No Name Display when using IOS Voicegateway configured as SIP

[john.jarboe]

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

[keglass]

John,

This community does not provide technical support and is not staffed with technical support experts. I recommend you post this and future technical support questions to the Cisco Support Community (https://supportforums.cisco.com/index.jspa) where our Cisco technical support experts provide debugging assistance. Another option is to open a ticket with the Cisco Technical Assistance Center (www.cisco.com/go/support) to get expert assistance.

We do encourage you to participate in the Cisco Collaboration Community and to also join our Cisco Collaboration User Group program!  In the community, we encourage your discussion/sharing around collaboration topics and Cisco Collaboration Solutions, including business and IT requirements, industry trends, process, culture/organization issues, how collaboration can be used to transform businesses, vendor selection, adoption, training, architecture, licensing, and product features/functionality. If you are a customer or partner, you can also join the user group program to be eligible for member-only events and influence product direction.

We hope to hear from you again.

Kelli Glass

Moderator for the Cisco Collaboration Community

[ryanticer]

Hi John,

Although this is not a technical forum, I will mention that on some PRIs they delay the calling name info until a later message. On the sip side, you may have to use a buffer invite timer to make it appear as though this calling name comes in immediately. For example:

Sip-ua

Timers buffer-invite 3000

(the 3000 above represents how long in ms you’re willing to wait for the calling-name)

Thanks!

Ryan Ticer Network/Convergence Engineer

916.577.1741 | rticer@team-sos.com<mailto:rticer@team-sos.com>