08-17-2009 02:18 PM - edited 03-15-2019 07:24 PM
I configured 3845 as CUBE to connect to Verizon SIP trunk and outbound calls connect fine however, I'm having problem with inbound calls. It just busy out and it appears that calls are getting to CUBE gateway but not forwarding calls to UCM 6.x cluster.
VZ is sending 10 digits and I configured a voice translation rule to 4 digit.
below is a copy of partial config.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol cisco
sip
early-offer forced
midcall-signaling passthru
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
!
!
!
!
--More-- !
!
!
!
!
!
!
!
!
voice translation-rule 1
rule 1 /xxxxxxx0730/ /0730/
!
voice translation-rule 2
rule 1 /^9\(.*\)/ /\1/
!
!
voice translation-profile Remove9
translate called 2
!
voice translation-profile SIP-Incoming
translate called 1
dial-peer voice 1 voip
description ***Incoming Calls from SIP Trunk***
translation-profile incoming SIP-Incoming
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:xxx.xxxx.xxxx.com
incoming called-number .
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
description ***Outbond calls to SIP-Trunk***
translation-profile outgoing Remove9
destination-pattern 9T
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target dns:xxxx.xxxx.xxxx.com
dtmf-relay rtp-nte
no vad
!
dial-peer voice 5000 voip
description ***To/From CUCM Sub for Voice***
destination-pattern 07..
voice-class codec 1
session protocol sipv2
session target ipv4:10.xxx.xxxx.131
dtmf-relay rtp-nte
no vad
!
dial-peer voice 5001 voip
description ***To/From CUCM Pub for Voice***
destination-pattern 07..
voice-class codec 1
session protocol sipv2
session target ipv4:10.xxx.xxx.130
dtmf-relay rtp-nte
no vad
!
!
sip-ua
retry invite 2
retry bye 2
retry cancel 2
sip-server dns:xxxxx.xxxxx.xxxx.com
g729-annexb override
!
08-17-2009 08:23 PM
I can try to guess based on the configuration, but 'debug ccsip message' is what you need to get a better idea.
Try:
-making sure that the outgoing interface you're using shown in 'show ip route 10.xx.xx.130' is the same IP address you've defined in CUCM for the SIP trunk.
-the CSS for the SIP trunk contains the partition for the 4 digit number you've created a DN for.
-the number you've defined in rule 1 is the exact number you're being presented by the SIP provider with.
-The CUCM group defined for this SIP trunk includes the .130 and .131 servers.
Hope that helps.
-nick
08-17-2009 08:59 PM
So, do I need to create another sip trunk in UCM to UCM servers?
I only have one sip trunk configured from UCM to SIP GW.
08-18-2009 05:53 AM
Whichever address that the router is using is the SIP trunk you need to configure. If there are 4 different IP addresses of different devices that are going to contact your CUCM you need 4 SIP trunks.
-nick
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