temporary failure' or 'no route to destination' when the ethernet is down. This is telco-dependent. You would not lose CLID in this scenario.
Please rate the helpful posts so that others can identify helpful information.
-nick
08-28-2009 04:44 PM
Hi Nick,
Sorry to bother you again.
For 2. I think you mean, IF phones register to CME1, then calls from E1 on CME2 will go straight to phones since here CMe is acting as a pur H323/SIP Gw.
Is that correct?
For 4. Do I need to give any specific CLI command on CME, to send 'temporary failure' or 'no route to destination'
Thanks
08-28-2009 05:14 PM
2 - phone register to CME1. If calls come on E1 on CME1 they go straight to phones. If calls come to CME2 it will travel H323/SIP to CME1, and then to phones.
4 - You will automatically disconnect with 'unallocated / unassigned number' I believe. This should be enough for your telco to reroute.
-nick
08-28-2009 05:17 PM
Gor 2. If calls come to CME2 it will travel H323/SIP to CME1, and then to phones. >> Is this the default behaviour, or I need to do some special config for this.>>> I think I need to isn't it? any commands?
08-28-2009 06:24 PM
Yes, there are commands. If you do not know anything about SIP/H323, dial peers, or call routing, I would suggest some reading on Cisco.com
Suggested searches:
H323 gateway configuration
SIP gateway configuration
Understanding inbound and outbound dial peers
Communications manager express administrator's guide
-nick
08-28-2009 06:36 PM
Hi Nick,
I know about H323/SIP 7 what you are mentioning.
But how will you forward call from CME2 to CME1? Don't you require IP TO IP GW IOS for that?
I have worked on CCMs & VGs but first time on CMEs.
Thanks
08-29-2009 05:40 AM
Hi Nick,
Sorry I think you are indication the pots/voip dial peer config to take the calls from CME1 to CME2.
Correct?
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