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connection tie-line question

ciscoforum
Level 1
Level 1

HQ PBX has two E&M T1 CAS, connecting to GW's two T1 CAS ports. DID numbers for HQ is not seperated between the CAS ports. There are two remote sites with GW connecting to its local PBX. Now my question is: Is it possible that using connection tie-line can make sure call from Rs1 to HQ always use T1 port 1 on HQ, and call from RS2 to use T1 port 2 on HQ. Remember HQ DID number are in one pool. Dial peer on RS are same. Thanks. Somebody mentioned to me that tie line is like a tunnel that you can define to use which port on the other end for outbound. But it seems that it doesn't work that way.

33 Replies 33

Yes, they have to match both "destination-pattern" in "dial-peer voip", this can be a pattern like 2...

and "destination-pattern" in "dial-peer pots" on the remote router. This must be an exact number like 2101.

so can we configure serveral number for one port on the remote router?

like :

dial-peer voice 1 pots

destination pattern 2001

port 1/0:1

dial-peer voice 2 pots

destination pattern 2002

port 1/0:1

So here both 2001 and 2002 will be sent over port 1/0:1, will this work?

again correct me if I am wrong, this is the real extension number.

Technically you can do that. The question is when do you do that?

If you were doing a trunk connection with strict ds0-to-ds0 mapping, it is wrong.

If you were doing a 'routed' connection where both routers and PBX participates in the dialplan, the above is equivalent to:

dial-peer voice XX pots

destination-pattern 200[12]

port 1/0:1

Note that for "trunking connections" you will have "no forward-digits" or "forward-digits 0", so no digits are passed to PBX.

Just quote yours here to discuss

"If you were doing a trunk connection with strict ds0-to-ds0 mapping, it is wrong"

To me this means the same concept I used to have: user doesn't have to enter the phone number, when they pick the phone, they directly connected to that ds0, this is called ds0-to-ds0 map. Right?so they can not dial the other parties.If they are able to dial the other party then it's not Ds0-ds0 map, right? But I remember you mentioned connection trunk also needs user to dial the number. Please confirm this with me.

I've no problem in discussing this although I'm not sure what are you missing to get this scenario that is actually quite simple.

To me this means the same concept I used to have: user doesn't have to enter the phone number, when they pick the phone, they directly connected to that ds0, this is called ds0-to-ds0 map. Right?

No. They will be connected to the DS0 only after the PBX seizes the trunk, either because of user dials digits, or because the user extension is configured as PLAR or "hot-line" at PBX level.

so they can not dial the other parties.If they are able to dial the other party then it's not Ds0-ds0 map, right?

The DS0-to-DS0 connection is made between PBX, not between users. How the PBX handles the user is immaterial to this, because the router doesn't know about and doesn't care.

But I remember you mentioned connection trunk also needs user to dial the number. Please confirm this with me.

Yes. If I am extension 235 at site R1 and I want to call extension 155 at site HQ, I will lift the phone and dial 155. The router will never 'see' or interpret these digits, it will transport them transparently from a PBX to another.

I guess I am getting there. Thanks for all your help. Really appriciate it.

Quoted here again.

"Yes. If I am extension 235 at site R1 and I want to call extension 155 at site HQ, I will lift the phone and dial 155. The router will never 'see' or intrpret these digits, it will transport them transparently from a PBX to another."

If router never looks at the digits, how the remote router knows it should send to port 1/0:1 or port 1/0:2?

Because the called number, configured in "connection trunk" and "destination-pattern" in the local router, is sent (together with other call parameters) in the H.323 or SIP signaling packet. In case of problems, you can see it with the appropriate debug commands.

Thank for your appreciation and good luck with your project.

I am not clear about this answer.

1. You mentioned before that connection trunk number is not the real extensions. So this number shouldn't be helpful for the remote router to route the call.

2. the destination-pattern on local router only help local router to determine where to send the call. Nothing to do with the remote site. The remote site has to make call routing decision based on it's own dial-peer. But if as you said if the router never looked at the digits, this will never work. Something wrong here. For sure.

I am not clear about this answer.

1. You mentioned before that connection trunk number is not the real extensions. So this number shouldn't be helpful for the remote router to route the call.

It is helpful indeed. This number is configured in the remote router as well:

dial-peer voice XX pots

destination-pattern 2101

port 1/0:1

forward-digits 0

2. the destination-pattern on local router only help local router to determine where to send the call. Nothing to do with the remote site. The remote site has to make call routing decision based on it's own dial-peer. But if as you said if the router never looked at the digits, this will never work. Something wrong here. For sure.

See above. The remote router doesn't look at the digits in-band, but it looks at whole called number in the signaling, as I indicated before.

I think that if you was to setup a mini-lab for testing and learning, the whole matter would become clear to you in seconds.

Quoted: "It is helpful indeed. This number is configured in the remote router as well:

dial-peer voice XX pots

destination-pattern 2101

port 1/0:1

forward-digits 0"

That's where you lost me. Indeed I know H323 and dial peer very well. If router(remote) use this dial-peer to send the call to PBX, then it contradicts with what you said(the router never looked the number).

2. H225 contains the called number. There is never an inband number per say in the RTP stream. (unless u talked about in band DTMF, but this is nothing to do with this for sure.).What I am confused or you confused me :) now was you said: router never looked at the called number versus the router will use port dial-peer to send the call to PBX. Bottom line, if router never looks at the number, then router should never look dial-peer.We know the dial-peer number is used to match the called number in the H225 signaling to route the call.

Quoted: "It is helpful indeed. This number is configured in the remote router as well:

dial-peer voice XX pots

destination-pattern 2101

port 1/0:1

forward-digits 0"

That's where you lost me. Indeed I know H323 and dial peer very well. If router(remote) use this dial-peer to send the call to PBX, then it contradicts with what you said(the router never looked the number).

I said, the routers (both) never looks at the number dialed by the users. Then I said, the router (remote) looks at the number in signaling and that you have configured as destination for the trunks. Including this one, I said it three times :).

2. H225 contains the called number. There is never an inband number per say in the RTP stream. (unless u talked about in band DTMF, but this is nothing to do with this for sure.).What I am confused or you confused me :) now was you said: router never looked at the called number versus the router will use port dial-peer to send the call to PBX. Bottom line, if router never looks at the number, then router should never look dial-peer.We know the dial-peer number is used to match the called number in the H225 signaling to route the call.

Absolutely. The router looks at the number in H225 (but could be SIP as well) and this is the number that you have configured as trunk destination, remember ?

The routes never look at the numbers in-band because the connection is established already soon the originating side seizes the trunk

or

The digits dialed by the phone user are never looked at by the router. Only numbers configured in "trunk" and "destination-pattern" are used by the router and these number are extraneous to the PBX dialplan

or

The routers will emulate a clear-channel T1 transparently to the PBXs. Optionally however, the voice can be compressed using a codec of your choice

I think it makes seven times I said the same thing now :)

This configuration works nicely for tens of thousands of cisco customers worldwide. We can go over and over and repeat and repeat - it won't change :)

Sorry I totally missed when you emphasized the trunk number/dial-peer pots versus user called number. I mixed them together. Now I guess I am really clear. Thanks you very much.

To summarize, now uses this link as a reference. Use the digital-to digital case. Not the digital to analog one. http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a00800afd65.shtml

Assumption(Based on your input and doc):

1. User need to dial the number

2. Connection Trunk mode is a permanent connection; the VoIP call is always connected independently of the plain old telephone service (POTS) port being on-hook or off-hook(from the link). Do we agree on this?

3. Three phones on r01, ext are 8001,8002,8003.

4. Three phones on r07, ext are 9001,9002,9003

5. We use the configuration in the link

6. trunk number is nothing to do with the PBX ext. Just for router to establish the permanent trunk when initially configure the router.

7. The permanent connection is RTP stream in our voip term.

Here is call flow I summarize:

1. There is connection already there between 1/0:1 on r01 to 1/0:1 on r07, 1/0:2 to 1/0:2 as soon as you have all the trunk/dialpeer, etc properly configured

2. 8001 user picks up the phone

3. He gets dial tone from the PBX

4. He dials 9001(most of the examples I saw the trunk number seem also are the extension number, that confused me. But now I know they are different since you said many times). If PBX configures PLAR, then PBX will seize the trunk without user to enter the number.

5. PBX seize the trunk 1/0:1 based on the PBX route plan where 9001 uses PBX's 1/0:1,

6. Now the 9001 as a DTMF tone sent inside the permanent connection from r01 to r07. r01 never need looks at the user called number(9001 this case)

7. r07 never need looks at this number either

8. Since r07 has permanent connection already with R01. It will use its 1/0:1 to send to its PBX. 9001 is passed to PBX at this moment.This is also called ds0-ds0 map.

9. Now 8001 talks to 9001.

10. Similarly 8001 can talk to 9002. If PBX programmed 9002 to use 1/0:2. That refers to what you said there is no limitation on the extensions.

11. Can I say there is no H225 set up at all in this call process? Because trunk is already there. It?s a permanent connection already. And also as you said router doesn?t look at the called number so I think there is no need to setup H225 for each call at all. Only time it requires H225 setup is when you have trunk created, router will use the trunk number to establish the connection.

Please comment on what I described. Hopefully that is really how the connection suppose to work. Thanks again.

Yes, things works as you described.

I'm happy that you understand now.

Now I have a few more questions for project particularyly.

1.If I have 4 remote branches where they all have PBXs and voice gateway connecting them using same manner. Central has only 1 T1, but PBX divided them into 4 trunks, each trunk has about 4-6 channels for each destination. Say in the PBX dial plan is configured 1/0:1-6 is for BR1, port 1/0:7-12 for BR2, port 1/0:13-18, port 1/0:19-24. They want incoming same way. Question is: with this connection solution, voice calls from remote site should be able use corresponded channels/ports to reach the central PBX? Because customer want to make sure the calls come in from remote site use allocated channels. I hope this solution can do this. That's what I am looking for. Thanks

Hi again.

Yes, with "connection trunk" incoming and outgoing calls will use exclusively the circuits as allocated.

On the other hand, even with "regular configuration" where the router interpret the dialed digits, there are techniques to send calls on specific ports / channels exclusively and/or limit the maxim number of calls between sites.

The second solution is usually better because give better allocation to the trunk, that is all resources are dynamically allocated as necessary, minimizes glare, and avoid taking up channels on the HQ PBX when talking branch-to-branch, but sometime it is hard to make customers (or PBX people) to understand that.

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