Inbound call problem with SIP trunk

Unanswered Question
Aug 17th, 2009

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

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voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

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--More-- !

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voice translation-rule 1

rule 1 /xxxxxxx0730/ /0730/

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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

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Nicholas Matthews Mon, 08/17/2009 - 20:23

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

mpower78 Mon, 08/17/2009 - 20:59

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.

Nicholas Matthews Tue, 08/18/2009 - 05:53

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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