connection tie-line question

Answered Question
Apr 2nd, 2007

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.

Correct Answer by Paolo Bevilacqua about 9 years 10 months ago

The router will make no confusion as long the numbers used in dial-peer are correctly configured. Eg, if you have a mix of "router private" and "PBX dialplan" numbers, these do not overlap.


If you like the idea of using "connection trunk" tecnique with PRI too, it is possible to do that as well. It is called t-ccs (transparent CCS). Signaling timeslot is transported transparently with codec tccs, and voice channels can be compressed and VAD applied as desired.

Personally, unless proprietary ISDN IEs are used and the router is unable to pass them, I see no advantage in doing that.



The links about sending calls coming from VoIP to specific POTS, based on calling number:


http://cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801c0a88.shtml

http://cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml



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Paolo Bevilacqua Mon, 04/02/2007 - 11:46

Hi,


either with connection plar or tie line, you define a number as destination. Now this number, is mapped to a voice port on the receiving gateway. Having two t1 you can configure from a minimum of two to a maximum of 48 voice ports (that is if you configure idividual ds0-group for each channel).

So yes you can define which t1 talks to which remote site. DID doesn't matter because the router will not even look at the ditis and pass them trasparently.


Hope this helps, if so please rate post!

ciscoforum Mon, 04/02/2007 - 12:18

Thanks for your response.


I got a few more questions:


1. Connection tie-line or connection trunk only can be followed by a single phone number not a masked number like 11..?

2. If point 1 is right, this means it only can support 48 numbers max?

3. When user pick up phone do they have enter the called number if not using plar?

4. If they have to enter called number, then router should look at the called number then use dial-peer to route the call.right?

5.Last question, how does the remote site know that it should use which ds0-group if not based on the dialed number? Which commands in the configuration will signal that? here is the reference link I am reading now http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a00800afd65.shtml


Sorry for so many questions.


Thanks very much.



Paolo Bevilacqua Mon, 04/02/2007 - 12:41

1 - yes

2 - no, 48 channels for two T1, not 48 numbers

3 - user always has to enter called number, unless the PB is set for hot-line.

4. - with connection plar, trunk or tie line, router does look at the number. PBX does.

5 - the number you configure is used in the DP to match port/channel. This number has nothing to do withe the numbers used by the PBX and can bee anything.


Now the question, at the remote sites do you have PBXs as well? If so, and if you want to keep them, connection trunk is right for you. If not, the router is able to control local ethernet phones and trunk them to the mainsite PBX. This is called CCME.


Hope this helps, please rate posts using scrollbox below!

ciscoforum Mon, 04/02/2007 - 13:07

Thanks for the prompt response.


2. since there is only one number on each connection-trunk and there are only 48 channels or ports, meaning I only can have 48 trunk connections with 48 numbers, correct?

4. I guess you meant router does not look at the number.right? If the router does not look at the dial-peer, how does the router know that it should send the call to remote A or remote B? So it still needs look at the dialed number. I think.


5. Is each phone paired with another phone at remote site, meaning the user only can dial one remote user? Here is what copied from the link in the previous email:

---DS0 to DS0 mapping is retained on digital trunks---


6.Generic question: Here comes a very basic question, why we need this kind of trunk connection solution at all, can't I just use normal way to establish H323 calls without connection command involved if looking at the following scenario? PBX1-E/M-R1--IP-H323---R2-E/M-PBX2. I guess I will be more clear if I understand this question.


Thanks again

Paolo Bevilacqua Mon, 04/02/2007 - 13:37

2 - yes, 48 numbers to configure. Again, these are used only for configuration and not know to users.


4 - router does not look at the number dialed by the user because it has established a trunk between pbxs already, so user numbers do not need to be interpreted.


5 - remember, not a phone to phone channle, but a PBX trunk to PBX trunl. Obviously a single PBX channel/trunk can serve many phones.


6 - It's a matter of design preference. With connection trunk what was working before works unchanged after you intorduce the router, no changes to dialplan and you just save the dedicated circuit. But you are right, things would work just the same if router was to interpret the digits and use dial-peers. The side advantage would be that call between remotes would not take any port on the mainsite (tandem-switching).


Would you please consider rating useful posts using the scrollbox below?

ciscoforum Mon, 04/02/2007 - 13:58

Hi


Thanks very much. I am getting clearer and clearer.

For Q2.One more point maybe you are interested is that. Actually user do not dial the number. Here is what I find from the link http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a00800942a5.shtml.


"Both Connection Trunk and Connection PLAR modes have statically configured endpoints and do not require the user dialing to connect calls."


So the user actually can not or do not dial the number , when they pick up, PBX and router will send the call to remote ds0 right away. That's why router does not need to look at the digits. Does this make sense?


6. If Q2 is right from the Cisco link, then the trunk is used to specifically when to meet the traditional PBX requirement: phone user do not need to dial extentions. Right?



Thanks.


Paolo Bevilacqua Mon, 04/02/2007 - 14:21

Q2: When the user goes off hook, it is up to the PBX to decide what to do. Usually dialtone is presented, but the PBX itself could be configured for a sort of connection PLAR. Now, if the user going off hook causes the PBX to engage one channel on the T1, then the router will transport signaling and voice (optionally compressed) to the other end.


The quote from the document is misleading. If user does not dial anything, how can you expect that it will be connected to a choice of different parties ?


Q6. again, remember that phones are isolated from the router because there is a PBX in between.


One way to understand better the difference between trunk and "regular" configuration, is to look at the job the routers are doing.


With trunk/tie-line/plar, routers are emulating a T1 circuit. No intelligence, but simply point-to-point bit transport.


With "regular" configuration, routers are emulating the PSTN, calls are effectively "routed" and the router must be aware of the dialplan details.


Each method has his own advantages, but usually the second one has more. For example, if you know that on the mainside you do not need as many as two T1 to talk to two remote sites, you can configure one only and the router will share dynamically for both remote sites.

Or, you can add more remote sites with different sigalling and things will still work.


Thanks for the nice rating and please continue rating posts as appropriate!


ciscoforum Mon, 04/02/2007 - 14:45

Confused again.

2. Acording to the document, the number followed by the connection trunk/tie-line command is the target phone you want to reach, right? for example,voice-port 1/1,

connection tie-line 5550100. Here the 5550100 is the remote phone number. As soon as you configure this number in the connection command, the router establish the RTP stream already between the two routers.

So when you pick the phone, PBX seize that channel(if PBX configured that way) and router will use the already established permanent link to send the ring to the other end. You don't have option to call the other parties. that's what so called ds0-to-dso map. So, you think user still have the ability to dial the other numbers and ds0 is not statically map to the other ds0? I like this idea, but how can I convince customer if I don't have Cisco document supporting me?


7. What's the difference between connection tie-line and trunk?

Paolo Bevilacqua Mon, 04/02/2007 - 15:27

From the documentation:

Use the connection tie-line command when the dial plan requires that additional digits be added in front of any digits dialed by the PBX and that the combined set of digits be used to route the call onto the network. The operation is similar to the connection plar command operation, but in this case the tie-line port waits to collect digits from the PBX. The tie-line digits are automatically stripped by a terminating port.


Everything boils down to what you want/need to do.

If you do trunk, and configure unique voice port per DS0, be assured you will have the perfect ds0-to-ds0 static map.

If you want the router to interpret digits and route, you can do that.

If you want the PBX have all the intelligence and the router none, you can do that.

If you want a mix of the above, like apparently 'tie-line" does, you can do that.


Discussing the right technique with the customer can lead out of scope at times. Some customers thinks they are so good technically but if that was true, they wouldn't be, ehm, customers. Better is to understand very well the requirements, and if these cannot be meet immediately, communicate an alternative.

Then, once you have the routers in place and the network is switched over, most likely they will change their minds anyway :)



ciscoforum Mon, 04/02/2007 - 17:08

Here is what the customer requirement:

They currently have IGX in the WAN. Central PBX has two T1/CASs connecting to the IGX. Two remote sites have 1 T1 each. Since IGX has PVC map to T1/CAS, so the two remote sites calls to the HQ use HQ's T1 correspondly, meaning RS1 uses HQ 1st T1, RS2 uses HQ2 T1. Now we are going to phase out their IGX and put in IP based high end router. Now since the HQ only has 1 phone number ranges(each T1 can reach all the HQ phones), we are not able to dedicate HQ's 1st T1 for RS1, and 2nd T1 for Rs2 as before.

So I am thinking to use "connection". But it seems this won't work either for the following reason:


1. Does Trunk/Tie line allow user to dial the number? I am still not clear about this.I wish it can as you said.

2. HQ has about 200 phones, we only can specify 24 ports for each T1, meaing only 24 numbers can be dialed.


What do you think?

Paolo Bevilacqua Tue, 04/03/2007 - 01:02

Hi,


First of all, be assured that if the IGX was working, the router will work as well, in any configuration. It is perfectly capable to do everything the IGX was doing, and more.


2nd, I think that you confuse extension numbers with channels. There is no direct mapping between these.


Yous say that "since HQ onlys has 1 extension range" you can't dedicate 1 T1 each for site. Beside that contradicts what you have been saying so far, again extension ranges has nothing to do with the trunks.


Now for the questions:

1. Surely trunk lets the user dial: user goes off hook. PBX give dialtone. User dials. PBX collects digits and seize trunk. Still, no digits collected by the router.


2. Not really. After seizing the circuit, digits are passed from caller to called. So you can call everyone with any number of channels.




ciscoforum Tue, 04/03/2007 - 06:37

Hi


In a non "connection trunk/tieline" world, I totally understand DID is nothing to do with Channels. I can have more than several hundreds exts and just with 1 T1. But in this "connection", to me it is one channel to one number or ext. Here is the from the link http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a00800afd65.shtml


R1

controller T1 1/0

framing esf

linecode b8zs

ds0-group 1 timeslots 1 type e & m-wink-start

ds0-group 2 timeslots 2 type e & m-wink-start

clock source line


voice-port 1/0:1

connection trunk 2000

voice-port 1/0:2

connection trunk 2001


R2:

dial-peer voice 2 pots

destination-pattern 2000

port 1/0:1

dial-peer voice 3 pots

destination-pattern 2001

port 1/0:2


based on this config, when port 1/0:1 on R1 is seized, it only can reach 2000 on R2, same on port 1/0:2, when this port or channel is seized it only can ring 2001 on R2. Is this right? and since 1 T1 only have 24 channels, then you only can create 24 ports or channels for 24 extension, am I correct. Otherwise what's the point to use the real extension on the connection trunk command? Thanks

Paolo Bevilacqua Tue, 04/03/2007 - 06:51

Hi,


As I was saying before two or three times, extension numbers used in "connection" commands are NOT PBX extension numbers.


Instead, are numbers needed only to map ports from a router to another. Often you will use mnemonics for this, e.g. port 1/0:1 on router 2 will be 2101.


With connection trunk the router is totally unware of the number handled by the PBX, and viceversa.


ciscoforum Tue, 04/03/2007 - 07:52

So this really is not the phone extentions. But these numbers have to match the destination numbers on the dial-peer voice voip, correct?

Paolo Bevilacqua Tue, 04/03/2007 - 08:10

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.

ciscoforum Tue, 04/03/2007 - 08:50

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.

Paolo Bevilacqua Tue, 04/03/2007 - 09:40

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.

ciscoforum Tue, 04/03/2007 - 09:55

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.

Paolo Bevilacqua Tue, 04/03/2007 - 10:11

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.


ciscoforum Tue, 04/03/2007 - 10:24

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?



Paolo Bevilacqua Tue, 04/03/2007 - 10:50

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.

ciscoforum Tue, 04/03/2007 - 11:29

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.


Paolo Bevilacqua Tue, 04/03/2007 - 12:37


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.


ciscoforum Tue, 04/03/2007 - 16:20

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.

Paolo Bevilacqua Tue, 04/03/2007 - 16:29

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 :)



ciscoforum Tue, 04/03/2007 - 18:36

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.


ciscoforum Wed, 04/04/2007 - 05:28

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

Paolo Bevilacqua Wed, 04/04/2007 - 06:23

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.

ciscoforum Wed, 04/04/2007 - 06:45

Thanks very much. I am one more step further now.

Now on the above scenario, I have to add 1 PRI on the HQ and add one more Branch, say BR5 which it only has PRI. Understand connection trunk not work on PRI, that's fine. I don't need that.

The requirement is: Make sure call from BR5 has to use this PRI only.

My question is: when I have connection trunk configured to exclusively for the other 4 T1 CAS sites, I still should be able to use regular dial-peer to route the call from BR5 to use PRI to PBX based on the called number. To me, these are two different process, the router should be able to identify without confusion ,right?


2. On the previous thread you mentioned there is another way to send call on specific ports. I will be more than interested in hearing that. All I know that router has to route the call based on the called number in regular way.

Correct Answer
Paolo Bevilacqua Wed, 04/04/2007 - 07:28

The router will make no confusion as long the numbers used in dial-peer are correctly configured. Eg, if you have a mix of "router private" and "PBX dialplan" numbers, these do not overlap.


If you like the idea of using "connection trunk" tecnique with PRI too, it is possible to do that as well. It is called t-ccs (transparent CCS). Signaling timeslot is transported transparently with codec tccs, and voice channels can be compressed and VAD applied as desired.

Personally, unless proprietary ISDN IEs are used and the router is unable to pass them, I see no advantage in doing that.



The links about sending calls coming from VoIP to specific POTS, based on calling number:


http://cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801c0a88.shtml

http://cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml



ciscoforum Wed, 04/04/2007 - 08:41

This link is very good. But we may not able to do the calling number translation on the tieline(E&M), if it does not support calling number on the signaling. I know only problably group D supports it.

Paolo Bevilacqua Wed, 04/04/2007 - 10:00

For "connection trunk" you never need to worry about matching on calling number, because the destination number is unique, and will match one destination port only.


However, if you wanted to insert and arbitrary calling number on ports that do not have it, you can use "station number" command for FXS, or a "translate called" for any type of port.


Calling number (ANI) is supported in E&M Feature Group B. Another variant of CAS that supports ANI is Feature Group D, that is used on some PSTN trunks.

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