Sip calls issue via H323

[Jean Lofrano]

Hi,

I have the follow environment:

ip phones (SCCP and SIP) ----------- CME-SRST(Site B) --------H323-----------CUCM (Site A)

                                                        |

                                                   PSTN

Every phones are registered on CUCM (site A) in SRST mode.

The SCCP phone are doing outbound call normally via PSTN local, but the SIP phones calls are dropping.

thanks. 

[Jean Lofrano]

Hi Gragory!!!

The SIP phones are making outbound calls now!!!

The dial-peer voip was the solution.

thanks.

[Gregory Garrian]

Good to hear!

Can you elaborate on the solution?

[Jean Lofrano]

I put the follow:

dial-peer voice 706 voip

corlist outgoing DF

description LIGAR-CURITIBA-1

destination-pattern 706.$

session target ipv4:172.20.1.106

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

At the moment of the call o codec is being negotiated right now.

thanks.

[Gregory Garrian]

You wouldn't need any dial peers for the directory numbers on the SIP phones.

It doesn't look like you have an MTP setup on the gateway (CME). 

On the gateway configuration for the CME router in CUCM do you have the "Require MTP" box checked off?

You may need to create the MTP on the gateway and register it to CUCM for it to work.