I am new to voice and trying to visualize how inbound dial peers are matched with outbound dial peers and need some verification.
I have read the cisco documents on dial peers and still need some clarification.
We have a CUBE route receiving SIP calls and my questions are pertaining to this type of configuration:
In our CUBE router, there is only a single pots dial peer and this was configured for testing FAX, all other dial-peers are voip.
The order in which the dial peers are configured is important in that the first dial peer matched top down is the one used, correct?
Do all SIP calls destined for our Call Manager have to match one of the voip dial peers, or is an inbound dial peer not neccesarily needed?
Inbound dial-peers can be matched by calls that are sourced from the Telco or our internal network (call manager), correct (any call inbound to the router)?
Is the "called number" always considered "inbound" either from the telco, or our internal network (call manager)?
And "calling number" can be sourced from either telco or internal network?
Below is an example of a dial peer configured in our CUBE, not the test FAX I mentioned earlier, this is for Xmedius:
dial-peer voice 1 voip
description Test FAX dial peer
incoming called-number 1234567890
dtmf-relay rtp-nte digit-drop
This dial peer meets the requirements of being valid (incoming called-number), but I don't see where the incoming called number goes from here, shouldn't it go to a session target of the Xmedius server?
You got a lot of questions. So lets break it down a bit.
"Calling Number" is the ANI or the number of the person placing the call (doing the calling)
"Called Number" is the DNIS or the number of the person your are trying to reach. (the person being called)
Either one can be used to match the inbound or the outbound dial peer. I bet that confuses things.
The order of the dial peers has no influence on how or if it is selected.
Also, the command # "Show dialplan number XXXX" will display which dialpeer matches the number that is being processed.
Will have to finish this tomorrow.
Thank you for the reply.
So, help me understand this, according to the Cisco documentation, it is important to understand the direction of the dial peer is in relation to how the router sees it.
So if there is a "called number" configured in the dial peer, won't the direction always be inbound to the router?
If I am sitting on my network, calling 1234567890 and there is a dial peer configured with "called number 1234567890", would that not be a call that the router would see as inbound to hit that dial peer?
on the dial peer shown in my earlier post, that has to be inbound in the direction from someone outside calling the Xmedius FAX machine. The "called number" is in our block of numbers, so it is inbound. But, how do I determine where the call goes from there, the way this is configured?
Doesn't there need to be a place to send the call configured in the dial peer?
Lets start over. whith a few basic rules or principles.
A command to used to view which call legs a active call is using is :# show voice calls status the last column "dial-peers" is listed as follows 1/2 . which means 1 is the inbound dial peer and 2 is the outbound dial peer (inbound dial-peer/outbound dial-peer).
Here are the matching priorities of dial peers:
Inbound dial peers
Inbound dial-peers are used only for controlling parmaters on the inbound call leg (ie codecs, dtmf-relay, etc) . they are never used for outbound routing. There is also a default dial-peer 0 (zero) that can not be configured that will be used when nothing else matches.
Outbound dial Peers
Outbound dial-peers are used to control the routing and various parmaters (digits manipulations, codecs, dtmf tones, etc) in the outbound direction.
The "sh voice call status" was very helpful on one of our gateways.
On the SIP trunk I see no active calls, even though I know there are calls established.
Is the above for ISND and is there a similar show command for SIP (that shows the inbound and outbound dial-peers being used)?
No this command "show voice call status" is not limited to isdn, it show all calls (Pots & VoIP). The SIP call are using dial-peers. They will show up. Something else is wrong.
Can you verify that calls are hitting your router?
this will help you to determine if the SIP traffic is hitting your router.
use the "debug ccsip"
and it will show the SIP traffic comming with its requested port and media request.
also the "debug voip dialpeer"
and it will show inbound dialpeer that was matched, any translations that took place and then the outbound dialpeer that matched.
The SIP trunk is working and calls are coming inbound and
it is working ok.
Thanks for the tip, I will look into that.
Is there a blueprint for the components needed in Call Manager and a remote gateway to establish a sucessful call?
For example, if I had a hub site with call manager and a branch gateway connected via MGCP,
Call Manager would require
Media Resource Group
Calling Search Space
Translation Pattern (if needed? not always needed?)
Remote Gateway would require
Serial Port config (for PRI)
I have never seen a blueprint for deploying Call Manager. It all depends on your particular situation. Are you familiar with Solution, Reference Design documents from Cisco? It will explain the purpose and how to implement each component you mentioned
They cand be found at www.cisco.com/go/srnd
I you need help understanding any of concepts i can probably help.