Connect two 2600 routers over T1 crossover using E&M trunk? Can't make a ca

[jaiello2000]

I have connected two 2600 routers with NM-HDV using T1 E&M signaling. I am trying to test SIP into one router, TDM voice to the other router and SIP onto the next ednpoint.

The problem is that I can see the digits on the "far end" router, but the call fails after the "near end" router sends the last digit. The "far end" is just waiting for more digits.

I hope I just missed something in the controller T1 configuration or some dial-peer configuration.

Any help would be appreciated.

Thanks,

joe

[Chester Rieman]

Try testing each leg of the call by using 'csim start xxxx' command on the router, where xxxx is the number you want to call. It will ring the phone but there will be no audio. good for testing step by step, to find out which leg of the call fails.

[jaiello2000]

I did this and it does the same thing as when calling from a nework phone.

I think the problem is that I need to identify one gateway to look like the PSTN and the other to look like cpe. It seems signaling on one end or ther other has to be different.

[yraps]

Are botrh controllers up? One in each rouer (2600)

'Sho controller T1"

[jaiello2000]

Yes, both are up. The call is initaited between the routers. The "far end" router received the digits, but that is it. I am not sure if the near end needs to indicate it is done dialing or if there is something special needed in the setup of the call.

It looks like both routers think they are the network and neither is acting like cpe.

One example is a PBX to 2600. The PBX thinks the 2600 is the PSTN. I need One router to act like a PBX .

I think the near ned router needs to be configured diffrently, just do not know what to do. I do not know if that can be done with Cisco gear.

[yraps]

Have you tried pressing '#' at the end?

It is built-in feature in Cisco that indicates the B# string termination.

I have connected back 2 back GW by using PRI. It is much easier to troubleshoot and it can support ANI and DNIS.

See the configuration below: (You can implement it on 2 different GW though it is taken from the same gateway.)

controller T1 0

framing esf

clock source line secondary 1

linecode b8zs

pri-group timeslots 1-24

description IVR Loop test termination (To Controler T1x3)

!

controller T1 3

framing esf

clock source line secondary 3

linecode b8zs

pri-group timeslots 1-24

description description IVR Loop test Origination (from Controler T1x0)

!

interface Serial0:23

no ip address

isdn switch-type primary-qsig

isdn protocol-emulate network

isdn incoming-voice modem

no isdn T309-enable

isdn T310 10000

isdn disconnect-cause 1

no cdp enable

!

interface Serial3:23

no ip address

isdn switch-type primary-qsig

isdn incoming-voice modem

isdn T310 10000

isdn disconnect-cause 1

no cdp enable

[jaiello2000]

Solved My problem....

Since I wanted two way calls, I set up both gateways similarly. On each, I created a single DS0-group with timeslots 1-24. This creates a single voice-port 1/0:0. On each gateway, I had a dial-peer for outbound calls. It appears that this flags the voice-port as a USER interface. I shutdown the dial-peer on one gateway and the call progressed as expected.

I should have created two ds0-groups on each gateway. One voice port would be bound to a dial-peer for outbound calls. The other used for inbound calls!

Live and learn!

Thanks for the suggestions.

Joe