Hi Jeff,
   Thanks for your response.  From what I understand from Cisco
documentation, gateway seems to act that way when using H323 but, sadly,
not for SIP.  I will push Cisco about this.

Assuming the worst, Is there any thing I can try on the proxy side to get
the desired results?

Regards,
--
Zahid


On Fri, Dec 18, 2015 at 9:44 AM, Jeff Pyle <[email protected]>
wrote:

> In Adtran TA900 series gateways (very Cisco-like) I'm able to configure
> the PRI interface to wait for the FACILITY message before sending the
> initial INVITE.  When the INVITE does leave the gateway towards the proxy,
> it has full caller name information.  Perhaps something like this is
> available on the Cisco.  I hope so, because if not, you're going to have a
> difficult time integrating the INFO message.
>
>
> - Jeff
>
>
> On Thu, Dec 17, 2015 at 2:53 PM, Zahid Mehmood <[email protected]> wrote:
>
>> Hi,
>>    I am having trouble figuring out how to process the calling-name
>> coming from the PRI. In my setup, PRI is connected to a Cisco media gateway
>> which sends traffic to the proxy servers.  Calling name is not coming  in
>> the ISDN setup message.  It is actually provided in a separate facility
>> message [1].
>>
>> Cisco gateway processes this secondary messages and generates a INFO
>> message.  Polycom phone sends the 200 ok message but there is no change in
>> the visible caller id.
>>
>> Does anyone have a working example or suggestion of how this is supposed
>> to work?
>>
>> Invite:
>>
>> U 2015/12/17 14:20:31.215540 10.10.1.1:50975 -> 10.10.2.2:5060
>> INVITE sip:[email protected]:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3A2284.
>> Remote-Party-ID: "1112223333" <sip:[email protected]
>> >;party=calling;screen=yes;privacy=off.
>> From: "1112223333" <sip:[email protected]>;tag=5745CCC-1C72.
>> To: <sip:[email protected]>.
>> Date: Thu, 17 Dec 2015 19:20:31 GMT.
>> Call-ID: [email protected].
>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat.
>> Min-SE:  1800.
>> Cisco-Guid: 0311776101-2754220517-2148597785-1445067520.
>> 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: 1450380031.
>> Contact: <sip:[email protected]:5060>.
>> Expires: 180.
>> Allow-Events: telephone-event.
>> Content-Type: application/sdp.
>> Content-Disposition: session;handling=required.
>> Content-Length: 279.
>> .
>> v=0.
>> o=CiscoSystemsSIP-GW-UserAgent 3918 6190 IN IP4 10.10.1.1.
>> s=SIP Call.
>> c=IN IP4 10.10.1.1.
>> t=0 0.
>> m=audio 18854 RTP/AVP 0 18 101.
>> c=IN IP4 10.10.1.1.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>>
>> Invite messages:
>>
>> U 2015/12/17 14:20:31.546310 10.10.1.1:50975 -> 10.10.2.2:5060
>> INFO sip:[email protected]:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3C1EC0.
>> From: "1112223333" <sip:[email protected]>;tag=5745CCC-1C72.
>> To: <sip:[email protected]>;tag=1768D8EC-CCDB1323.
>> Date: Thu, 17 Dec 2015 19:20:31 GMT.
>> Call-ID: [email protected].
>> User-Agent: Cisco-SIPGateway/IOS-12.x.
>> Max-Forwards: 70.
>> Route: <sip:10.10.2.2;lr=on;ftag=5745CCC-1C72>.
>> Timestamp: 1450380031.
>> CSeq: 103 INFO.
>> Contact: <sip:[email protected]:5060>.
>> Remote-Party-ID: "WIRELESS CALLER" <sip:[email protected]
>> >;party=calling;screen=no;privacy=off.
>> Content-Length: 0.
>> .
>>
>>
>> Best Regards,
>>
>> --
>> Zahid
>>
>> [1]
>> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-isdn.html#GUID-53D5C9AB-AAC4-4178-8158-0DAEFB5BC33E
>> (figure 2 is close to what we are seeing)
>>
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