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

Bogdan-Andrei Iancu bogdan at opensips.org
Wed Jan 6 10:49:44 CET 2016


Hi Zahid,

When in the dialog is the INFO injected ? in the early stage ? or after 
the call is established ?

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.12.2015 17:12, Zahid Mehmood wrote:
> 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 
> <jeff.pyle at fidelityvoice.com <mailto:jeff.pyle at fidelityvoice.com>> 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 <zm23 at columbia.edu
>     <mailto: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
>         <http://sip:10301@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
>         <mailto:sip%3A1112223333 at 10.10.1.1>>;party=calling;screen=yes;privacy=off.
>         From: "1112223333" <sip:1112223333 at 10.10.1.1
>         <mailto:sip%3A1112223333 at 10.10.1.1>>;tag=5745CCC-1C72.
>         To: <sip:10301 at 10.10.2.2 <mailto:sip%3A10301 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
>         <mailto: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
>         <http://sip:1112223333@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
>         <http://sip:10301@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
>         <mailto:sip%3A1112223333 at 10.10.1.1>>;tag=5745CCC-1C72.
>         To: <sip:10301 at 10.10.2.2
>         <mailto:sip%3A10301 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
>         <mailto: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
>         <http://sip:1112223333@10.10.1.1:5060>>.
>         Remote-Party-ID: "WIRELESS CALLER" <sip:1112223333 at 10.10.1.1
>         <mailto:sip%3A1112223333 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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