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No Name Display when using IOS Voicegateway configured as SIP

john.jarboe
Level 1
Level 1

Hello,

I am having trouble with name display on inbound pri calls from my IOS voice gateway when configured as SIP to the CUCM.  If I configure as MGCP it works fine.  I can debug Q931 and see both the name and number comming in on the PRI.

Any special settings that need to be configured on the CUCM or IOS voicegateway when using SIP between the two?

Here are some snippits from my config:

Voice Gateway image:  c3900e-universalk9-mz.SPA.151-4.M4.bin

network-clock-participate wic 0

network-clock-select 1 T1 0/0/0

network-clock-select 2 T1 0/0/1

isdn switch-type primary-ni

trunk group AT&T

hunt-scheme sequential

voice service voip

allow-connections sip to sip

signaling forward unconditional

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

modem passthrough nse codec g711ulaw

sip     

  registrar server expires max 600 min 60

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

controller T1 0/0/0

cablelength short 110

pri-group timeslots 1-24

controller T1 0/0/1

cablelength short 110

pri-group timeslots 1-24

interface Serial0/0/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-ni

isdn incoming-voice voice

isdn supp-service name calling

trunk-group AT&T

no cdp enable

interface Serial0/0/1:23

no ip address

encapsulation hdlc

isdn switch-type primary-ni

isdn incoming-voice voice

isdn supp-service name calling

trunk-group AT&T

no cdp enable

voice-port 0/0/0:23

bearer-cap Speech

voice-port 0/0/1:23

bearer-cap Speech

dial-peer voice 100 pots

trunkgroup AT&T

description ***incoming PRI calls***

incoming called-number .

direct-inward-dial

dial-peer voice 101 pots

trunkgroup AT&T

description ***outgoing PRI calls***

destination-pattern 9T

dial-peer voice 502 voip

description *** SIP calls to CUCM ***

destination-pattern 502.......

session protocol sipv2

session target ipv4:192.168.1.15

voice-class codec 1 

dtmf-relay rtp-nte

no vad

dial-peer voice 1000 voip

description *** incomming SIP Calls ***

session protocol sipv2

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

no vad

2 Replies 2