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Defintiy G3 integration with 3640 VoIP Implementation

gmatteson
Level 1
Level 1

I have been trying to implement a 3662 VoIp integration of two Definty G3 PBX systems, using the HDV-2T1 VWIC cards in each of the two 3662 routers in this environement. Problem is, these G3s share an Audix voice mail system that will require the use of the Lucent DCS PBX communications protocol. It is my understanding that the 36xxs series will accomplish this useing CCS frame forwarding, basically by setting up DS0 gorups under the controller, etc. I have hit that wall, and niether the PBX guy( understandably, they really hate when you put voice over data :) or cisco has been very helpful in getting this resolved ( I have opened a TAC Case) I am wondering if anyone has had any experience whatsover in integrating Definity G3 PBXs in a VoIP infrastructure. Any help would eb greatly appreciated, BTW I have been all over Ciscos web page...

Thanks in advance!

Gary Matteson

Replies to either this board or my email address...

4 Replies 4

dgoodwin
Cisco Employee
Cisco Employee

Are the G3 switches configured to use DCS or DCS+ to speak to each other? With DCS, 2 signaling channels are used and DCS+ uses only one.

Either one works correctly using the Transparent CCS feature (not CCS frame-forwarding which utilizes VoFR). Have you looked at this specific link:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dt_tccs.htm

Dave, Thanks for your reply. We are definitely on the same wavelength with this, and this confirms my belief that this is probably the appropriate configuration. The Definity G3s are configured to use DCS (not DCS+) signalling, and in fact I had configured two signalling channels for the DCS connection using a separate DS0 group for channel 23 and channel 24. I have attached a sanitized configuration that I think is appropriate for this implementation. As for your link above, this is the link I had used to accomplish this configuration. I cannot get basic error free T1 connectivity to the Definity PBXs and therefore cannot test what I believe to be the correct configuration. I am looking for specific experience on the G3 switch, to verify that the G3 is configured appropriately. Would you have any ideas of resources to go to for this information? thanks in advance for your help

voice-card 1

!

voice service voip

!

controller T1 1/0

framing esf

linecode b8zs

channel-group 0 timeslots 1-24

!

controller T1 1/1

framing esf

clock source line primary

linecode b8zs

ds0-group 0 timeslots 23 type ext-sig

ds0-group 1 timeslots 13-22 type e&m-wink-start

ds0-group 2 timeslots 24 type ext-sig

!

!

interface FastEthernet0/0

ip address x.x.x.x

!

voice-port 1/1:0

local-alerting

timeouts wait-release 3

connection trunk 98 answer-mode

!

voice-port 1/1:1

!

voice-port 1/1:2

local-alerting

timeouts wait-release 3

connection trunk 96 answer-mode

!

dial-peer voice 6000 voip

destination-pattern 6...

session target ipv4:y.y.y.y

!

dial-peer voice 5000 pots

destination-pattern 5...

!

dial-peer voice 4000 voip

destination-pattern 4...

session target ipv4:y.y.y.y

!

dial-peer voice 3000 voip

destination-pattern 3...

session target ipv4:y.y.y.y

!

dial-peer voice 98 voip

destination-pattern 98

session target ipv4:y.y.y.y

codec clear-channel

!

dial-peer voice 99 pots

destination-pattern 99

port 1/1:0

!

dial-peer voice 97 pots

destination-pattern 97

port 1/1:2

!

dial-peer voice 96 voip

destination-pattern 96

session target ipv4:y.y.y.y

codec clear-channel

Regards,

Gary matteson CCIE 5101

Midwave Corporation

Obviously you must first get an error free T1 connection to the G3 switch in order to get an accurate test.

Some of the things you have in your config are correct, some need to change.

First, you have to use type ext-sig for all the ds0-groups. Also, you must define a separate ds0-group for each channel to be used -- both bearers and signaling. And you must use connection trunk on both the bearer and signaling voice-ports, not just the signaling ones. You had the right idea to use a normal codec for the bearer voip dial-peers and clear channel codec fo rthe signaling dial-peers. Here is how I would set up the config for the router you did above (copy/pasted appropriate parts then modified, only including what is relevant to this config). Hopefully this is clear to you.

It looks like you are still trying to achieve call routing decisions on the router but with DCS on the G3 switches and T-CCS feature on the routers, all the destination-patterns are dummy and only significant to setup the trunk connections between all the timeslots on each side of the data network and to make sure that timeslot 13 on side A is trunked to timeslot 13 on side B, etc.

controller T1 1/1

framing esf

clock source line primary

linecode b8zs

ds0-group 0 timeslots 23 type ext-sig ! DCS

ds0-group 1 timeslots 24 type ext-sig ! DCS

ds0-group 2 timeslots 13 type ext-sig ! bearer

ds0-group 3 timeslots 14 type ext-sig ! bearer

.

.

ds0-group 10 timeslots 21 type ext-sig ! bearer

ds0-group 11 timeslots 22 type ext-sig ! bearer

!

!

interface FastEthernet0/0

ip address x.x.x.x

!

voice-port 1/1:0

connection trunk 301 answer-mode

description DCS 1: we are 401, remote is 301

!

voice-port 1/1:1

connection trunk 302 answer-mode

description DCS 2: we are 402, remote is 302

!

voice-port 1/1:2

connection trunk 101 answer-mode

description bearer timeslot 13: we are 201, remote is 101

!

voice-port 1/1:3

connection trunk 102 answer-mode

!

voice-port 1/1:4

connection trunk 103 answer-mode

!

voice-port 1/1:5

connection trunk 104 answer-mode

.

.

voice-port 1/1:11

connection trunk 110 answer-mode

!

!

dial-peer voice 300 voip

destination-pattern 3.. ! this is for DCS channels

session target ipv4:y.y.y.y

codec clear-channel

!

dial-peer voice 100 voip

destination-pattern 1.. ! this is for bearers

session target ipv4:y.y.y.y

!

dial-peer voice 401 pots

destination-pattern 401 ! our DCS 1

port 1/1:0

!

dial-peer voice 402 pots

destination-pattern 402 ! our DCS 2

port 1/1:1

!

dial-peer voice 201 pots

destination-pattern 201 ! our bearer timeslot 13

port 1/1:2

!

dial-peer voice 202 pots

destination-pattern 202 ! our bearer timeslot 14

port 1/1:3

.

.

dial-peer voice 210 pots

destination-pattern 210 ! our bearer timeslot 22

port 1/1:11

Dave,

Thanks for your reply on this. What you are saying makes sense. I hadn't been able to determine if I needed to set up a separate DS0 group for each bearer channel, but suspected that might be the case. I will test these changes out and report the findings back here.

Regards,

Gary Matteson