Caller ID not working on H323 Gateway

Hi,
I have a CCM Version 4.1.3 set-up to cater for 200 phones, I have a VG gateway in one of the sites that uses an FXO card for local calls, this VG is set-up as an H323 gateway, however calls coming throguh the PSTN line onto the FXO card and then onto an IP phone are not displaying Caller ID.
Just wondering if anyone has seen this before, I know that on an MGCP gateway the FXO cards don't support Caller-ID but we have it set-up as H323.
Thanks
Paul

Funny, this is a question straight from one of the CCVP tests, although I won't say which one.
Try "caller-id enable" in global mode and "station-id name" on the voice-port.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml

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yashshankar wrote:
Hi
I recently purschased an Online number but the caller id does not work for incoming calls.How do we resolve this problem?.
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Location/Ubicacion: Arizona USA
Time Zone/Hora Local: UTC/GMT -7
If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
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The external calls hit gateway and it matches the pots dialpeer where my fax connected to. From the "debug voip vtsp all", I see Primary Fax Protocol=CISCO_FAX_RELAY, which is not enabled and we are using T38 relay and modem passthrough method for faxing. I believe PVDM3 does not work with Cisco Relay and couldn't allocate any dsp resources for the call and it drops. But still I wonder how gateway takes cisco relay.
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voice service voip
allow-connections h323 to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
emptycapability
modem passthrough nse codec g711ulaw
dial-peer voice 1001 pots
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description PSTN Incoming, Pattern:.
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Logs :-
TL_INFO(TF_PROTOCOL) [1]0868.0F1C::07/12/2014-06:22:03.292.0000002f (S4,SipMessage.DataLoggingHelper:sipmessage.cs(686))[3168828821]
<<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_310339F>], 172.29.55.162:5068<-172.29.58.15:47485
INVITE sip:[email protected]:5068 SIP/2.0
FROM: "Deepak Srivastava" <sip:[email protected]>;tag=35075~3a834c6d-74d2-4f7c-89b1-cae86be13da3-48284953
TO: <sip:[email protected]>
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VIA: SIP/2.0/TCP 172.29.58.15:5066;branch=z9hG4bK10663aa09f0
ALLOW-EVENTS: presence
CONTACT: <sip:[email protected]:5066;transport=tcp>
CONTENT-LENGTH: 1187
DATE: Sat, 12 Jul 2014 06:22:03 GMT
EXPIRES: 180
SUPPORTED: timer,resource-priority,replaces
SUPPORTED: X-cisco-srtp-fallback
SUPPORTED: Geolocation
USER-AGENT: Cisco-CUCM8.6
CONTENT-TYPE: multipart/mixed;boundary=uniqueBoundary
ALLOW: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
P-ASSERTED-IDENTITY: "Deepak Srivastava" <sip:[email protected]>
MIME-VERSION: 1.0
Min-SE: 1800
Cisco-Guid: 3581525248-0000065536-0000000053-0255466924
Session-Expires: 1800
Geolocation: <cid:[email protected]>;inserted-by="172.29.58.15"
Remote-Party-ID: "Deepak Srivastava" <sip:[email protected]>;party=calling;screen=yes;privacy=off
--uniqueBoundary
Content-Type: application/sdp
v=0
o=CiscoSystemsCCM-SIP 35075 1 IN IP4 172.29.58.15
s=SIP Call
c=IN IP4 172.29.58.15
t=0 0
m=audio 28288 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--uniqueBoundary
Content-Type: application/pidf+xml
Content-ID: [email protected]
<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10"
xmlns:cl=" urn:ietf:params:xml:ns:pidf:geopriv10:civicLoc"
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xmlns:caps="urn:ietf:params:xml:ns:pidf:caps"
xmlns:cisco="http://www.cisco.com"
entity="pres:[email protected]">
<dm:device id="sg89ae">
<caps:devcaps>
<cisco:gateway>false</cisco:gateway>
</caps:devcaps>
<gp:geopriv>
<gp:location-info>
<cl:civicAddress>
<cl:country>NO</cl:country>
</cl:civicAddress>
</gp:location-info>
<gp:usage-rules>
<gp:retransmission-allowed>yes</gp:retransmission-allowed>
<gp:retention-expiry>2014-07-13T06:22:03Z</gp:retention-expiry>
</gp:usage-rules>
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<timestamp>2014-07-12T06:22:03Z</timestamp>
</dm:device>
</presence>
--uniqueBoundary--
------------EndOfIncoming SipMessage
TL_INFO(TF_PROTOCOL) [0]0868.08B8::07/12/2014-06:22:03.351.000000bf (S4,SipMessage.DataLoggingHelper:sipmessage.cs(613))[3168828821]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_310339F>], 172.29.55.162:5068->172.29.58.15:47485
SIP/2.0 488 Not Acceptable Here
FROM
: "Deepak Srivastava"<sip:[email protected]>;tag=35075~3a834c6d-74d2-4f7c-89b1-cae86be13da3-48284953
TO
: <sip:[email protected]>;epid=E254C31B54;tag=e8831d44ab
CSEQ
: 101 INVITE
CALL-ID
: [email protected]
VIA
: SIP/2.0/TCP 172.29.58.15:5066;branch=z9hG4bK10663aa09f0
CONTENT-LENGTH
: 0
SERVER
: RTCC/4.0.0.0 MediationServer
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]0868.0F1C::07/12/2014-06:22:03.532.000002a7 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(686))[3358199169]
<<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_2F779B4>], 172.29.55.162:59333<-172.29.55.24:5061
SIP/2.0 404 Not Found
FROM: <sip:031986;[email protected];user=phone>;epid=A6D7517671;tag=38bdbeec1c
TO: <sip:031986;[email protected];user=phone>;tag=D911B6138F470C253AEC6130E12D8647
CSEQ: 151 SERVICE
CALL-ID: e240b47e04744efaa88c778f9dcabe1d
VIA: SIP/2.0/TLS 172.29.55.162:59333;branch=z9hG4bK894ba852;ms-received-port=59333;ms-received-cid=4896700
CONTENT-LENGTH: 0
SERVER: RTC/4.0
ms-diagnostics: 1003;reason="User does not exist";TargetUri="[email protected]";source="sip.abc.com"
ms-edge-proxy-message-trust: ms-source-type=EdgeProxyGenerated;ms-ep-fqdn=edge.abc.com;ms-source-verified-user=verified
------------EndOfIncoming SipMessage
Thanks,
Deepak

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