SIP trunk to CUCM gateway

[Natalie Ramirez]

Do SIP trunks have to terminate to the CUCM?  Or can the terminate to a gateway on the CUCM?  Everything I have read indicates it terminates to the CUCM directly, and I have a couple working like that, but I would like to have it terminate to a router remotely if possible.

[Nadeem Ahmed]

Hello Clark,

Yes you can add a SIP trunk to Gateway couple of things needs to be done more on GW side like dial-peer, Binding, allowing connection

what's exactly your topology and purpose that you are going to used

Br,
Nadeem 

Please rate all useful post.

[Chris Deren]

Typically Ciso recommends SIP trunk (assuming PSTN trunk) connection to CUBE (voice GW router with CUBE licenses) rather than direct connection to CUCM for many reason, such as demarcation, security, authentication, header modification, etc.

HTH,

Chris

[Natalie Ramirez]

On the gateway router, I have a PRI connected at the present time.  I have another location with a CME - SIP PSTN connection.  When I went to setup 4 digit dialing between CUCM and CME, I configured the Trunk in the CUCM and had to connect the CME to the CUCM directly.  When I attempted to configure the CME to attach to the CUCM gateway router, I get fast busy "unknown number" on phone and when I debug "debug voice ccapi inout" it shows disconnect cause 1.

The site that is currently a CME, will eventually become a gateway on the CUCM.  Since the only way I could get the SIP trunk to work was to tie it directly to the CUCM, I thought it was just not possible.  So, how do I configure the router to recieve a SIP trunk for the CUCM?  I have read 4 different forum posts here that document how to create SIP trunk directly to the CUCM, but none that specify how to terminate it to the gateway router.

[Kevin Monteiro]

Hi Clark,

Ip-phone>>cucm>>sip-trunk>>cme>>ipphone.

This is your call-flow, do correct me if  i am wrong.

Now this is what you need to do:

On the CME:

voice service voip

allow h323 to h323

allow h323 to sip

allow sip to sip

allow sip to h323

Dial-peer voice 100 voip

sess protocol sipv2

destination-patter 4...$

session target ipv4:192.168.4.4>>>>Ip-address of the CUCM

codec g711ulaw

dtmf-relay sip-notify

no vad

This dial-peer would be for the incoming call from the CUCM:

Dial-peer voice 1 voip

sess protocol sipv2

incoming called-number .

dtmf-relay sip-notify

On the CUCM SIP-trunk:

Make sure you have ip-address of the CME added in the destination address:

SIP Information

Destination

Destination Address is an SRV

			Destination Address

Destination Address	Destination Address IPv6	Destination Port
	1

Do rate the post accordingly.

Regards,

Kevin

[Natalie Ramirez]

That is what I already have, and that works fine.  What I want to know is how do it create a call-flow as follows:

ip phone --> cme --> sip-trunk --> Gateway router --> cucm --> ip phone

This is because eventually, the cme will become a gateway router for the cucm, and the cme currently has sip trunk from the PSTN terminating to it at the remote site.  With cme, it is very easy to recieve a SIP trunk from the PSTN.  How do I get the PSTN sip trunk to route calls in to the gateway router (like a PRI), without directly terminating the PSTN sip trunk to the cucm?

[Kevin Monteiro]

Then this should work, Can you send me the output of the following debug:

debug voip ccapi inout

debug ccsip messages

do let me  know the calling and the called numbers withe range of number on the CUCM where you would want to route the calls to.

Regards,

Kevin

[Nishant Savalia]

Hi clark,

In your case you have to create two dial-peers on your CME (which is your gateway router).

• 1st dial-peer to accept incoming call from PSTN SIP
• 2nd dial-peer is pointing towards CUCM SIP Trunk

Regards,
Nishant Savalia