01-30-2012 02:02 PM - edited 03-16-2019 09:18 AM
Is it possible to manipulate a specific B Channel to use a dialpeer? For example on a T1 with 24 channels, I need the last 4 B-channels to choose an specific dialpeer, same concepts as matching incoming DID's but on a B-Channel slot.
If a call arrives on B-channel 20, I need that call to be sent to a specific DN at the CallManager.
I am using H.323 on the gateway side.
thanks in advanced.
Oscar
01-31-2012 12:22 PM
The only way I can think to do this is if the T1 is using CAS and not ISDN. With CAS you would get a voice-port for each channel. You could apply a voice translation-profile to the last four voice-ports. The translation profile could prefix a unique string to the called number that you could use to match a separate incoming dial-peer.
01-31-2012 12:41 PM
I don't know if you could work with this, and I have never used this feature, but kept this post I saw a while ago - it looks like you can do trunk groups for PRI now:
You can't have multiple D channels, but the following feature might help you -
The customer had two different users and he wanted to split a single T1 PRI service between these users so that one user would use certain B channels and the other user would use different one for outwards calls on the CME router.
The phone company was able to assign these B channel groups with different number ranges for inwards calls.
There is a new feature in 12.4 code that allows you to define individual timeslots of the PRI and assign them to different trunk groups. These trunk groups can be used instead of a voice port in a POTS dial peer, so you can direct calls to very specific B channels on the PRI. Following is a config that demonstrates this feature on an E1 interface (30 B channels - same concept applies for a T1).
!
controller E1 1/0
pri-group timeslots 1-31
trunk-group Customer1 timeslots 1-10
trunk-group Customer2 timeslots 11-20
trunk-group Customer3 timeslots 21-30
!
!
dial-peer voice 1 pots
description - enable DID on group 1
incoming called-number .
direct-inward-dial
trunkgroup Customer1
!
dial-peer voice 2 pots
description - enable DID on group 2
incoming called-number .
direct-inward-dial
trunkgroup Customer2
!
dial-peer voice 3 pots
description - enable DID on group 3
incoming called-number .
direct-inward-dial
trunkgroup Customer3
!
!
dial-peer voice 5500 pots
trunkgroup Customer1
description - Customer 1 : access code 771
destination-pattern 771
!
dial-peer voice 5501 pots
trunkgroup Customer2
description - Customer 2 : access code 772
destination-pattern 772
!
dial-peer voice 5502 pots
trunkgroup Customer3
description - Customer 3 : access code 773
destination-pattern 773
!
In this case, Customer1 users would dial 771 as an access code, Customer2 would dial 772 and Customer3 would dial 773. These access codes would 'steer' the call to the appropriate timeslots on the PRI
https://supportforums.cisco.com/message/979725#979725
Mary Beth
02-01-2012 07:16 AM
Thanks for your reply, my only option is a T1 PRI, so CAS won't work here.
Mary, I have read that about the new feature on splitting channels on a PRI T1, this only works for outgoing calls, my need is for incoming calls at the very last 3 channels on a PRI T1.
I am going to check with the Telco and see if they can mask a number only for those 3 channels and route those DIDs to the desired destination.
any other suggestion is greately appreciated.
Oscar Perez
02-01-2012 07:29 AM
Oscar,
As Jonathan & Mary have indicated (+5 J + M )
You can control what is sent outbound, however on ISDN-PRI you have no control over how calls leave the telephone exchange (CO) like what B-channel they use. Here in the UK they will normally send out from B-Chan 1 ---> 30 but we can ask if we want round robin or sequential hunting of the B-Channels.
In older days we could use DASS but this is being phased out.Not suitable for CUCM without DASS to Q931 converters anyway.
You are only left with asking your service provider to perform some fancy tricks at their end.
Good luck
Regards
Alex
02-01-2012 09:55 AM
I think your only other hope would be if you could do something with translation rules, but as Alex has pointed out, you can't control what hits what channel coming in from the PSTN - this would translate everything coming in on that trunk group to 5555 - I think:
controller t1 1/0
pri-group timeslots 1-24
trunk-group Customer1 timeslots 1-19
trunk-group Customer2 timeslots 20-24
dial-peer voice 2 pots
description - enable DID on group 2
incoming called-number .
translation-profile incoming xlateDID
direct-inward-dial
trunkgroup Customer2
voice translation-rule 1
rule 1 / / /5555/
voice translation-profile xlateDID
translate called 1
02-01-2012 10:05 AM
As others have mentioned already, you cannot expect telco to send calls on specific channels. They will not even understand (or pretend) what you want.
In ISDN setup, there is always a called number. If you use the correct configuration tecniques, you can route you calls based on that, using translation-rules or TCL scripting.
02-07-2012 01:58 PM
Thanks everybody for your feedback, I am aware that an inbound call can not be controlled on which B-Channel is going to ended up. I think Mary's suggestion will provide the result I am looking for, which is that no matter the DID but if it is coming on TimeSlots 20 - 24 I should be able to manipulate that call to an specific number in callmanager/Unity. I knew about the trunkgroup capability under a T1/E1 controller but I have never tested for inbound porpuses, only for outboud. I will be testing the follow config at the customer site and will report back with the results.
======================================
dial-peer voice 2 pots
description - enable DID on group 2
incoming called-number .
translation-profile incoming xlateDID
direct-inward-dial
trunkgroup Customer2
voice translation-rule 1
rule 1 / / /5555/
voice translation-profile xlateDID
translate called 1
=====================================
Best regards,
Oscar Perez
11-11-2013 05:50 AM
Hi!
I'm stuck in a similar situation at the moment, and hope someone has solved this issue.
I have a "back to back connection" between two PBX'es that communicate using Q.931. I have had to replace my old hardware running a legacy CCS solution due to its incapability of understanding overlap signalling. I have replaced it with two Cisco routers running E1 (Q931). Between the units I have a high latency low bandwidth network.
The issue is that the PBX'es are configured in a way that requires calls between the PBX'es to use the same time slot at both ends. I'm running a VoIP network with CUCM between the two sites.
=============================================
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-21
trunk-group TS01 timeslots 1
trunk-group TS02 timeslots 2
trunk-group TS03 timeslots 3
trunk-group TS04 timeslots 4
trunk-group TS05 timeslots 5
trunk-group TS06 timeslots 6
trunk-group TS07 timeslots 7
trunk-group TS08 timeslots 8
trunk-group TS09 timeslots 9
trunk-group TS10 timeslots 10
trunk-group TS11 timeslots 11
trunk-group TS12 timeslots 12
trunk-group TS13 timeslots 13
trunk-group TS14 timeslots 14
trunk-group TS15 timeslots 15
trunk-group TS17 timeslots 17
trunk-group TS18 timeslots 18
trunk-group TS19 timeslots 19
trunk-group TS20 timeslots 20
trunk-group TS21 timeslots 21
!
dial-peer voice 81030001 pots
trunkgroup TS01
description ** PBX TS01 **
translation-profile incoming 61031401
translation-profile outgoing 25
destination-pattern 81030001T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
!
dial-peer voice 81030002 pots
trunkgroup TS02
description ** PBX TS02 **
translation-profile incoming 61031402
translation-profile outgoing 25
destination-pattern 81030002T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
!
(...)
voice translation-rule 25
rule 1 /^810300../ //
!
voice translation-rule 61031401
rule 1 // /61031401\1/
!
voice translation-rule 61031402
rule 1 // /61031402\1/
!
(...)
voice translation-profile 25
translate called 25
!
voice translation-profile 61031401
translate called 61031401
!
voice translation-profile 61031402
translate called 61031402
!
(...)
=====================================
The idea is that I do not know (or care) what numbers are used as SOURCE or DESTINATION of the original call. My network should be transparent to the PBX number plan. I need to add a prefix, and it should be based on the timeslot the call comes in on. I route the traffic between the routers using the prefix.
The configuration excerpt above should add 61031401 prefix to all calls entering on TS01, and 61031402 to all calls entering on TS02 etc. Calls from the remote should have corresponding prefixes 81030001 for TS01 and 81030002 for TS02 etc.
The outbound (from voip to pots) routing of the above configuration works.
However I have a challenge with the incoming prefixing.
All calls inbound end up using "dial-peer 81030001 pots".
I believe the reason this dial-peer "takes" all of the calls inbound from pots is due to the line "incoming called-number ."
Removing this makes no inbound pots call work as the "destination-pattern 8103001T" is never matched.
Removing "destination-pattern 8103001T" from the dial-peer is not working as it kills the voip to pots routing of inbound calls from the remote router.
Anyone got a good idea for me?
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