Hi,
We have a customer that has a problem with incoming calls (ISDN) that have a restricted calling-id (private calls).
Such calls to a private extension work, no problem. Call is announced as "Anonymous".
When the private extension is forwarded to an external location the call fails.
My guess is that CUCM doesn't know the call is private. The SIP debugs do not show any "privacy" attributes except for the name "Anonymous".
Is this normal?
IOS 12.4(25b) on a 2800 series router.
Regards,
Erik Tamminga
The following was caputered via debug ccsip and debug isdn q931:
688: 4416316: .Nov 3 10:11:07: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0049
689: <009>Sending Complete
690: <009>Bearer Capability i = 0x8090A3
691: <009><009>Standard = CCITT
692: <009><009>Transfer Capability = Speech
693: <009><009>Transfer Mode = Circuit
694: <009><009>Transfer Rate = 64 kbit/s
695: <009>Channel ID i = 0xA18381
696: <009><009>Preferred, Channel 1
697: <009>Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info
698: <009>Calling Party Number i = 0x00A3, N/A
699: <009><009>Plan:Unknown, Type:Unknown
700: <009>Called Party Number i = 0xA1, '20xxxxxxx'
701: <009><009>Plan:ISDN, Type:National
702: 4416317: .Nov 3 10:11:07: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
703: 4416318: .Nov 3 10:11:07: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
704: Sent:
705: INVITE sip:20xxxxxxx@10.x.x.y:5060 SIP/2.0
706: Via: SIP/2.0/TCP 10.x.x.x;branch=z9hG4bK25F6414F7
707: From: "anonymous" <sip:10.x.x.x>;tag=FF3D4A74-257E
708: To: <sip:20xxxxxxx@10.x.x.y>
709: Date: Thu, 03 Nov 2011 08:11:07 GMT
710: Call-ID:
37B41D2E-52A11E1-9E70F3E0-33A091A8@10.x.x.x
711: Supported: 100rel,timer,replaces
712: Min-SE: 1800
713: Cisco-Guid: 934430902-86643169-2992242724-3300127776
714: User-Agent: Cisco-SIPGateway/IOS-12.x
715: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, U
716: PDATE, REGISTER
717: CSeq: 101 INVITE
718: Max-Forwards: 70
719: Timestamp: 1320307867
720: Contact: <sip:10.x.x.x:5060;transport=tcp>
721: Expires: 180
722: Allow-Events: telephone-event
723: Content-Type: application/sdp
724: Content-Length: 499
725:
726: v=0
727: o=CiscoSystemsSIP-GW-UserAgent 8918 7322 IN IP4 10.x.x.x
728: s=SIP Call
729: c=IN IP4 10.x.x.x
730: t=0 0
731: m=audio 18328 RTP/AVP 0 8 18 100 101
732: c=IN IP4 10.x.x.x
733: a=rtpmap:0 PCMU/8000
734: a=rtpmap:8 PCMA/8000
735: a=rtpmap:18 G729/8000
736: a=fmtp:18 annexb=no
737: a=rtpmap:100 X-NSE/8000 688: 4416316: .Nov 3 10:11:07: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0049
689: <009>Sending Complete
690: <009>Bearer Capability i = 0x8090A3
691: <009><009>Standard = CCITT
692: <009><009>Transfer Capability = Speech
693: <009><009>Transfer Mode = Circuit
694: <009><009>Transfer Rate = 64 kbit/s
695: <009>Channel ID i = 0xA18381
696: <009><009>Preferred, Channel 1
697: <009>Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info
698: <009>Calling Party Number i = 0x00A3, N/A
699: <009><009>Plan:Unknown, Type:Unknown
700: <009>Called Party Number i = 0xA1, '20xxxxxxx'
701: <009><009>Plan:ISDN, Type:National
702: 4416317: .Nov 3 10:11:07: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
703: 4416318: .Nov 3 10:11:07: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
704: Sent:
705: INVITE sip:20xxxxxxx@10.x.x.y:5060 SIP/2.0
706: Via: SIP/2.0/TCP 10.x.x.x;branch=z9hG4bK25F6414F7
707: From: "anonymous" <sip:10.x.x.x>;tag=FF3D4A74-257E
708: To: <sip:20xxxxxxx@10.x.x.y>
709: Date: Thu, 03 Nov 2011 08:11:07 GMT
710: Call-ID: 37B41D2E-52A11E1-9E70F3E0-33A091A8@10.x.x.x
711: Supported: 100rel,timer,replaces
712: Min-SE: 1800
713: Cisco-Guid: 934430902-86643169-2992242724-3300127776
714: User-Agent: Cisco-SIPGateway/IOS-12.x
715: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, U
716: PDATE, REGISTER
717: CSeq: 101 INVITE
718: Max-Forwards: 70
719: Timestamp: 1320307867
720: Contact: <sip:10.x.x.x:5060;transport=tcp>
721: Expires: 180
722: Allow-Events: telephone-event
723: Content-Type: application/sdp
724: Content-Length: 499
725:
726: v=0
727: o=CiscoSystemsSIP-GW-UserAgent 8918 7322 IN IP4 10.x.x.x
728: s=SIP Call
729: c=IN IP4 10.x.x.x
730: t=0 0
731: m=audio 18328 RTP/AVP 0 8 18 100 101
732: c=IN IP4 10.x.x.x
733: a=rtpmap:0 PCMU/8000
734: a=rtpmap:8 PCMA/8000
735: a=rtpmap:18 G729/8000
736: a=fmtp:18 annexb=no
737: a=rtpmap:100 X-NSE/8000
