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.
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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:
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.
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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.
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.
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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.
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).
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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?
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.
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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?
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 :)
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?
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.
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
controller T1 1/0
ds0-group 1 timeslots 1 type e & m-wink-start
ds0-group 2 timeslots 2 type e & m-wink-start
clock source line
connection trunk 2000
connection trunk 2001
dial-peer voice 2 pots
dial-peer voice 3 pots
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
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.
So this really is not the phone extentions. But these numbers have to match the destination numbers on the dial-peer voice voip, correct?