First letter incoming caller ID stripped on a SIP trunk

[Pierrot.damen]

Hi,

I'm having an issue with the first letter of the caller id being stripped when displayed on the phone.
I see the Caller ID coming in correctly from the ISDN on the voice gateway. the voice gateway is connected to the CUCM via SIP trunk.

Any suggestions on how to resolve this issue?

Thanks!

Pierrot

[ansarjavaid54]

Hi there.

Please check your dial-pears pointing to CM and voice translation-rules....

would u please share ur debugs and dial-pear configs to figure out the picture....

[Pierrot.damen]

Hi

here are the debug isdn q931 and debug ccsip messages output. i'll paste the dial-peers when i have the time. 

May 25 10:02:25.764 EDT+1: ISDN Se0/0/0:23 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x1014
May 25 10:02:25.768 EDT+1: %ISDN-6-CONNECT: Interface Serial0/0/0:3 is now connected to ****8280 ANTOINE Stanley
May 25 10:02:25.768 EDT+1: ISDN Se0/0/0:23 Q931: RX <- STATUS pd = 8 callref = 0x1014
Cause i = 0x80E34C - Information element not implemented
Call State i = 0x0A
May 25 10:02:52.820 EDT+1: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x07F3
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE980830E
Exclusive, Interface 0, Channel 14
Display i = 0xB1, 'STANLEY ANTOINE'
Calling Party Number i = 0x2183, '514*******'
Plan:ISDN, Type:National
Called Party Number i = 0xC1, '8280'
Plan:ISDN, Type:Subscriber(local)
May 25 10:02:52.824 EDT+1: ISDN Se0/0/0:23 Q931: Received SETUP callref = 0x87F3 callID = 0x397E switch = primary-ni interface = User
May 25 10:02:52.844 EDT+1: //1418797/34FF922BB635/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:****8280@10.102.89.163:5060 SIP/2.0

Via: SIP/2.0/UDP 10.103.133.249:5060;branch=z9hG4bKAF626CE5

Remote-Party-ID: "1STANLEY ANTOINE" <sip:514*******@10.103.133.249>;party=calling;screen=yes;privacy=off

From: "1STANLEY ANTOINE" <sip:514*******@10.103.133.249>;tag=B0F85660-1477

To: <sip:****8280@10.102.89.163>

Date: Wed, 25 May 2016 14:02:52 GMT

Call-ID: 35020363-21B811E6-9681BCB1-215C8F11@10.103.133.249

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 0889164331-0565711334-3056926745-0109376704

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1464184972

Contact: <sip:514*******@10.103.133.249:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 360

[Pierrot.damen]

Dial-peer and voice translation rules 

dial-peer voice 200 voip
description === to subscriber1 QBC ===
preference 1
destination-pattern 7226....
modem passthrough nse codec g711ulaw
session protocol sipv2
session target ipv4:10.102.89.163
voice-class codec 10
voice-class sip options-keepalive retry 3
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax rate 14400
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 201 voip
description === to subscriber2 QBC ===
huntstop
preference 2
destination-pattern 7226....
modem passthrough nse codec g711ulaw
session protocol sipv2
session target ipv4:10.102.88.172
voice-class codec 10
voice-class sip options-keepalive retry 3
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax rate 14400
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 100 voip
description *** DEFAULT VoIP Incoming Peer ***
modem passthrough nse codec g711ulaw
session protocol sipv2
incoming called-number T
voice-class codec 10
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax rate 14400
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 6 pots
translation-profile incoming AddLeading9
incoming called-number .
direct-inward-dial
port 0/0/0:23

voice translation-rule 1
rule 1 // // type any abbreviated plan any private
!
voice translation-rule 10
rule 1 /9136/ /72261300/
rule 2 /7\(...$\)/ /72263\1/
rule 3 // /7226/
!
voice translation-rule 11
rule 1 // /9011/ type international international
!
!
voice translation-profile 4to8digits
translate called 10
!
voice translation-profile AddLeading9
translate calling 11
!
voice translation-profile ChangeTypePlan
translate calling 1
translate called 1

[ansarjavaid54]

Hay there....

Well, when call came from isdn the caller id was (ANTOINE Stanley). But when it routed towards sip trunk it became (1STANLEY ANTOINE). You said that  striped on phone but i think there is addition of one word (1)  before the name.

Can u please tell me what is actual ID u want on your phone. Remote party sending you caller-id without 1.

secondly, you shared only SENT invite. Please share your receive invite from remote party to gateway.

[Pierrot.damen]

hi, 

Tried with another user and this gave the following result.

on my notepad it displayed as a 1 again and in translator X as a space, but for the cisco TAC engineer that is currently working on the case it appeared as this bizar symbol. (should have been an 'M' normally)

Via: SIP/2.0/UDP 10.104.240.149:5060;branch=z9hG4bK2503FA17C7

Remote-Party-ID: "ҍARIO CHAYER"

also in the screenshot it says xB1Mario and the isdn setup is says

Display i = 0xB1, 'MARIO CHAYER'

Could this have some relevancy to the case perhaps?

Regards,

Pierrot

[ansarjavaid54]

I think from config side everything seems okay. Do one thing its not a solution but a guess. Just remove all your sip trunk config and configure them again. 

Try it out and tell me if this solves or not and also keep in touch with tac team....

[Pierrot.damen]

Hi, sorry for the late reply but the issue was resolved.

The isdn switch-type was set to primary-ni and setting this to primary-dms100 resolved the issue.

Thanks for your time and assistence :)

regards,

Pierrot

[ansarjavaid54]

You welcome...