MULTI SITE dial plan for ad hoc appearance

Unanswered Question
Jul 12th, 2010

I'm  looking to do something which I think should be simple with multisite.  I have a customer whohas different resaurants (some local some long distance), but key here  is that the main site will be managing phones calls for all  restaurants.  So, the opperator would like to be able to place a call and  'appear' as if they were calling from other sites by the press of a  button.  They have standard analogue phone lines, so if main site opperator presses button  "site C" then that would have to be some sort of dial-peer or  translation to 'site C' and then 'site C' would need to send the call  out its own FXO.

Anyone have any recommendations on  this

I have this problem too.
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Brandon Buffin Mon, 07/12/2010 - 13:01

One way to do this is with a prefix or site code. So, you could have users dial 9+number for Site A, 8+number for Site B, 7+number for Site C, etc. You could also have a longer site code such as 101+number, 102+number, etc. You would need to configure dial peers to make this work such as:


dial-peer voice 1 voip

destination-pattern 8T

Site C

dial-peer voice 1 voip

incoming called-number 8T

dial-peer voice 2 pots

destination-pattern 8T

port 0/1/0

There are certainly other ways to configure the dial peers such as using translation rules to strip the 8, change the 8 to a 9 to use existing outbound dial peers, etc. This is just one way to accomplish it.

Hope this helps.


Christian Isla Mon, 07/12/2010 - 14:43

Thanks for your quick reply.  I will start to configure as it seem to be going down the right track.  But there is one other piece that i *must* have for this customer, and that is the *automatic* portion.  That is I guess I need to make the prefix automatic.  Physically dialing a prefix is out of the question here, but if it happened in the back end that's ok.  So when I press the button "SiteC" I don't have to dial any differently then I would if I was sending a call off the local system UC if I pressed button "HQ".  This is really the part I'm stuggling with.  I'm not sure if I have to make it an auto "A" speed dial, or a CO trunk (can I even make an H232 peer ackt like a CO trunk?)

any takers?

David Trad Mon, 07/12/2010 - 19:12

Hi There,

The suggestion from Brandon would work quite well, other then that you could do the following, but this requires some planning and foresight:

At the head office you would create dial-peers that point to the remote site system, then you would create dialout dial-peers for specific numbers to also point out via the remote site, for instance, lets play out the following scenario;

Site A (Head Office):

HQ resides in the 555 zone

Site B (Remote Site):

Remote site B resides in the 554 calling zone

You would create a dial-peer on the HQ system that when they press "0" and then dial 554XXXXXX numbers it would automatically point to the Remote site

You would create these dial-peers for all remote sites, this way the calls first travel On-Net from system to system, then they hand of locally at the remote site on what ever trunk lines are configured there, this would give the impression that all calls are originating from that remote site, but are in fact coming from the HQ site.

I hope I explained myself well enough for you to understand it, I have a bad habit of confusing people at times with my explanations.



Brandon Buffin Tue, 07/13/2010 - 05:27

Good suggestion, David. If you want to be able to dial any number out of any location, you could create a prefix for each location and add a speed dial that dials this prefix. So, make a speed dial labeled "Site C" and have it dial 7, for example. After the user presses "Site C" they can dial the number normally and the call will be sent out the Site C gateway.

Hope this helps.


Christian Isla Wed, 07/21/2010 - 12:51

Hi Brandon, I see were you're coming from and I totally understand and agree that this should work.

I'm having some difficulty in getting the call from HQ site to SiteC.

Here is the config at HQ:

voice translation-rule 5511
rule 1 /^8383/ //
voice translation-profile site-select
translate called 5511

dial-peer voice 5511 voip
description Outcall through SiteC
translation-profile outgoing site-select
preference 1
destination-pattern 8383[2-9]..[2-9]......
session protocol sipv2
session target ipv4:
voice-class sip dtmf-relay force rtp-nte
voice-class sip outbound-proxy ipv4:
dtmf-relay rtp-nte
codec g711ulaw
no vad

When I press the SiteC button, i get a pause so I dial the number and I can tell that that part works.  I have debugs of the SIP INVITE to SiteC with just the proper 10 digit number, but I never see that SIP INVITE come in.  I'm struggling to determine if an ACL at either site is actually blocking this.. can you give some insite?

Brandon Buffin Wed, 07/21/2010 - 12:56

Do you have the inbound dial peer configured at Site C? If you run "debug voip dialpeer" at Site C, do you see the call hit the proper dial peer?


Christian Isla Thu, 07/22/2010 - 06:55

I was running 'debug ccsip messages' at SiteC and I never see the INVITE come in.  Should I not see the INVITE come in?  I can see the INVITE go out from HQ and it's going to SiteC which is in my IPSEC VPN and reachable.  I also have dial-peers 'H.323' for site to site extension dialing which is working perfect through the VPN.


Brandon Buffin Thu, 07/22/2010 - 07:44

Can Site C dial the number that you are dialing from the HQ? Do you have an inbound dial peer configured on the Site C gateway? Can you post the config from Site C?


Christian Isla Thu, 07/22/2010 - 08:24

sorry, no i don't have an inbound dial peer in Site C.  I though that the UC would just accept the INVITE and send it.  what should the inbound dial peer look like?  something like this?

dial-peer voice  5512 voip
description Trunk From HQ
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number [2-9]..[2-9]......
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

and yes, i can dial the number i'm calling from SiteC.


Brandon Buffin Thu, 07/22/2010 - 08:38

Shouldn't need "session target sip-server". Otherwise, looks good. Does site C dial a prefix such as 9 for outbound calls. Is there an outbound dial peer at Site C that will match this call with no 9?


Christian Isla Thu, 07/22/2010 - 13:46

So I believe I have another prblem here.  My INVITE is just not making it to siteC.  It may not be going through my VPN.  I've created the siteC inbound dial-peer and I've used 'debug ccsip messages' and 'debug voip dialpeer' and nothing shows up.  At HQ I can see that the INVITE is going out to (SiteC) but I never see any reply come back into HQ from SiteC.

Iv'e even set it up a dial-peer as "838333." where 33x is an extension at SiteC (i dialed 8383335 and INVITE went out but nothing received at SiteC.  my inboung at SiteC was changed to reflect this.).

My assumption at this point is that either my source address is not correct for my VPN out the HQ. or not correct somewhere, but either way, I don't thing IP is routing properly at this point.

Brandon Buffin Thu, 07/22/2010 - 13:49

Sounds like Site C may not have a route back to HQ. Can you ping from Site C to HQ? HQ to Site C?


Christian Isla Thu, 07/22/2010 - 14:00

i take it my HQ source sould be my voice vlan 100 ( or should it be my CUE at

Brandon Buffin Thu, 07/22/2010 - 14:00

Could also be that SIP is sourcing from an interface that does not have a route to/from Site C. This can be controlled as follows:

voice service voip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0

From this doc:

Hope this helps.


Christian Isla Thu, 07/22/2010 - 14:05

haha... i just used that to bind it to my vlan 1 (cause I now it pings) and I'm finally seeing my INVITE come in... I'll work on this for a while now..thanks.



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