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E1 PRI as a SIP voice gateway

Hi All,

Can we configure E1 PRI as a SIP gateway ??

Thanks

Deeps

7 Replies 7

j.huizinga
Level 6
Level 6

You mean as a CUBE?

That should be possible

JH

Hi JH,

I mean SIP gateway using  E1 PRI instead of  SIP trunk..

Deeps

So you have a voice gateway configured with a SIP trunk? And you want to have this device to connect to an E1 PRI?

Also possible, you need DSP's and an E1 interface card

JH

Hi JH

Thanks for feedback

Yes I have E1 PRI connected on VWIC 2MFT E1 card on  Cisco 3900 series router,and I want to configure this router as a SIP gateway

Deeps

Deepak,

If what you mean is the following:

CUCM ------SIP------3900---E1/PRI----PSTN/

Yes that is definitely possible. You will need to setup your PRI and incoming/outgoing POTS dial-peers for that. Once that is complete, you will also need to setup your SIP trunk back to CUCM via your VOIP dial-peers. This is generally referred to as a SIP gateway with TDM PSTN transport.

Thanks,

FG

Hi Francisco,

Do you have a sample configuration stated above? Also, If we are using hardware conferencing does the SCCP configuration needed like the following if it is configured for SIP?

sccp local Loopback0
sccp ccm x.x.x.x identifier 1 version 4.1
sccp ccm x.x.x.x identifier 2 version 4.1
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 1 register CFBxxxxxxxxx
 switchback method graceful
!
dspfarm profile 1 conference  
 codec g729br8
 codec g729r8
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 maximum sessions 8
 associate application SCCP

You are correct on the DSP resource configuration. Keep in mind that CFBXXX will only be used if it is assigned to a MGRL/MRG of a phone or device pool.

As far as a config, I can upload one a bit later. In the meantime I will post some of the key elements to make it work.

Necessary steps:

- Setup your PRI. Ensure it is in "multiple frame established" when doing "sh isdn status"

- Configure your incoming and outgoing pots dial-peers.

- Configure global SIP parameters under "voice service voip". At a minimum: a trusted ip address list of your CUCM to avoid issues in the future. And bind your signaling/media to an interface (whichever IP you'll point the trunk in CUCM to).

- Configure your voip dial-peers, don't forget to specify "sipv2" because by default voip dial-peers are h323. Point the session target ipv4 to your CUCMs, include multiple dial-peers pointing to different subscribers for redundancy, and increase the preference accordingly so they do not round-robin. Feel free to ask for more help on this one, it's broad and I'm not sure how much mileage you have.

- Configure your SIP trunk in CUCM, pointing to the IP address you binded your media/signaling to in the router(cube).

- Add your trunk to a RG/RL or simply point your CUCM patterns to the trunk. Your choice.

It probably would have been faster to just type a config for you but I want you to get the concept as well. I'll pull a config from a working environment so you can see what I am doing.

Thanks,

FG