[OpenSIPS-Users] Processing calling-name(CNAM) from PRI

Jeff Pyle jeff.pyle at fidelityvoice.com
Fri Dec 18 15:44:16 CET 2015


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 <zm23 at columbia.edu> 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:10301 at 10.10.2.2:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3A2284.
> Remote-Party-ID: "1112223333" <sip:1112223333 at 10.10.1.1
> >;party=calling;screen=yes;privacy=off.
> From: "1112223333" <sip:1112223333 at 10.10.1.1>;tag=5745CCC-1C72.
> To: <sip:10301 at 10.10.2.2>.
> Date: Thu, 17 Dec 2015 19:20:31 GMT.
> Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E at 10.10.1.1.
> 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:1112223333 at 10.10.1.1:5060>.
> 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:10301 at 10.219.136.69:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bK3C1EC0.
> From: "1112223333" <sip:1112223333 at 10.10.1.1>;tag=5745CCC-1C72.
> To: <sip:10301 at 10.10.2.2>;tag=1768D8EC-CCDB1323.
> Date: Thu, 17 Dec 2015 19:20:31 GMT.
> Call-ID: 12968BB5-A42A11E5-8062F2AF-E28C686E at 10.10.1.1.
> 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:1112223333 at 10.10.1.1:5060>.
> Remote-Party-ID: "WIRELESS CALLER" <sip:1112223333 at 10.10.1.1
> >;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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