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cmonks
Beginner

UC500 Multi-Site 3 digit dialing?

I will be setting up a UC560/UC540 multi-site solution, and would like to know if anyone has setup multi-site with three digit extensions. The two sites currently have two digit extensions, i would like to make them three digit extensions and be able to simply dial extensions back and forth. For example Site A is 1xx, Site B is 2xx.

I really want to avoid the CCA way of doing it, 81xxx, 82xxx, etc, its just too complicated to have that many dialing digits, and i need the auto attendant at Site A to be able to have callers dial extension 1xx and 2xx without having to dial 81xxx, 82xxx, etc.

I can do this all via CLI, just looking for anyone who has successfully (or unsuccessfully) set this up, and if any problems were encountered setting up the 1xx and 2xx dial peers.

Thanks,

12 REPLIES 12
danplacek
Enthusiast

I've done it many times with very few issues.

Just be sure to create a good dial plan ahead of time.

Ex:

1XX = Site A

2XX = Site B

3XX = Site C

6XXX = Direct Voicemail

7XX = Park Slots, Meetme Bridges, and other `system` extensions

9T = Outside Line

etc...

I've generally found that using H323 for the inter-site trunks to be easier and less problem-prone than SIP.

I've also gotten presence to work between two sites by setting their presence servers to eachother. (For blf-speed-dials)

-Dan

Please rate useful posts.

Great, sounds promising. Can you by any chance send me your H323 and dialpeer config for this? I was sure that Presense was going to be a pain (or not possible), thats good to hear that you have it working, can you send me that config also?

Also a question on the extensions, you say to use 7xx as system extensions, but wouldn't those need to be in the site range as well? I couldnt have a 7xx park slots at both sites, because it couldn't tell the difference between them. Wouldn't I need 1xx park slots and 2xx park slots? same with voicemail, AA, etc?

For the extensions -- it depends on what you want to make accessible to other sites. Voicemail for each site should be put in their extension range obviously (Usually X99 or something). I have had issues picking up park slots at another site and that has not been a critical feature for us, so I have just kept park slots local.

dial-peer voice 5000 voip

destination-pattern 1..

session target ipv4:172.16.1.2

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

dial-peer voice 5001 voip

session target ipv4:172.16.1.2

incoming called-number 2..

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

presence

sccp blf-speed-dial retry-interval 70 limit 20

presence call-list

server 172.16.1.2

watcher all

allow subscribe

Be sure to add "allow watch" to any DNs you want to be monitored.

blf-speed-dial should be used. (Not speed-dial, watch, or monitor)

Under ephone: blf-speed-dial 1 101 label "Name"

To see current presence subscriptions: sh presence subscription

If you do `debug ccsip messages` you can see the presence messages exchanged.

For the connection between the boxes -- I have had success with: Tunnel provided by other firewall, MPLS, IPSEC tunnel, and GRE tunnel (no encryption).

If you don't already have something in place, GRE is the easiest, but offers no encryption.

I like GRE because it is really quick and easy to both setup and troubleshoot; IPSEC can always be layered on top of it after you have things working.

Configuring a GRE tunnel: https://supportforums.cisco.com/docs/DOC-2569

-Dan

Please rate useful posts.

Hi Dan,

Thanks for the config examples, I'm currently trying to get two UC560's showing respective BLF's from each other on SPA525G2 phones... Setup the presence config as you suggest above, but don't seem to be getting a win.  

Is the BLF indication possible on SPA525 phones? or only on the 7xxx series phones?

I'm seeing the SUBSCRIBE messages, but also seeing a 404 not found message block following it in debug. 

Wondering if this is only possible on certain phones and I'm wasting my time, or if it 'should work' on SPA phones and where to go to next to resolve.

Thanks. (Hope you still read this thread) :)

 

Andrew,

 

It should work with any (Cisco) phone.

Do you have "allow watch" enabled on each of the directory numbers (ephone-dn's)?

 

-Dan

Hi Dan,

I believe so.

I'm using a single phone on each system as a test. System one (IP: 10.1.2.251), system two (10.4.1.2).

On system one I've a dn:

ephone-dn  16  dual-line
 number 8020 secondary XXXX8020 no-reg both
 pickup-group 1
 label Andrew - 8020
 description Andrew
 name Andrew
 allow watch

On system two:

ephone-dn  595  dual-line
 number 4013 no-reg both
 pickup-group 1
 label TestBLF
 description TestBLF
 name TestBLF
 allow watch

If the ephone for each I've added similar to: 

 presence call-list
 blf-speed-dial 1 8020 label "Ext Name"

 

'sh presence global' is displaying:

Presence Global Configuration Information:
=============================================
Presence feature enable            : TRUE
Presence allow external watchers   : TRUE
Presence max subscription allowed  : 100
Presence number of subscriptions   : 0
Presence allow external subscribe  : TRUE
Presence call list enable          : TRUE
Presence server IP address         : 10.1.2.251 (IP of 'other' system)
Presence sccp blfsd retry interval : 70
Presence sccp blfsd retry limit    : 20
Presence router mode               : CME mode

 

On the phone display, it shows the label with an "I" with blue background icon.

Dial peers - I use SIP, noticed you setup using h245 (doubt this will effect it)

dial-peer voice 4000 voip
 description ***BLF-Test***
 destination-pattern 4...
 session target ipv4:10.4.1.2
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

Also, with Dial peers you have a pairing "incoming called-number" peer which I don't have, never needed it for calls to work - but I wonder if this would effect anything with the BLF subscribing?

 

I noticed overnight that the 'subscriptions had dropped to 0. Even the 'local' phone had gone away. If I 'restart' a phone, I get the following in 'sh presence subscription' on the system with that phone.

Presence Active Subscription Records:
=============================================

Subscription ID         : 67
  Watcher               : 8020@10.1.2.12
  Presentity            : 4013@10.1.2.251
  Expires               : 3600 seconds
  Subscription Duration : 2515 seconds
  line status           : unknown
  watcher type          : local
  presentity type       : remote
  Watcher phone type    : SCCP [BLF Speed Dial]
  subscription type     : Incoming Indication
  retry limit           : 1

 

Both nothing seems to allow it to cross-subscribe. 

 

Note: I am using a SIP trunk for calls out to the world, but as I understand it, the sip-ua section is not related to this cross-system SIP comms. Perhaps I need to switch to h245 to get success ?

Appreciate your comments and insight :)

 

 

 

 

 

Interesting.

Firsts things first - the dial peer you posted (4000) is H.323.

If "session protocol" is not specified on a dial-peer on an IOS router, H.323 is the default.

 

In any case,

 

1. Could you post (sanitized as necessary) the "presence" and "voice service voip" sections from your config?

2. Do you have a trace of the SUBSCRIBE messages? (terminal monitor & debug ccsip messages)

3. Just to verify - both systems have their presence server set to each-other?

Hi Dan,

(1) Presence block:

presence
 sccp blf-speed-dial retry-interval 70 limit 20
 presence call-list
 server 10.4.1.2
 watcher all
 allow subscribe
 presence call-list

(and the other system has the same, with the 'other' systems IP)

 

And on each the voice service blocks are as follows:

(Note: We only do outbound SIP calls, and in-bound ISDN(BRI) so there is no requirement to allow/trust the proxy in the list.

voice service voip
 ip address trusted list
  ipv4 10.0.0.0 255.0.0.0
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 sip
  registrar server expires max 3600 min 3600
  localhost dns:<sip-provider-domain>
  outbound-proxy dns:<sip-provider-proxy>
  no update-callerid
  sip-profiles 1000

 

 

 

Again this morning I noticed nothing in 'show presence subscription". So to force an attempt, I've "restart" my ephone and it produces this:

 

PBX01(config-ephone)#restart
restarting 4055.39A2.A732
PBX01(config-ephone)#end
162231: Apr 28 08:46:51.617: %IPPHONE-6-REG_ALARM: Name=SEP405539A2A732  Load=7.4.9a Last=Hard+Unknown
162232: Apr 28 08:46:51.821: %IPPHONE-6-UNREGISTER_ABNORMAL: ephone-7:SEP405539A2A732 IP:10.1.2.12 Socket:11 DeviceType:Phone has unregistered abnormally.
162233: Apr 28 08:46:51.821: %IPPHONE-6-REGISTER: ephone-7:SEP405539A2A732 IP:10.1.2.12 Socket:12 DeviceType:Phone has registered.
162234: Apr 28 08:46:52.545: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SUBSCRIBE sip:4013@10.4.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.251:5060;branch=z9hG4bK2E8F0683
From: <sip:8020@10.1.2.251>;tag=4D46C0AC-F74
To: <sip:4013@10.4.1.2>
Call-ID: 21C6D0F6-EC6611E4-9948C8DE-ECD14605@<sip-provider-domain>
CSeq: 101 SUBSCRIBE
Max-Forwards: 70
Date: Mon, 27 Apr 2015 22:46:52 GMT
User-Agent: Cisco-SIPGateway/IOS-12.x
Event: presence
Expires: 3600
Contact: <sip:10.1.2.251:5060>
Accept:  application/pidf+xml
Content-Length: 0


162235: Apr 28 08:46:52.601: //0/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 10.1.2.251:5060;branch=z9hG4bK2E8F0683
From: <sip:8020@10.1.2.251:5060>;tag=4D46C0AC-F74
To: <sip:4013@10.4.1.2:5060>;tag=SD20sg299-17049706-1430174812574
Call-ID: 21C6D0F6-EC6611E4-9948C8DE-ECD14605@<sip-provider-domain>
CSeq: 101 SUBSCRIBE
Content-Length: 0

 

 

What concerns me is the Call-ID containing the outbound SIP domain, but I'm assuming that it just uses the "localhost dns:" domain from the voice servcie voip section here.

I also had 'debug ccsip messages' enabled on the other PBX and when the send subscribe happens, I don't see anything on that other system. So perhaps it's not sending the request where I think it should?

Also, in the 'sip-ua' block, should I need "presence enable" here or is this only the SIP client that talks outside to the proxy - and it doesn't play a roll in the cross-system presence?

 

For fun, I recently tried pointing the two systems an a third external "presence" server (running OpenSIPS).

It never sees a network connection from the PBX's.

*bangs head on desk*

Can't see what's causing the messages to never leave the PBX and head where I expect them to go. So I'm back at "UC5xx - Cisco's most frustrating product ever invented"

I believe your problem lies with the "outbound-proxy" option you have enabled.

 

From Cisco's Docs on the matter:

You can use the outbound-proxy command in voice service SIP configuration mode to specify outbound proxy settings globally for a Cisco IOS voice gateway. You can also use the voice-class sip outbound-proxy command in dial peer voice configuration mode to configure settings for an individual dial peer that override or defer to the global settings for the gateway.

 

Specifies the SIP outbound proxy globally for a Cisco IOS voice gateway; all SIP dialog-initiating requests are sent to the SIP server.

 

I suspect you may need to create a voice-class specifically for your inter-site dial peers to override that setting.

 

-Dan

Hi Dan,

 

Whilst playing a bit yesterday, I changed my site to site dial-peer's over to sip, and it of course required adding a voice-class sip outbound-proxy to each to ensure that those calls went to the correct trunk. I was hoping this might also effect the subscribes, but it appears to not help.

I can see that one can set-up multiple username/credential's and registrar's in the sip-ua and voice service voip|sip blocks respectively. But that's more about defining redundant / alternate main SIP trunk destinations.

The only thing I can think of is try and remove the outbound-proxy from the global sip config, and specifically add voice-class sip outbound-proxy to my main SIP trunk dial-peers.  Only question with this is, will registration still work... I think this is my next step - but unfortunately I have to wait till I can potentially 'break' my sip trunk to prove it :) 

Seems this stuff would work easily if no SIP trunks where involved, like if my in/out bound lines where just FXO or BRI/PRI.  But bringing global SIP into play just makes it confused about where to send things you want.

Thanks for the input.

Success!!

Okay so removing the global outbound-proxy did the trick. Outbound calls via the ITSP trunk still work no issues, and presence appears to be registering now.

Next is getting the buttons to light-up - Currently I have a test blf on a 525 screen button so I can see a blue "I" or red "B" indicator, but the light in the button doesn't show. which makes it useless for a 500S.

Also I did need to have the "presence enabled" in the sip-ua section to allow the receiver to run on the phone system and accept the remote incoming subscriptions.

This is progress!