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Beginner

Simultaneous Ring on two different extensions

Hi

I'm trying to set up simultaneous ringing on two internal extensions using a Remote Destination Profile. I can get the simultaneous ringing to work fine if I use my mobile or home number for the remote destination, however, if I substitute this external number for an internal extension it does not work?

What am I missing here?

16 REPLIES 16
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Beginner

are your phones in a partition? do you have the css that includes this partition under the RDP rerouting css?

also, are you doing this as a voice lab or something, presumably your not just overcomplicating a shared line configuration?

please rate helpful posts

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Beginner

The CSS definitely has access to the partition which the extensions are in.

My reason for doing this is because we are setting up a "Proof of Concept" for MS OCS simultaneous ring. This requires that you fork calls down a SIP trunk to the OCS mediation server as well as to the handset. From what I've read elsewhere setting up a remote destination is the correct way to go for this.

If I am overcomplicating things please let me know!

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mmm, i've never tried this on an internal number, maybe it skipps the internal routing logic.....

can you try puting the remote destination number to a number matched by a translation pattern which changes the number back to the extenstion number.

make sure the rerouting css and pt are correct;

I know it sounds a bit loopy but unless im missing something obvious here it could be that it has to match a route pattern or translation pattern....

Highlighted
Beginner

Hi!

I think, you may acheve the same result by placing your extensions in a line group. After that you may assign "broadcast" distribution alorithm to this group and assign this group to some Hunt Pilot. Also, this method does not require additional DLU's, opposed to RDP method.

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OK, here's the result:

If I configure a translation pattern for my mobile number to be masked with an extension number (1302), and dial the mobile from any extension the extension (1302) rings. (As you'd expect) Unfortunately, if I configure my mobile as the RD number on an extension, when I dial the Ext the mobile no longer rings. If I then remove the RP and dial the ext the mobile starts ringing again.

I did wonder about using a hunt group with the "Broadcast" algorithm, however, I've never been able to get any more than one extension to ring at the same time?

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Please see attached file.

First you create a line group with BC disribution algorithm, then you add this group to a hunt list and then you assign this hunt list to a Hunt Pilot and voila!

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Hi James,

Mobile Connect (SNR) algorithm to a Remote Destination (RD) will NOT work for internal extensions. SNR will work only if the RD is across an ICT / SIP trunk / PRI Gateway.

When the digit analysis is done on CallManager for the RD, the Device Manager will identify the process through which the call needs to be sent to. If it is an MGCP PRI gateway, it would MGCP9n (or something along those lines). If it is a routelist, then the process would be routelistcontrol.

When you configure RD as internal extension (with translation pattern or not), the final destination would be LineControl (for a line on a phone registered to CallManager) - so SNR algorithm will reject this proposed call to an internal RD.

Hope this explain it.

- Sriram

Please rate helpful posts !

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OK, this does make sense, however, I'm still getting this issue:

1. I have a SIP trunk configured connecting to the OCS Mediation Server.

2. I have a RP configured to point 7XXX to the SIP Trunk. (7XXX is not configured anywhere on the UCM)

3. If I dial 7911 the call correctly routes to the Mediation Server and OCS rings accordingly.

4. I have an extension (6911) with a RD of 7911 but when you ring the extension the RD (OCS) does not ring?

I only have one Partition configured as this is lab kit so all of the relevant CSS's are allowed access to everything.

Is it possible I have something missing on the SIP Trunk configuration? I have attached the outbound calls section, do I need to add in CSS's here?

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Hi James,

The Rerouting CSS on the RDP will be used to route the call to the RD. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsmobmgr.html#wp1369159

- Can you please check if the Rerouting Calling Search Space can access the RP 7XXX ?

- I would suggest setting the 'Delay before dialing' timer on RD to 0, so that the call to the RD is extended immediately when the RDP starts ringing

- SNR call to RD will take into consideration any applicable Application Dial Rule (Call Routing > Dial Rules > Application Dial Rules). Make sure that the 7911 number is not being altered by any application dial rule

Let me know if this helps.

- Sriram

Please rate helpful posts !

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Hi Sriram

Thanks for your assistance with this. As you suggested, I already had the delay time set to 0 and I only have 1 CSS which has access to my 1 and only Partition, and this is configured on the RD Reroute CSS (as well as everywhere else!) so I can't see this being the issue.

I do have a single App Dialling rule but it's for a completely different pattern (+44207158) so again I don't think this would be relevant.

Is there a way of seeing if my digits are being passed across to the SIP Trunk? At least that way I could prove to the Wintel team that the call is being passed to them?

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- Set the CallManager traces to detailed (including SIP call processing and SIP stack traces):
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtml#calm

Please ensure that SIP trunks are not processing a lot of calls when SIP Stack Trace is enabled - can cause performance issues.

- Make a test call and collect the following logs from ALL servers using RTMT, during the time period of the call :
    - Cisco CallManager

Attach the logs to this thread.

Let me know :
- Calling party
- Called party

- RD
- Time stamp of the call

- Sriram

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- Calling party     6390

- Called party     6911

- RD                  7911
- Time stamp of the call     16:39(ish)

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Hi James,

I don't see the call in the logs that you collected.

- How many Call processing servers do you have ?

- Did you set the CallManager traces to detailed on ALL servers ? Did you collect the logs from all servers ?

As a quick check before you upload the logs, you can download the logs, open the ccm*.txt files, and search for dd="". As example, if you dialed digits 6,9,1 and 1, you would search for dd="6911". If you find a hit for that for the time stamp you called, upload the logs - I'll take a look at it.

- Sriram

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OK, think I've got the correct trace:

Time of call is 09:16a

All of the other information is the same.