[cisco-voip] Issue with anonymous calls on a SIP trunk

[Andy]

Hi Brian,
I only have a sniffer trace to hand at the moment
I've changed the ip addressing and the numbers to protect the innocent.

Internet Protocol Version 4, Src: 10.1.1.2 (10.1.1.2), Dst: 10.1.1.1 
(10.1.1.1)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
     Status-Line: SIP/2.0 200 OK
         Status-Code: 200
         [Resent Packet: False]
     Message Header
         Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bKbjrn5f305g111q46j2j1.1
             Transport: UDP
             Sent-by Address: 10.1.1.1
             Sent-by port: 5060
             Branch: z9hG4bKbjrn5f305g111q46j2j1.1
         From: 
"Anonymous"<sip:anonymous at 10.1.1.1>;tag=140140856-1386321996272-
             SIP Display info: "Anonymous"
             SIP from address: sip:anonymous at 10.1.1.1
                 SIP from address User Part: anonymous
                 SIP from address Host Part: 10.1.1.1
             SIP tag: 140140856-1386321996272-
         To: "44InboundDDI 
44InBoundDDI"<sip:44InBoundDDI@"Domain">;tag=585CA458-26EF
             SIP Display info: "44InboundDDI 44InBoundDDI"
             SIP to address: sip:44InBoundDDI@"Domain"
                 SIP to address User Part: 44InBoundDDI
                 SIP to address Host Part: "Domain"
             SIP tag: 585CA458-26EF
         Date: Fri, 06 Dec 2013 09:26:36 GMT
         Call-ID: BW092636272061213411136895 at 10.81.253.80
         CSeq: 597521657 INVITE
         Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY, INFO, REGISTER
         Allow-Events: telephone-event
         Contact: <sip:44InBoundDDI at 10.1.1.2:5060>
             Contact-URI: sip:44InBoundDDI at 10.1.1.2:5060
         Supported: replaces
         Supported: sdp-anat
         Server: Cisco-SIPGateway/IOS-15.3.2.T1
         Supported: timer
         Content-Type: application/sdp
         Content-Disposition: session;handling=required
         Content-Length: 246
     Message Body
         Session Description Protocol
             Session Description Protocol Version (v): 0
             Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 
9234 6163 IN IP4 10.1.1.2
                 Owner Username: CiscoSystemsSIP-GW-UserAgent
                 Session ID: 9234
                 Session Version: 6163
                 Owner Network Type: IN
                 Owner Address Type: IP4
                 Owner Address: 10.1.1.2
             Session Name (s): SIP Call
             Connection Information (c): IN IP4 "CallManager IP Address"
             Time Description, active time (t): 0 0
                 Session Start Time: 0
                 Session Stop Time: 0
             Media Description, name and address (m): audio 26000 
RTP/AVP 0 101
             Connection Information (c): IN IP4 "CallManager IP Address"
             Media Attribute (a): rtpmap:0 PCMU/8000
             Media Attribute (a): rtpmap:101 telephone-event/8000
             Media Attribute (a): fmtp:101 0-15
             Media Attribute (a): ptime:20

Regards

Andy

On 10/12/2013 14:36, Brian Meade (brmeade) wrote:
> Andy,
>
> Can you copy what the Update message looks like so we can see what header the CUCM IP address is in?  You should be able to use a SIP Profile on the CUBE to change this to the CUBE's external IP address.
>
> Brian
>
> -----Original Message-----
> From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andy
> Sent: Tuesday, December 10, 2013 6:32 AM
> To: Cisco VoIP List
> Subject: [cisco-voip] Issue with anonymous calls on a SIP trunk
>
> Hi,
> I have an issue with anonymous (callerid witheld) calls on a SIP trunk which I can't figure out.
>
> Call comes in over sip trunk via a cube to cucm, if the callerid is know then the call gets placed to the dialed number ok.
> But if the number is withheld on the inbound call leg their is an additional update message which contains the CUCM ip address, but the SIP provider is unable to route to this address so one way voice is the result.
>
> Does anyone have any idea how to fix this?
>
> --
> Regards
>
> Andy
>
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