01-12-2012 06:27 AM - edited 03-16-2019 08:58 AM
Hello to all ,
I have a CUCM 8.6.2.20000-2 and I need to configure the external calls . In the past we has configured the calls and gateways in MGCP , but today we need to configure in H.323 :
GATEWAY CONFIGURATION ( MGCP Configured )
!
voice service voip
h323
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
!
!
!
voice class h323 1
h225 timeout tcp establish 2
!
!
!
!
voice class custom-cptone 1
dualtone disconnect
frequency 450
cadence 300 200
!
!
!
!
ccm-manager fallback-mgcp
ccm-manager mgcp
!
!
mgcp
mgcp call-agent X.X.X.X 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
!
mgcp profile default
!
Thanks for the help !
Hugo
01-12-2012 07:02 AM
There are plenty of docs on CCO on CUCM with H323 configuration, have you searched for any of those and tried to follow the instructions???
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk
01-12-2012 07:26 AM
Hello javalenc ,
Thanks for your time , I am looking for an expert user who can give me some advice in this problem .
I still trying to resolve this
01-12-2012 07:28 AM
And what problem is exactly you're facing??? your post only mentions you want to configure calls using H323 and for that the links on CCO will suffice
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk
01-12-2012 07:56 AM
Hello Hugo,
Take a look on this links:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b06gtway.html#wp1311511
Take a look on CUCM Guides and Documentations like Jaime mentioned...
Thanks
01-12-2012 08:28 AM
Thanks Leonardo and Jaime :
I am reading the documentation now . My problem is with the outbound calls , I guess is about dial-peer voice configuration in h323 .
Thanks in advance
Hugo
01-12-2012 08:34 AM
You need the dial-peers pointing to the PSTN, and you need to send the digits on the format accepted by the Telco.
If you are using an E1/T1 you use a pots dial-peer or if you re using a SIP Trunk you use a voip dial-peer pointing to the SIP Proxy.
Take a look on this doc:
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_confg.html
Rate this if helps you
Regards
Leonardo
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