12-06-2018 03:55 AM - edited 12-06-2018 03:59 AM
Dear All,
I'm migrating from ISDN/MGCP to SIP and I've noticed with SIP when my ITSP sends calls where the caller id (ANI) is withheld the phone displays the empty string rather than "Unknown Number". This is disturbing for our end users.
My Call Flow is:
ITSP -> CUBE v12 -> CUCM v10.5
The SIP Invite on the CUBE is as follows:
! ITSP Invite: INVITE sip:+44123123123@199.0.0.1:5060 SIP/2.0 Via: SIP/2.0/TCP 199.0.0.2:5060;branch=z9hG4bK548491459 From: <sip:anonymous@anonymous.invalid>;tag=5257BBDE-17E1 To: <sip:+44123123123@199.0.0.1> Date: Thu, 06 Dec 2018 10:59:56 GMT Call-ID: 84E55FCC-F87411E8-80F198FC-B323C3A5@199.0.0.2 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 360 User-Agent: Cisco-SIPGateway/IOS Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1544090396 Contact: <sip:anonymous@199.0.0.2:5060;transport=tcp> Expires: 180 Allow-Events: telephone-event Max-Forwards: 26 Resource-Priority: dsn-000000.routine Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 271 [...] ! CUBE Relays this to the CM: INVITE sip:+44123123123@10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/TCP 10.0.0.226:5060;branch=z9hG4bKED51DF1 From: <sip:anonymous@10.0.0.226>;tag=221C185-2312 To: <sip:+44123123123@10.0.0.2> Date: Thu, 06 Dec 2018 09:59:56 GMT Call-ID: 84E58722-F87411E8-AA5DAFDC-B1823B0D@10.0.0.226 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 360 User-Agent: Cisco-SIPGateway/IOS Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1544090396 Contact: <sip:anonymous@10.0.0.226:5060;transport=tcp> Expires: 180 Allow-Events: telephone-event Max-Forwards: 25 Resource-Priority: dsn-000000.routine Session-ID: fcd5850e8020511b9cdc442e72084001;remote=00000000000000000000000000000000 Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 250
Is anyone aware of a smart way on CUBE or CM to indicate the ringing phone should display "Unknown Number"?
The only solution that comes to mind is to insert a PAI field, then modify it, but this seems unusual that a modification like this would be needed to keep the original behaviour under MGCP.
voice service voip sip asserted-id pai ! voice class sip-profiles 1 rule X request ANY sip-header P-Asserted-Identity modify "<sip:anonymous@" "\"Unknown Number\" <sip:anonymous@"
Regards
James.
12-22-2018 07:13 AM
12-26-2018 03:09 PM
Hi Jonathan,
Thank you for pointing me to the RFC. I'm not sure that the ITSP is adhering to the standard though because section 8.1.1.3 states:
A UAC SHOULD use the display name "Anonymous", along with a syntactically correct, but otherwise meaningless URI
I would say the ITSP failed to send the display name "Anonymous" so I have added it via the PAI as "Unknown Number". This appears to be working so far with my testing.
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